BnCrypto: fix use-before-init in CREATE_PLUGIN am: 4bbfb6d881 am: a9296786e3 am: 245e44d9cf Change-Id: I480db45b78b567a832d591df0834dbdae6cf88d0

diff --git a/Android.bp b/Android.bp
index a3679b1..e4f12c8 100644
--- a/Android.bp
+++ b/Android.bp

@@ -2,5 +2,6 @@
     "camera",
     "drm/*",
     "media/*",
+    "services/*",
     "soundtrigger",
 ]

diff --git a/camera/Android.bp b/camera/Android.bp
index c76ae50..24b3918 100644
--- a/camera/Android.bp
+++ b/camera/Android.bp

@@ -29,12 +29,7 @@
         // AIDL files for camera interfaces
         // The headers for these interfaces will be available to any modules that
         // include libcamera_client, at the path "aidl/package/path/BnFoo.h"
-        "aidl/android/hardware/ICameraService.aidl",
-        "aidl/android/hardware/ICameraServiceListener.aidl",
-        "aidl/android/hardware/ICameraServiceProxy.aidl",
-        "aidl/android/hardware/camera2/ICameraDeviceCallbacks.aidl",
-        "aidl/android/hardware/camera2/ICameraDeviceUser.aidl",
-
+        ":libcamera_client_aidl",
 
         // Source for camera interface parcelables, and manually-written interfaces
         "Camera.cpp",
@@ -81,3 +76,25 @@
     ],
 
 }
+
+// AIDL interface between camera clients and the camera service.
+filegroup {
+    name: "libcamera_client_aidl",
+    srcs: [
+        "aidl/android/hardware/ICameraService.aidl",
+        "aidl/android/hardware/ICameraServiceListener.aidl",
+        "aidl/android/hardware/ICameraServiceProxy.aidl",
+        "aidl/android/hardware/camera2/ICameraDeviceCallbacks.aidl",
+        "aidl/android/hardware/camera2/ICameraDeviceUser.aidl",
+    ],
+}
+
+// Extra AIDL files that are used by framework.jar but not libcamera_client
+// because they have hand-written native implementations.
+filegroup {
+    name: "libcamera_client_framework_aidl",
+    srcs: [
+        "aidl/android/hardware/ICamera.aidl",
+        "aidl/android/hardware/ICameraClient.aidl",
+    ],
+}

diff --git a/camera/CaptureResult.cpp b/camera/CaptureResult.cpp
index e6c0d00..928a6bc 100644
--- a/camera/CaptureResult.cpp
+++ b/camera/CaptureResult.cpp

@@ -60,6 +60,39 @@
     return OK;
 }
 
+status_t PhysicalCaptureResultInfo::readFromParcel(const android::Parcel* parcel) {
+    status_t res;
+
+    mPhysicalCameraId.remove(mPhysicalCameraId.size());
+    mPhysicalCameraMetadata.clear();
+
+    if ((res = parcel->readString16(&mPhysicalCameraId)) != OK) {
+        ALOGE("%s: Failed to read camera id: %d", __FUNCTION__, res);
+        return res;
+    }
+
+    if ((res = mPhysicalCameraMetadata.readFromParcel(parcel)) != OK) {
+        ALOGE("%s: Failed to read metadata from parcel: %d", __FUNCTION__, res);
+        return res;
+    }
+    return OK;
+}
+
+status_t PhysicalCaptureResultInfo::writeToParcel(android::Parcel* parcel) const {
+    status_t res;
+    if ((res = parcel->writeString16(mPhysicalCameraId)) != OK) {
+        ALOGE("%s: Failed to write physical camera ID to parcel: %d",
+                __FUNCTION__, res);
+        return res;
+    }
+    if ((res = mPhysicalCameraMetadata.writeToParcel(parcel)) != OK) {
+        ALOGE("%s: Failed to write physical camera metadata to parcel: %d",
+                __FUNCTION__, res);
+        return res;
+    }
+    return OK;
+}
+
 CaptureResult::CaptureResult() :
         mMetadata(), mResultExtras() {
 }
@@ -67,6 +100,7 @@
 CaptureResult::CaptureResult(const CaptureResult &otherResult) {
     mResultExtras = otherResult.mResultExtras;
     mMetadata = otherResult.mMetadata;
+    mPhysicalMetadatas = otherResult.mPhysicalMetadatas;
 }
 
 status_t CaptureResult::readFromParcel(android::Parcel *parcel) {
@@ -79,6 +113,7 @@
     }
 
     mMetadata.clear();
+    mPhysicalMetadatas.clear();
 
     status_t res = OK;
     res = mMetadata.readFromParcel(parcel);
@@ -89,6 +124,34 @@
     }
     ALOGV("%s: Read metadata from parcel", __FUNCTION__);
 
+    int32_t physicalMetadataCount;
+    if ((res = parcel->readInt32(&physicalMetadataCount)) != OK) {
+        ALOGE("%s: Failed to read the physical metadata count from parcel: %d", __FUNCTION__, res);
+        return res;
+    }
+    if (physicalMetadataCount < 0) {
+        ALOGE("%s: Invalid physical metadata count from parcel: %d",
+                __FUNCTION__, physicalMetadataCount);
+        return BAD_VALUE;
+    }
+
+    for (int32_t i = 0; i < physicalMetadataCount; i++) {
+        String16 cameraId;
+        if ((res = parcel->readString16(&cameraId)) != OK) {
+            ALOGE("%s: Failed to read camera id: %d", __FUNCTION__, res);
+            return res;
+        }
+
+        CameraMetadata physicalMetadata;
+        if ((res = physicalMetadata.readFromParcel(parcel)) != OK) {
+            ALOGE("%s: Failed to read metadata from parcel: %d", __FUNCTION__, res);
+            return res;
+        }
+
+        mPhysicalMetadatas.emplace(mPhysicalMetadatas.end(), cameraId, physicalMetadata);
+    }
+    ALOGV("%s: Read physical metadata from parcel", __FUNCTION__);
+
     res = mResultExtras.readFromParcel(parcel);
     if (res != OK) {
         ALOGE("%s: Failed to read result extras from parcel.",
@@ -118,6 +181,27 @@
     }
     ALOGV("%s: Wrote metadata to parcel", __FUNCTION__);
 
+    int32_t physicalMetadataCount = static_cast<int32_t>(mPhysicalMetadatas.size());
+    res = parcel->writeInt32(physicalMetadataCount);
+    if (res != OK) {
+        ALOGE("%s: Failed to write physical metadata count to parcel: %d",
+                __FUNCTION__, res);
+        return BAD_VALUE;
+    }
+    for (const auto& physicalMetadata : mPhysicalMetadatas) {
+        if ((res = parcel->writeString16(physicalMetadata.mPhysicalCameraId)) != OK) {
+            ALOGE("%s: Failed to write physical camera ID to parcel: %d",
+                    __FUNCTION__, res);
+            return res;
+        }
+        if ((res = physicalMetadata.mPhysicalCameraMetadata.writeToParcel(parcel)) != OK) {
+            ALOGE("%s: Failed to write physical camera metadata to parcel: %d",
+                    __FUNCTION__, res);
+            return res;
+        }
+    }
+    ALOGV("%s: Wrote physical camera metadata to parcel", __FUNCTION__);
+
     res = mResultExtras.writeToParcel(parcel);
     if (res != OK) {
         ALOGE("%s: Failed to write result extras to parcel", __FUNCTION__);

diff --git a/camera/aidl/android/hardware/ICameraServiceProxy.aidl b/camera/aidl/android/hardware/ICameraServiceProxy.aidl
index 5dc23eb..7575948 100644
--- a/camera/aidl/android/hardware/ICameraServiceProxy.aidl
+++ b/camera/aidl/android/hardware/ICameraServiceProxy.aidl

@@ -46,8 +46,14 @@
     const int CAMERA_FACING_EXTERNAL = 2;
 
     /**
+     * Values for notifyCameraState api level
+     */
+     const int CAMERA_API_LEVEL_1 = 1;
+     const int CAMERA_API_LEVEL_2 = 2;
+
+    /**
      * Update the status of a camera device.
      */
     oneway void notifyCameraState(String cameraId, int facing, int newCameraState,
-            String clientName);
+            String clientName, int apiLevel);
 }

diff --git a/camera/aidl/android/hardware/camera2/ICameraDeviceCallbacks.aidl b/camera/aidl/android/hardware/camera2/ICameraDeviceCallbacks.aidl
index 28252c0..58b19a3 100644
--- a/camera/aidl/android/hardware/camera2/ICameraDeviceCallbacks.aidl
+++ b/camera/aidl/android/hardware/camera2/ICameraDeviceCallbacks.aidl

@@ -18,6 +18,7 @@
 
 import android.hardware.camera2.impl.CameraMetadataNative;
 import android.hardware.camera2.impl.CaptureResultExtras;
+import android.hardware.camera2.impl.PhysicalCaptureResultInfo;
 
 /** @hide */
 interface ICameraDeviceCallbacks
@@ -30,12 +31,14 @@
     const int ERROR_CAMERA_REQUEST = 3;
     const int ERROR_CAMERA_RESULT = 4;
     const int ERROR_CAMERA_BUFFER = 5;
+    const int ERROR_CAMERA_DISABLED = 6;
 
     oneway void onDeviceError(int errorCode, in CaptureResultExtras resultExtras);
     oneway void onDeviceIdle();
     oneway void onCaptureStarted(in CaptureResultExtras resultExtras, long timestamp);
     oneway void onResultReceived(in CameraMetadataNative result,
-                                 in CaptureResultExtras resultExtras);
+                                 in CaptureResultExtras resultExtras,
+                                 in PhysicalCaptureResultInfo[] physicalCaptureResultInfos);
     oneway void onPrepared(int streamId);
 
     /**

diff --git a/camera/aidl/android/hardware/camera2/ICameraDeviceUser.aidl b/camera/aidl/android/hardware/camera2/ICameraDeviceUser.aidl
index 0771fc8..4ced08c 100644
--- a/camera/aidl/android/hardware/camera2/ICameraDeviceUser.aidl
+++ b/camera/aidl/android/hardware/camera2/ICameraDeviceUser.aidl

@@ -79,8 +79,9 @@
      * <p>
      * @param operatingMode The kind of session to create; either NORMAL_MODE or
      *     CONSTRAINED_HIGH_SPEED_MODE. Must be a non-negative value.
+     * @param sessionParams Session wide camera parameters
      */
-    void endConfigure(int operatingMode);
+    void endConfigure(int operatingMode, in CameraMetadataNative sessionParams);
 
     void deleteStream(int streamId);
 
@@ -140,5 +141,7 @@
 
     void prepare2(int maxCount, int streamId);
 
+    void updateOutputConfiguration(int streamId, in OutputConfiguration outputConfiguration);
+
     void finalizeOutputConfigurations(int streamId, in OutputConfiguration outputConfiguration);
 }

diff --git a/camera/aidl/android/hardware/camera2/impl/PhysicalCaptureResultInfo.aidl b/camera/aidl/android/hardware/camera2/impl/PhysicalCaptureResultInfo.aidl
new file mode 100644
index 0000000..78d9b7b
--- /dev/null
+++ b/camera/aidl/android/hardware/camera2/impl/PhysicalCaptureResultInfo.aidl

@@ -0,0 +1,20 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.hardware.camera2.impl;
+
+/** @hide */
+parcelable PhysicalCaptureResultInfo cpp_header "camera/CaptureResult.h";

diff --git a/camera/camera2/CaptureRequest.cpp b/camera/camera2/CaptureRequest.cpp
index 0597950..1843ec4 100644
--- a/camera/camera2/CaptureRequest.cpp
+++ b/camera/camera2/CaptureRequest.cpp

@@ -18,6 +18,7 @@
 // #define LOG_NDEBUG 0
 #define LOG_TAG "CameraRequest"
 #include <utils/Log.h>
+#include <utils/String16.h>
 
 #include <camera/camera2/CaptureRequest.h>
 
@@ -42,16 +43,46 @@
         return BAD_VALUE;
     }
 
-    mMetadata.clear();
     mSurfaceList.clear();
+    mStreamIdxList.clear();
+    mSurfaceIdxList.clear();
+    mPhysicalCameraSettings.clear();
 
     status_t err = OK;
 
-    if ((err = mMetadata.readFromParcel(parcel)) != OK) {
-        ALOGE("%s: Failed to read metadata from parcel", __FUNCTION__);
+    int32_t settingsCount;
+    if ((err = parcel->readInt32(&settingsCount)) != OK) {
+        ALOGE("%s: Failed to read the settings count from parcel: %d", __FUNCTION__, err);
         return err;
     }
-    ALOGV("%s: Read metadata from parcel", __FUNCTION__);
+
+    if (settingsCount <= 0) {
+        ALOGE("%s: Settings count %d should always be positive!", __FUNCTION__, settingsCount);
+        return BAD_VALUE;
+    }
+
+    for (int32_t i = 0; i < settingsCount; i++) {
+        String16 id;
+        if ((err = parcel->readString16(&id)) != OK) {
+            ALOGE("%s: Failed to read camera id!", __FUNCTION__);
+            return BAD_VALUE;
+        }
+
+        CameraMetadata settings;
+        if ((err = settings.readFromParcel(parcel)) != OK) {
+            ALOGE("%s: Failed to read metadata from parcel", __FUNCTION__);
+            return err;
+        }
+        ALOGV("%s: Read metadata from parcel", __FUNCTION__);
+        mPhysicalCameraSettings.push_back({std::string(String8(id).string()), settings});
+    }
+
+    int isReprocess = 0;
+    if ((err = parcel->readInt32(&isReprocess)) != OK) {
+        ALOGE("%s: Failed to read reprocessing from parcel", __FUNCTION__);
+        return err;
+    }
+    mIsReprocess = (isReprocess != 0);
 
     int32_t size;
     if ((err = parcel->readInt32(&size)) != OK) {
@@ -61,7 +92,7 @@
     ALOGV("%s: Read surface list size = %d", __FUNCTION__, size);
 
     // Do not distinguish null arrays from 0-sized arrays.
-    for (int i = 0; i < size; ++i) {
+    for (int32_t i = 0; i < size; ++i) {
         // Parcel.writeParcelableArray
         size_t len;
         const char16_t* className = parcel->readString16Inplace(&len);
@@ -88,12 +119,32 @@
         mSurfaceList.push_back(surface);
     }
 
-    int isReprocess = 0;
-    if ((err = parcel->readInt32(&isReprocess)) != OK) {
-        ALOGE("%s: Failed to read reprocessing from parcel", __FUNCTION__);
+    int32_t streamSurfaceSize;
+    if ((err = parcel->readInt32(&streamSurfaceSize)) != OK) {
+        ALOGE("%s: Failed to read streamSurfaceSize from parcel", __FUNCTION__);
         return err;
     }
-    mIsReprocess = (isReprocess != 0);
+
+    if (streamSurfaceSize < 0) {
+        ALOGE("%s: Bad streamSurfaceSize %d from parcel", __FUNCTION__, streamSurfaceSize);
+        return BAD_VALUE;
+    }
+
+    for (int32_t i = 0; i < streamSurfaceSize; ++i) {
+        int streamIdx;
+        if ((err = parcel->readInt32(&streamIdx)) != OK) {
+            ALOGE("%s: Failed to read stream index from parcel", __FUNCTION__);
+            return err;
+        }
+        mStreamIdxList.push_back(streamIdx);
+
+        int surfaceIdx;
+        if ((err = parcel->readInt32(&surfaceIdx)) != OK) {
+            ALOGE("%s: Failed to read surface index from parcel", __FUNCTION__);
+            return err;
+        }
+        mSurfaceIdxList.push_back(surfaceIdx);
+    }
 
     return OK;
 }
@@ -106,32 +157,62 @@
 
     status_t err = OK;
 
-    if ((err = mMetadata.writeToParcel(parcel)) != OK) {
+    int32_t settingsCount = static_cast<int32_t>(mPhysicalCameraSettings.size());
+
+    if ((err = parcel->writeInt32(settingsCount)) != OK) {
+        ALOGE("%s: Failed to write settings count!", __FUNCTION__);
         return err;
     }
 
-    int32_t size = static_cast<int32_t>(mSurfaceList.size());
+    for (const auto &it : mPhysicalCameraSettings) {
+        if ((err = parcel->writeString16(String16(it.id.c_str()))) != OK) {
+            ALOGE("%s: Failed to camera id!", __FUNCTION__);
+            return err;
+        }
 
-    // Send 0-sized arrays when it's empty. Do not send null arrays.
-    parcel->writeInt32(size);
-
-    for (int32_t i = 0; i < size; ++i) {
-        // not sure if readParcelableArray does this, hard to tell from source
-        parcel->writeString16(String16("android.view.Surface"));
-
-        // Surface.writeToParcel
-        view::Surface surfaceShim;
-        surfaceShim.name = String16("unknown_name");
-        surfaceShim.graphicBufferProducer = mSurfaceList[i]->getIGraphicBufferProducer();
-        if ((err = surfaceShim.writeToParcel(parcel)) != OK) {
-            ALOGE("%s: Failed to write output target Surface %d to parcel: %s (%d)",
-                    __FUNCTION__, i, strerror(-err), err);
+        if ((err = it.settings.writeToParcel(parcel)) != OK) {
+            ALOGE("%s: Failed to write settings!", __FUNCTION__);
             return err;
         }
     }
 
     parcel->writeInt32(mIsReprocess ? 1 : 0);
 
+    if (mSurfaceConverted) {
+        parcel->writeInt32(0); // 0-sized array
+    } else {
+        int32_t size = static_cast<int32_t>(mSurfaceList.size());
+
+        // Send 0-sized arrays when it's empty. Do not send null arrays.
+        parcel->writeInt32(size);
+
+        for (int32_t i = 0; i < size; ++i) {
+            // not sure if readParcelableArray does this, hard to tell from source
+            parcel->writeString16(String16("android.view.Surface"));
+
+            // Surface.writeToParcel
+            view::Surface surfaceShim;
+            surfaceShim.name = String16("unknown_name");
+            surfaceShim.graphicBufferProducer = mSurfaceList[i]->getIGraphicBufferProducer();
+            if ((err = surfaceShim.writeToParcel(parcel)) != OK) {
+                ALOGE("%s: Failed to write output target Surface %d to parcel: %s (%d)",
+                        __FUNCTION__, i, strerror(-err), err);
+                return err;
+            }
+        }
+    }
+
+    parcel->writeInt32(mStreamIdxList.size());
+    for (size_t i = 0; i < mStreamIdxList.size(); ++i) {
+        if ((err = parcel->writeInt32(mStreamIdxList[i])) != OK) {
+            ALOGE("%s: Failed to write stream index to parcel", __FUNCTION__);
+            return err;
+        }
+        if ((err = parcel->writeInt32(mSurfaceIdxList[i])) != OK) {
+            ALOGE("%s: Failed to write surface index to parcel", __FUNCTION__);
+            return err;
+        }
+    }
     return OK;
 }
 

diff --git a/camera/camera2/OutputConfiguration.cpp b/camera/camera2/OutputConfiguration.cpp
index 468a1eb..feb04c2 100644
--- a/camera/camera2/OutputConfiguration.cpp
+++ b/camera/camera2/OutputConfiguration.cpp

@@ -1,6 +1,6 @@
 /*
 **
-** Copyright 2015, The Android Open Source Project
+** Copyright 2015-2018, The Android Open Source Project
 **
 ** Licensed under the Apache License, Version 2.0 (the "License");
 ** you may not use this file except in compliance with the License.
@@ -64,6 +64,10 @@
     return mIsShared;
 }
 
+String16 OutputConfiguration::getPhysicalCameraId() const {
+    return mPhysicalCameraId;
+}
+
 OutputConfiguration::OutputConfiguration() :
         mRotation(INVALID_ROTATION),
         mSurfaceSetID(INVALID_SET_ID),
@@ -139,6 +143,8 @@
         return err;
     }
 
+    parcel->readString16(&mPhysicalCameraId);
+
     mRotation = rotation;
     mSurfaceSetID = setID;
     mSurfaceType = surfaceType;
@@ -153,19 +159,20 @@
         mGbps.push_back(surface.graphicBufferProducer);
     }
 
-    ALOGV("%s: OutputConfiguration: rotation = %d, setId = %d, surfaceType = %d",
-            __FUNCTION__, mRotation, mSurfaceSetID, mSurfaceType);
+    ALOGV("%s: OutputConfiguration: rotation = %d, setId = %d, surfaceType = %d,"
+          " physicalCameraId = %s", __FUNCTION__, mRotation, mSurfaceSetID,
+          mSurfaceType, String8(mPhysicalCameraId).string());
 
     return err;
 }
 
 OutputConfiguration::OutputConfiguration(sp<IGraphicBufferProducer>& gbp, int rotation,
-        int surfaceSetID) {
+        int surfaceSetID, bool isShared) {
     mGbps.push_back(gbp);
     mRotation = rotation;
     mSurfaceSetID = surfaceSetID;
     mIsDeferred = false;
-    mIsShared = false;
+    mIsShared = isShared;
 }
 
 status_t OutputConfiguration::writeToParcel(android::Parcel* parcel) const {
@@ -204,6 +211,9 @@
     err = parcel->writeParcelableVector(surfaceShims);
     if (err != OK) return err;
 
+    err = parcel->writeString16(mPhysicalCameraId);
+    if (err != OK) return err;
+
     return OK;
 }
 

diff --git a/camera/cameraserver/cameraserver.rc b/camera/cameraserver/cameraserver.rc
index fea5a1d..a9aae0b 100644
--- a/camera/cameraserver/cameraserver.rc
+++ b/camera/cameraserver/cameraserver.rc

@@ -4,3 +4,4 @@
     group audio camera input drmrpc
     ioprio rt 4
     writepid /dev/cpuset/camera-daemon/tasks /dev/stune/top-app/tasks
+    rlimit rtprio 10 10

diff --git a/camera/include/camera/CaptureResult.h b/camera/include/camera/CaptureResult.h
index 917d953..56fa178 100644
--- a/camera/include/camera/CaptureResult.h
+++ b/camera/include/camera/CaptureResult.h

@@ -91,14 +91,36 @@
     virtual status_t                readFromParcel(const android::Parcel* parcel) override;
     virtual status_t                writeToParcel(android::Parcel* parcel) const override;
 };
+
+struct PhysicalCaptureResultInfo : public android::Parcelable {
+
+    PhysicalCaptureResultInfo()
+        : mPhysicalCameraId(),
+          mPhysicalCameraMetadata() {
+    }
+    PhysicalCaptureResultInfo(const String16& cameraId,
+            const CameraMetadata& cameraMetadata)
+            : mPhysicalCameraId(cameraId),
+              mPhysicalCameraMetadata(cameraMetadata) {
+    }
+
+    String16  mPhysicalCameraId;
+    CameraMetadata mPhysicalCameraMetadata;
+
+    virtual status_t                readFromParcel(const android::Parcel* parcel) override;
+    virtual status_t                writeToParcel(android::Parcel* parcel) const override;
+};
+
 } // namespace impl
 } // namespace camera2
 } // namespace hardware
 
 using hardware::camera2::impl::CaptureResultExtras;
+using hardware::camera2::impl::PhysicalCaptureResultInfo;
 
 struct CaptureResult : public virtual LightRefBase<CaptureResult> {
     CameraMetadata          mMetadata;
+    std::vector<PhysicalCaptureResultInfo> mPhysicalMetadatas;
     CaptureResultExtras     mResultExtras;
 
     CaptureResult();

diff --git a/camera/include/camera/camera2/CaptureRequest.h b/camera/include/camera/camera2/CaptureRequest.h
index 0180183..506abab 100644
--- a/camera/include/camera/camera2/CaptureRequest.h
+++ b/camera/include/camera/camera2/CaptureRequest.h

@@ -40,14 +40,35 @@
     CaptureRequest(CaptureRequest&& rhs) noexcept;
     virtual ~CaptureRequest();
 
-    CameraMetadata          mMetadata;
+    struct PhysicalCameraSettings {
+        std::string id;
+        CameraMetadata settings;
+    };
+    std::vector<PhysicalCameraSettings> mPhysicalCameraSettings;
+
+    // Used by NDK client to pass surfaces by stream/surface index.
+    bool                    mSurfaceConverted = false;
+
+    // Starting in Android O, create a Surface from Parcel will take one extra
+    // IPC call.
     Vector<sp<Surface> >    mSurfaceList;
+    // Optional way of passing surface list since passing Surface over binder
+    // is expensive. Use the stream/surface index from current output configuration
+    // to represent an configured output Surface. When stream/surface index is used,
+    // set mSurfaceList to zero length to save unparcel time.
+    Vector<int>             mStreamIdxList;
+    Vector<int>             mSurfaceIdxList; // per stream surface list index
+
     bool                    mIsReprocess;
 
+    void*                   mContext; // arbitrary user context from NDK apps, null for java apps
+
     /**
      * Keep impl up-to-date with CaptureRequest.java in frameworks/base
      */
+    // used by cameraserver to receive CaptureRequest from java/NDK client
     status_t                readFromParcel(const android::Parcel* parcel) override;
+    // used by NDK client to send CaptureRequest to cameraserver
     status_t                writeToParcel(android::Parcel* parcel) const override;
 };
 

diff --git a/camera/include/camera/camera2/OutputConfiguration.h b/camera/include/camera/camera2/OutputConfiguration.h
index 8e641c7..a80f44b 100644
--- a/camera/include/camera/camera2/OutputConfiguration.h
+++ b/camera/include/camera/camera2/OutputConfiguration.h

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2015 The Android Open Source Project
+ * Copyright (C) 2015-2018 The Android Open Source Project
  *
  * Licensed under the Apache License, Version 2.0 (the "License");
  * you may not use this file except in compliance with the License.
@@ -46,6 +46,7 @@
     int                        getHeight() const;
     bool                       isDeferred() const;
     bool                       isShared() const;
+    String16                   getPhysicalCameraId() const;
     /**
      * Keep impl up-to-date with OutputConfiguration.java in frameworks/base
      */
@@ -64,7 +65,7 @@
     OutputConfiguration(const android::Parcel& parcel);
 
     OutputConfiguration(sp<IGraphicBufferProducer>& gbp, int rotation,
-            int surfaceSetID = INVALID_SET_ID);
+            int surfaceSetID = INVALID_SET_ID, bool isShared = false);
 
     bool operator == (const OutputConfiguration& other) const {
         return ( mRotation == other.mRotation &&
@@ -74,7 +75,8 @@
                 mHeight == other.mHeight &&
                 mIsDeferred == other.mIsDeferred &&
                 mIsShared == other.mIsShared &&
-                gbpsEqual(other));
+                gbpsEqual(other) &&
+                mPhysicalCameraId == other.mPhysicalCameraId );
     }
     bool operator != (const OutputConfiguration& other) const {
         return !(*this == other);
@@ -102,6 +104,9 @@
         if (mIsShared != other.mIsShared) {
             return mIsShared < other.mIsShared;
         }
+        if (mPhysicalCameraId != other.mPhysicalCameraId) {
+            return mPhysicalCameraId < other.mPhysicalCameraId;
+        }
         return gbpsLessThan(other);
     }
     bool operator > (const OutputConfiguration& other) const {
@@ -110,6 +115,7 @@
 
     bool gbpsEqual(const OutputConfiguration& other) const;
     bool gbpsLessThan(const OutputConfiguration& other) const;
+    void addGraphicProducer(sp<IGraphicBufferProducer> gbp) {mGbps.push_back(gbp);}
 private:
     std::vector<sp<IGraphicBufferProducer>> mGbps;
     int                        mRotation;
@@ -119,8 +125,7 @@
     int                        mHeight;
     bool                       mIsDeferred;
     bool                       mIsShared;
-    // helper function
-    static String16 readMaybeEmptyString16(const android::Parcel* parcel);
+    String16                   mPhysicalCameraId;
 };
 } // namespace params
 } // namespace camera2

diff --git a/camera/ndk/NdkCameraCaptureSession.cpp b/camera/ndk/NdkCameraCaptureSession.cpp
index 2a6b182..fd95296 100644
--- a/camera/ndk/NdkCameraCaptureSession.cpp
+++ b/camera/ndk/NdkCameraCaptureSession.cpp

@@ -135,3 +135,19 @@
     }
     return session->abortCaptures();
 }
+
+EXPORT
+camera_status_t ACameraCaptureSession_updateSharedOutput(ACameraCaptureSession* session,
+        ACaptureSessionOutput* output) {
+    ATRACE_CALL();
+    if (session == nullptr) {
+        ALOGE("%s: Error: session is null", __FUNCTION__);
+        return ACAMERA_ERROR_INVALID_PARAMETER;
+    }
+
+    if (session->isClosed()) {
+        ALOGE("%s: session %p is already closed", __FUNCTION__, session);
+        return ACAMERA_ERROR_SESSION_CLOSED;
+    }
+    return session->updateOutputConfiguration(output);
+}

diff --git a/camera/ndk/NdkCameraDevice.cpp b/camera/ndk/NdkCameraDevice.cpp
index 9f6d1f7..812a312 100644
--- a/camera/ndk/NdkCameraDevice.cpp
+++ b/camera/ndk/NdkCameraDevice.cpp

@@ -103,11 +103,74 @@
                 __FUNCTION__, window, out);
         return ACAMERA_ERROR_INVALID_PARAMETER;
     }
-    *out = new ACaptureSessionOutput(window);
+    *out = new ACaptureSessionOutput(window, false);
     return ACAMERA_OK;
 }
 
 EXPORT
+camera_status_t ACaptureSessionSharedOutput_create(
+        ANativeWindow* window, /*out*/ACaptureSessionOutput** out) {
+    ATRACE_CALL();
+    if (window == nullptr || out == nullptr) {
+        ALOGE("%s: Error: bad argument. window %p, out %p",
+                __FUNCTION__, window, out);
+        return ACAMERA_ERROR_INVALID_PARAMETER;
+    }
+    *out = new ACaptureSessionOutput(window, true);
+    return ACAMERA_OK;
+}
+
+EXPORT
+camera_status_t ACaptureSessionSharedOutput_add(ACaptureSessionOutput *out,
+        ANativeWindow* window) {
+    ATRACE_CALL();
+    if ((window == nullptr) || (out == nullptr)) {
+        ALOGE("%s: Error: bad argument. window %p, out %p",
+                __FUNCTION__, window, out);
+        return ACAMERA_ERROR_INVALID_PARAMETER;
+    }
+    if (!out->mIsShared) {
+        ALOGE("%s: Error trying to insert a new window in non-shared output configuration",
+                __FUNCTION__);
+        return ACAMERA_ERROR_INVALID_OPERATION;
+    }
+    if (out->mWindow == window) {
+        ALOGE("%s: Error trying to add the same window associated with the output configuration",
+                __FUNCTION__);
+        return ACAMERA_ERROR_INVALID_PARAMETER;
+    }
+
+    auto insert = out->mSharedWindows.insert(window);
+    camera_status_t ret = (insert.second) ? ACAMERA_OK : ACAMERA_ERROR_INVALID_PARAMETER;
+    return ret;
+}
+
+EXPORT
+camera_status_t ACaptureSessionSharedOutput_remove(ACaptureSessionOutput *out,
+        ANativeWindow* window) {
+    ATRACE_CALL();
+    if ((window == nullptr) || (out == nullptr)) {
+        ALOGE("%s: Error: bad argument. window %p, out %p",
+                __FUNCTION__, window, out);
+        return ACAMERA_ERROR_INVALID_PARAMETER;
+    }
+    if (!out->mIsShared) {
+        ALOGE("%s: Error trying to remove a  window in non-shared output configuration",
+                __FUNCTION__);
+        return ACAMERA_ERROR_INVALID_OPERATION;
+    }
+    if (out->mWindow == window) {
+        ALOGE("%s: Error trying to remove the same window associated with the output configuration",
+                __FUNCTION__);
+        return ACAMERA_ERROR_INVALID_PARAMETER;
+    }
+
+    auto remove = out->mSharedWindows.erase(window);
+    camera_status_t ret = (remove) ? ACAMERA_OK : ACAMERA_ERROR_INVALID_PARAMETER;
+    return ret;
+}
+
+EXPORT
 void ACaptureSessionOutput_free(ACaptureSessionOutput* output) {
     ATRACE_CALL();
     if (output != nullptr) {
@@ -157,5 +220,21 @@
                 __FUNCTION__, device, outputs, callbacks, session);
         return ACAMERA_ERROR_INVALID_PARAMETER;
     }
-    return device->createCaptureSession(outputs, callbacks, session);
+    return device->createCaptureSession(outputs, nullptr, callbacks, session);
+}
+
+EXPORT
+camera_status_t ACameraDevice_createCaptureSessionWithSessionParameters(
+        ACameraDevice* device,
+        const ACaptureSessionOutputContainer*       outputs,
+        const ACaptureRequest* sessionParameters,
+        const ACameraCaptureSession_stateCallbacks* callbacks,
+        /*out*/ACameraCaptureSession** session) {
+    ATRACE_CALL();
+    if (device == nullptr || outputs == nullptr || callbacks == nullptr || session == nullptr) {
+        ALOGE("%s: Error: invalid input: device %p, outputs %p, callbacks %p, session %p",
+                __FUNCTION__, device, outputs, callbacks, session);
+        return ACAMERA_ERROR_INVALID_PARAMETER;
+    }
+    return device->createCaptureSession(outputs, sessionParameters, callbacks, session);
 }

diff --git a/camera/ndk/NdkCaptureRequest.cpp b/camera/ndk/NdkCaptureRequest.cpp
index 5b4c180..ac1856b 100644
--- a/camera/ndk/NdkCaptureRequest.cpp
+++ b/camera/ndk/NdkCaptureRequest.cpp

@@ -142,3 +142,40 @@
     delete request;
     return;
 }
+
+EXPORT
+camera_status_t ACaptureRequest_setUserContext(
+        ACaptureRequest* request, void* context) {
+    if (request == nullptr) {
+        ALOGE("%s: invalid argument! request is NULL", __FUNCTION__);
+        return ACAMERA_ERROR_INVALID_PARAMETER;
+    }
+    return request->setContext(context);
+}
+
+EXPORT
+camera_status_t ACaptureRequest_getUserContext(
+        const ACaptureRequest* request, /*out*/void** context) {
+    if (request == nullptr || context == nullptr) {
+        ALOGE("%s: invalid argument! request %p, context %p",
+                __FUNCTION__, request, context);
+        return ACAMERA_ERROR_INVALID_PARAMETER;
+    }
+    return request->getContext(context);
+}
+
+EXPORT
+ACaptureRequest* ACaptureRequest_copy(const ACaptureRequest* src) {
+    ATRACE_CALL();
+    if (src == nullptr) {
+        ALOGE("%s: src is null!", __FUNCTION__);
+        return nullptr;
+    }
+
+    ACaptureRequest* pRequest = new ACaptureRequest();
+    pRequest->settings = new ACameraMetadata(*(src->settings));
+    pRequest->targets  = new ACameraOutputTargets();
+    *(pRequest->targets)  = *(src->targets);
+    pRequest->context = src->context;
+    return pRequest;
+}

diff --git a/camera/ndk/impl/ACameraCaptureSession.cpp b/camera/ndk/impl/ACameraCaptureSession.cpp
index b9c159d..f60e5fd 100644
--- a/camera/ndk/impl/ACameraCaptureSession.cpp
+++ b/camera/ndk/impl/ACameraCaptureSession.cpp

@@ -148,6 +148,23 @@
     return ret;
 }
 
+camera_status_t ACameraCaptureSession::updateOutputConfiguration(ACaptureSessionOutput *output) {
+    sp<CameraDevice> dev = getDeviceSp();
+    if (dev == nullptr) {
+        ALOGE("Error: Device associated with session %p has been closed!", this);
+        return ACAMERA_ERROR_SESSION_CLOSED;
+    }
+
+    camera_status_t ret;
+    dev->lockDeviceForSessionOps();
+    {
+        Mutex::Autolock _l(mSessionLock);
+        ret = dev->updateOutputConfigurationLocked(output);
+    }
+    dev->unlockDevice();
+    return ret;
+}
+
 ACameraDevice*
 ACameraCaptureSession::getDevice() {
     Mutex::Autolock _l(mSessionLock);

diff --git a/camera/ndk/impl/ACameraCaptureSession.h b/camera/ndk/impl/ACameraCaptureSession.h
index 339c665..a2068e7 100644
--- a/camera/ndk/impl/ACameraCaptureSession.h
+++ b/camera/ndk/impl/ACameraCaptureSession.h

@@ -24,7 +24,8 @@
 using namespace android;
 
 struct ACaptureSessionOutput {
-    explicit ACaptureSessionOutput(ANativeWindow* window) : mWindow(window) {};
+    explicit ACaptureSessionOutput(ANativeWindow* window, bool isShared = false) :
+            mWindow(window), mIsShared(isShared) {};
 
     bool operator == (const ACaptureSessionOutput& other) const {
         return mWindow == other.mWindow;
@@ -40,6 +41,8 @@
     }
 
     ANativeWindow* mWindow;
+    std::set<ANativeWindow *> mSharedWindows;
+    bool           mIsShared;
     int            mRotation = CAMERA3_STREAM_ROTATION_0;
 };
 
@@ -89,6 +92,8 @@
             int numRequests, ACaptureRequest** requests,
             /*optional*/int* captureSequenceId);
 
+    camera_status_t updateOutputConfiguration(ACaptureSessionOutput *output);
+
     ACameraDevice* getDevice();
 
   private:

diff --git a/camera/ndk/impl/ACameraDevice.cpp b/camera/ndk/impl/ACameraDevice.cpp
index af977b8..907debc 100644
--- a/camera/ndk/impl/ACameraDevice.cpp
+++ b/camera/ndk/impl/ACameraDevice.cpp

@@ -59,7 +59,8 @@
         mWrapper(wrapper),
         mInError(false),
         mError(ACAMERA_OK),
-        mIdle(true) {
+        mIdle(true),
+        mCurrentSession(nullptr) {
     mClosing = false;
     // Setup looper thread to perfrom device callbacks to app
     mCbLooper = new ALooper;
@@ -98,18 +99,30 @@
 
 // Device close implementaiton
 CameraDevice::~CameraDevice() {
-    Mutex::Autolock _l(mDeviceLock);
-    if (!isClosed()) {
-        disconnectLocked();
-    }
-    if (mCbLooper != nullptr) {
-        mCbLooper->unregisterHandler(mHandler->id());
-        mCbLooper->stop();
+    sp<ACameraCaptureSession> session = mCurrentSession.promote();
+    {
+        Mutex::Autolock _l(mDeviceLock);
+        if (!isClosed()) {
+            disconnectLocked(session);
+        }
+        mCurrentSession = nullptr;
+        if (mCbLooper != nullptr) {
+            mCbLooper->unregisterHandler(mHandler->id());
+            mCbLooper->stop();
+        }
     }
     mCbLooper.clear();
     mHandler.clear();
 }
 
+void
+CameraDevice::postSessionMsgAndCleanup(sp<AMessage>& msg) {
+    msg->post();
+    msg.clear();
+    sp<AMessage> cleanupMsg = new AMessage(kWhatCleanUpSessions, mHandler);
+    cleanupMsg->post();
+}
+
 // TODO: cached created request?
 camera_status_t
 CameraDevice::createCaptureRequest(
@@ -144,21 +157,23 @@
 camera_status_t
 CameraDevice::createCaptureSession(
         const ACaptureSessionOutputContainer*       outputs,
+        const ACaptureRequest* sessionParameters,
         const ACameraCaptureSession_stateCallbacks* callbacks,
         /*out*/ACameraCaptureSession** session) {
+    sp<ACameraCaptureSession> currentSession = mCurrentSession.promote();
     Mutex::Autolock _l(mDeviceLock);
     camera_status_t ret = checkCameraClosedOrErrorLocked();
     if (ret != ACAMERA_OK) {
         return ret;
     }
 
-    if (mCurrentSession != nullptr) {
-        mCurrentSession->closeByDevice();
+    if (currentSession != nullptr) {
+        currentSession->closeByDevice();
         stopRepeatingLocked();
     }
 
     // Create new session
-    ret = configureStreamsLocked(outputs);
+    ret = configureStreamsLocked(outputs, sessionParameters);
     if (ret != ACAMERA_OK) {
         ALOGE("Fail to create new session. cannot configure streams");
         return ret;
@@ -264,7 +279,7 @@
         msg->setPointer(kContextKey, session->mUserSessionCallback.context);
         msg->setObject(kSessionSpKey, session);
         msg->setPointer(kCallbackFpKey, (void*) session->mUserSessionCallback.onActive);
-        msg->post();
+        postSessionMsgAndCleanup(msg);
     }
     mIdle = false;
     mBusySession = session;
@@ -275,13 +290,93 @@
     return ACAMERA_OK;
 }
 
+camera_status_t CameraDevice::updateOutputConfigurationLocked(ACaptureSessionOutput *output) {
+    camera_status_t ret = checkCameraClosedOrErrorLocked();
+    if (ret != ACAMERA_OK) {
+        return ret;
+    }
+
+    if (output == nullptr) {
+        return ACAMERA_ERROR_INVALID_PARAMETER;
+    }
+
+    if (!output->mIsShared) {
+        ALOGE("Error output configuration is not shared");
+        return ACAMERA_ERROR_INVALID_OPERATION;
+    }
+
+    int32_t streamId = -1;
+    for (auto& kvPair : mConfiguredOutputs) {
+        if (kvPair.second.first == output->mWindow) {
+            streamId = kvPair.first;
+            break;
+        }
+    }
+    if (streamId < 0) {
+        ALOGE("Error: Invalid output configuration");
+        return ACAMERA_ERROR_INVALID_PARAMETER;
+    }
+
+    sp<IGraphicBufferProducer> iGBP(nullptr);
+    ret = getIGBPfromAnw(output->mWindow, iGBP);
+    if (ret != ACAMERA_OK) {
+        ALOGE("Camera device %s failed to extract graphic producer from native window",
+                getId());
+        return ret;
+    }
+
+    OutputConfiguration outConfig(iGBP, output->mRotation, OutputConfiguration::INVALID_SET_ID,
+            true);
+
+    for (auto& anw : output->mSharedWindows) {
+        ret = getIGBPfromAnw(anw, iGBP);
+        if (ret != ACAMERA_OK) {
+            ALOGE("Camera device %s failed to extract graphic producer from native window",
+                    getId());
+            return ret;
+        }
+        outConfig.addGraphicProducer(iGBP);
+    }
+
+    auto remoteRet = mRemote->updateOutputConfiguration(streamId, outConfig);
+    if (!remoteRet.isOk()) {
+        switch (remoteRet.serviceSpecificErrorCode()) {
+            case hardware::ICameraService::ERROR_INVALID_OPERATION:
+                ALOGE("Camera device %s invalid operation: %s", getId(),
+                        remoteRet.toString8().string());
+                return ACAMERA_ERROR_INVALID_OPERATION;
+                break;
+            case hardware::ICameraService::ERROR_ALREADY_EXISTS:
+                ALOGE("Camera device %s output surface already exists: %s", getId(),
+                        remoteRet.toString8().string());
+                return ACAMERA_ERROR_INVALID_PARAMETER;
+                break;
+            case hardware::ICameraService::ERROR_ILLEGAL_ARGUMENT:
+                ALOGE("Camera device %s invalid input argument: %s", getId(),
+                        remoteRet.toString8().string());
+                return ACAMERA_ERROR_INVALID_PARAMETER;
+                break;
+            default:
+                ALOGE("Camera device %s failed to add shared output: %s", getId(),
+                        remoteRet.toString8().string());
+                return ACAMERA_ERROR_UNKNOWN;
+        }
+    }
+    mConfiguredOutputs[streamId] = std::make_pair(output->mWindow, outConfig);
+
+    return ACAMERA_OK;
+}
+
 camera_status_t
 CameraDevice::allocateCaptureRequest(
         const ACaptureRequest* request, /*out*/sp<CaptureRequest>& outReq) {
     camera_status_t ret;
     sp<CaptureRequest> req(new CaptureRequest());
-    req->mMetadata = request->settings->getInternalData();
+    req->mPhysicalCameraSettings.push_back({std::string(mCameraId.string()),
+            request->settings->getInternalData()});
     req->mIsReprocess = false; // NDK does not support reprocessing yet
+    req->mContext = request->context;
+    req->mSurfaceConverted = true; // set to true, and fill in stream/surface idx to speed up IPC
 
     for (auto outputTarget : request->targets->mOutputs) {
         ANativeWindow* anw = outputTarget.mWindow;
@@ -292,7 +387,31 @@
             return ret;
         }
         req->mSurfaceList.push_back(surface);
+
+        bool found = false;
+        // lookup stream/surface ID
+        for (const auto& kvPair : mConfiguredOutputs) {
+            int streamId = kvPair.first;
+            const OutputConfiguration& outConfig = kvPair.second.second;
+            const auto& gbps = outConfig.getGraphicBufferProducers();
+            for (int surfaceId = 0; surfaceId < (int) gbps.size(); surfaceId++) {
+                if (gbps[surfaceId] == surface->getIGraphicBufferProducer()) {
+                    found = true;
+                    req->mStreamIdxList.push_back(streamId);
+                    req->mSurfaceIdxList.push_back(surfaceId);
+                    break;
+                }
+            }
+            if (found) {
+                break;
+            }
+        }
+        if (!found) {
+            ALOGE("Unconfigured output target %p in capture request!", anw);
+            return ret;
+        }
     }
+
     outReq = req;
     return ACAMERA_OK;
 }
@@ -300,7 +419,7 @@
 ACaptureRequest*
 CameraDevice::allocateACaptureRequest(sp<CaptureRequest>& req) {
     ACaptureRequest* pRequest = new ACaptureRequest();
-    CameraMetadata clone = req->mMetadata;
+    CameraMetadata clone = req->mPhysicalCameraSettings.begin()->settings;
     pRequest->settings = new ACameraMetadata(clone.release(), ACameraMetadata::ACM_REQUEST);
     pRequest->targets  = new ACameraOutputTargets();
     for (size_t i = 0; i < req->mSurfaceList.size(); i++) {
@@ -308,6 +427,7 @@
         ACameraOutputTarget outputTarget(anw);
         pRequest->targets->mOutputs.insert(outputTarget);
     }
+    pRequest->context = req->mContext;
     return pRequest;
 }
 
@@ -328,7 +448,7 @@
         return;
     }
 
-    if (session != mCurrentSession) {
+    if (mCurrentSession != session) {
         // Session has been replaced by other seesion or device is closed
         return;
     }
@@ -342,14 +462,14 @@
     }
 
     // No new session, unconfigure now
-    camera_status_t ret = configureStreamsLocked(nullptr);
+    camera_status_t ret = configureStreamsLocked(nullptr, nullptr);
     if (ret != ACAMERA_OK) {
         ALOGE("Unconfigure stream failed. Device might still be configured! ret %d", ret);
     }
 }
 
 void
-CameraDevice::disconnectLocked() {
+CameraDevice::disconnectLocked(sp<ACameraCaptureSession>& session) {
     if (mClosing.exchange(true)) {
         // Already closing, just return
         ALOGW("Camera device %s is already closing.", getId());
@@ -361,9 +481,8 @@
     }
     mRemote = nullptr;
 
-    if (mCurrentSession != nullptr) {
-        mCurrentSession->closeByDevice();
-        mCurrentSession = nullptr;
+    if (session != nullptr) {
+        session->closeByDevice();
     }
 }
 
@@ -404,7 +523,7 @@
     // This should never happen because creating a new session will close
     // previous one and thus reject any API call from previous session.
     // But still good to check here in case something unexpected happen.
-    if (session != mCurrentSession) {
+    if (mCurrentSession != session) {
         ALOGE("Camera %s session %p is not current active session!", getId(), session);
         return ACAMERA_ERROR_INVALID_OPERATION;
     }
@@ -415,12 +534,13 @@
     }
 
     mFlushing = true;
+
     // Send onActive callback to guarantee there is always active->ready transition
     sp<AMessage> msg = new AMessage(kWhatSessionStateCb, mHandler);
     msg->setPointer(kContextKey, session->mUserSessionCallback.context);
     msg->setObject(kSessionSpKey, session);
     msg->setPointer(kCallbackFpKey, (void*) session->mUserSessionCallback.onActive);
-    msg->post();
+    postSessionMsgAndCleanup(msg);
 
     // If device is already idling, send callback and exit early
     if (mIdle) {
@@ -428,7 +548,7 @@
         msg->setPointer(kContextKey, session->mUserSessionCallback.context);
         msg->setObject(kSessionSpKey, session);
         msg->setPointer(kCallbackFpKey, (void*) session->mUserSessionCallback.onReady);
-        msg->post();
+        postSessionMsgAndCleanup(msg);
         mFlushing = false;
         return ACAMERA_OK;
     }
@@ -472,17 +592,11 @@
 CameraDevice::getIGBPfromAnw(
         ANativeWindow* anw,
         sp<IGraphicBufferProducer>& out) {
-    if (anw == nullptr) {
-        ALOGE("Error: output ANativeWindow is null");
-        return ACAMERA_ERROR_INVALID_PARAMETER;
+    sp<Surface> surface;
+    camera_status_t ret = getSurfaceFromANativeWindow(anw, surface);
+    if (ret != ACAMERA_OK) {
+        return ret;
     }
-    int value;
-    int err = (*anw->query)(anw, NATIVE_WINDOW_CONCRETE_TYPE, &value);
-    if (err != OK || value != NATIVE_WINDOW_SURFACE) {
-        ALOGE("Error: ANativeWindow is not backed by Surface!");
-        return ACAMERA_ERROR_INVALID_PARAMETER;
-    }
-    const sp<Surface> surface(static_cast<Surface*>(anw));
     out = surface->getIGraphicBufferProducer();
     return ACAMERA_OK;
 }
@@ -506,7 +620,8 @@
 }
 
 camera_status_t
-CameraDevice::configureStreamsLocked(const ACaptureSessionOutputContainer* outputs) {
+CameraDevice::configureStreamsLocked(const ACaptureSessionOutputContainer* outputs,
+        const ACaptureRequest* sessionParameters) {
     ACaptureSessionOutputContainer emptyOutput;
     if (outputs == nullptr) {
         outputs = &emptyOutput;
@@ -526,7 +641,8 @@
             return ret;
         }
         outputSet.insert(std::make_pair(
-                anw, OutputConfiguration(iGBP, outConfig.mRotation)));
+                anw, OutputConfiguration(iGBP, outConfig.mRotation,
+                        OutputConfiguration::INVALID_SET_ID, outConfig.mIsShared)));
     }
     auto addSet = outputSet;
     std::vector<int> deleteList;
@@ -568,7 +684,7 @@
         msg->setObject(kSessionSpKey, mBusySession);
         msg->setPointer(kCallbackFpKey, (void*) mBusySession->mUserSessionCallback.onReady);
         mBusySession.clear();
-        msg->post();
+        postSessionMsgAndCleanup(msg);
     }
     mIdle = true;
 
@@ -601,7 +717,11 @@
         mConfiguredOutputs.insert(std::make_pair(streamId, outputPair));
     }
 
-    remoteRet = mRemote->endConfigure(/*isConstrainedHighSpeed*/ false);
+    CameraMetadata params;
+    if ((sessionParameters != nullptr) && (sessionParameters->settings != nullptr)) {
+        params.append(sessionParameters->settings->getInternalData());
+    }
+    remoteRet = mRemote->endConfigure(/*isConstrainedHighSpeed*/ false, params);
     if (remoteRet.serviceSpecificErrorCode() == hardware::ICameraService::ERROR_ILLEGAL_ARGUMENT) {
         ALOGE("Camera device %s cannnot support app output configuration: %s", getId(),
                 remoteRet.toString8().string());
@@ -716,19 +836,26 @@
             setCameraDeviceErrorLocked(ACAMERA_ERROR_CAMERA_SERVICE);
             return;
         }
-        ANativeWindow* anw = outputPairIt->second.first;
 
-        ALOGV("Camera %s Lost output buffer for ANW %p frame %" PRId64,
-                getId(), anw, frameNumber);
+        const auto& gbps = outputPairIt->second.second.getGraphicBufferProducers();
+        for (const auto& outGbp : gbps) {
+            for (auto surface : request->mSurfaceList) {
+                if (surface->getIGraphicBufferProducer() == outGbp) {
+                    ANativeWindow* anw = static_cast<ANativeWindow*>(surface.get());
+                    ALOGV("Camera %s Lost output buffer for ANW %p frame %" PRId64,
+                            getId(), anw, frameNumber);
 
-        sp<AMessage> msg = new AMessage(kWhatCaptureBufferLost, mHandler);
-        msg->setPointer(kContextKey, cbh.mCallbacks.context);
-        msg->setObject(kSessionSpKey, session);
-        msg->setPointer(kCallbackFpKey, (void*) onBufferLost);
-        msg->setObject(kCaptureRequestKey, request);
-        msg->setPointer(kAnwKey, (void*) anw);
-        msg->setInt64(kFrameNumberKey, frameNumber);
-        msg->post();
+                    sp<AMessage> msg = new AMessage(kWhatCaptureBufferLost, mHandler);
+                    msg->setPointer(kContextKey, cbh.mCallbacks.context);
+                    msg->setObject(kSessionSpKey, session);
+                    msg->setPointer(kCallbackFpKey, (void*) onBufferLost);
+                    msg->setObject(kCaptureRequestKey, request);
+                    msg->setPointer(kAnwKey, (void*) anw);
+                    msg->setInt64(kFrameNumberKey, frameNumber);
+                    postSessionMsgAndCleanup(msg);
+                }
+            }
+        }
     } else { // Handle other capture failures
         // Fire capture failure callback if there is one registered
         ACameraCaptureSession_captureCallback_failed onError = cbh.mCallbacks.onCaptureFailed;
@@ -746,7 +873,7 @@
         msg->setPointer(kCallbackFpKey, (void*) onError);
         msg->setObject(kCaptureRequestKey, request);
         msg->setObject(kCaptureFailureKey, failure);
-        msg->post();
+        postSessionMsgAndCleanup(msg);
 
         // Update tracker
         mFrameNumberTracker.updateTracker(frameNumber, /*isError*/true);
@@ -769,6 +896,9 @@
         case kWhatCaptureBufferLost:
             ALOGV("%s: Received msg %d", __FUNCTION__, msg->what());
             break;
+        case kWhatCleanUpSessions:
+            mCachedSessions.clear();
+            return;
         default:
             ALOGE("%s:Error: unknown device callback %d", __FUNCTION__, msg->what());
             return;
@@ -842,6 +972,7 @@
                 return;
             }
             sp<ACameraCaptureSession> session(static_cast<ACameraCaptureSession*>(obj.get()));
+            mCachedSessions.push(session);
             sp<CaptureRequest> requestSp = nullptr;
             switch (msg->what()) {
                 case kWhatCaptureStart:
@@ -1053,7 +1184,7 @@
         msg->setObject(kSessionSpKey, cbh.mSession);
         msg->setPointer(kCallbackFpKey, (void*) cbh.mCallbacks.onCaptureSequenceAborted);
         msg->setInt32(kSequenceIdKey, sequenceId);
-        msg->post();
+        postSessionMsgAndCleanup(msg);
     } else {
         // Use mSequenceLastFrameNumberMap to track
         mSequenceLastFrameNumberMap.insert(std::make_pair(sequenceId, lastFrameNumber));
@@ -1110,7 +1241,7 @@
             // before cbh goes out of scope and causing we call the session
             // destructor while holding device lock
             cbh.mSession.clear();
-            msg->post();
+            postSessionMsgAndCleanup(msg);
         }
 
         // No need to track sequence complete if there is no callback registered
@@ -1137,6 +1268,7 @@
         return ret; // device has been closed
     }
 
+    sp<ACameraCaptureSession> session = dev->mCurrentSession.promote();
     Mutex::Autolock _l(dev->mDeviceLock);
     if (dev->mRemote == nullptr) {
         return ret; // device has been closed
@@ -1145,10 +1277,10 @@
         case ERROR_CAMERA_DISCONNECTED:
         {
             // Camera is disconnected, close the session and expect no more callbacks
-            if (dev->mCurrentSession != nullptr) {
-                dev->mCurrentSession->closeByDevice();
-                dev->mCurrentSession = nullptr;
+            if (session != nullptr) {
+                session->closeByDevice();
             }
+            dev->mCurrentSession = nullptr;
             sp<AMessage> msg = new AMessage(kWhatOnDisconnected, dev->mHandler);
             msg->setPointer(kContextKey, dev->mAppCallbacks.context);
             msg->setPointer(kDeviceKey, (void*) dev->getWrapper());
@@ -1216,6 +1348,7 @@
             dev->setCameraDeviceErrorLocked(ACAMERA_ERROR_CAMERA_DEVICE);
             return ret;
         }
+
         sp<AMessage> msg = new AMessage(kWhatSessionStateCb, dev->mHandler);
         msg->setPointer(kContextKey, dev->mBusySession->mUserSessionCallback.context);
         msg->setObject(kSessionSpKey, dev->mBusySession);
@@ -1223,7 +1356,7 @@
         // Make sure we clear the sp first so the session destructor can
         // only happen on handler thread (where we don't hold device/session lock)
         dev->mBusySession.clear();
-        msg->post();
+        dev->postSessionMsgAndCleanup(msg);
     }
     dev->mIdle = true;
     dev->mFlushing = false;
@@ -1265,7 +1398,7 @@
         msg->setPointer(kCallbackFpKey, (void*) onStart);
         msg->setObject(kCaptureRequestKey, request);
         msg->setInt64(kTimeStampKey, timestamp);
-        msg->post();
+        dev->postSessionMsgAndCleanup(msg);
     }
     return ret;
 }
@@ -1273,7 +1406,9 @@
 binder::Status
 CameraDevice::ServiceCallback::onResultReceived(
         const CameraMetadata& metadata,
-        const CaptureResultExtras& resultExtras) {
+        const CaptureResultExtras& resultExtras,
+        const std::vector<PhysicalCaptureResultInfo>& physicalResultInfos) {
+    (void) physicalResultInfos;
     binder::Status ret = binder::Status::ok();
 
     sp<CameraDevice> dev = mDevice.promote();
@@ -1328,7 +1463,7 @@
         msg->setPointer(kCallbackFpKey, (void*) onResult);
         msg->setObject(kCaptureRequestKey, request);
         msg->setObject(kCaptureResultKey, result);
-        msg->post();
+        dev->postSessionMsgAndCleanup(msg);
     }
 
     if (!isPartialResult) {

diff --git a/camera/ndk/impl/ACameraDevice.h b/camera/ndk/impl/ACameraDevice.h
index 855efe1..1369148 100644
--- a/camera/ndk/impl/ACameraDevice.h
+++ b/camera/ndk/impl/ACameraDevice.h

@@ -36,12 +36,13 @@
 #include <camera/camera2/OutputConfiguration.h>
 #include <camera/camera2/CaptureRequest.h>
 
-#include <camera/NdkCameraDevice.h>
+#include <camera/NdkCameraManager.h>
+#include <camera/NdkCameraCaptureSession.h>
 #include "ACameraMetadata.h"
 
 namespace android {
 
-// Wrap ACameraCaptureFailure so it can be ref-counter
+// Wrap ACameraCaptureFailure so it can be ref-counted
 struct CameraCaptureFailure : public RefBase, public ACameraCaptureFailure {};
 
 class CameraDevice final : public RefBase {
@@ -59,6 +60,7 @@
 
     camera_status_t createCaptureSession(
             const ACaptureSessionOutputContainer*       outputs,
+            const ACaptureRequest* sessionParameters,
             const ACameraCaptureSession_stateCallbacks* callbacks,
             /*out*/ACameraCaptureSession** session);
 
@@ -72,7 +74,8 @@
         binder::Status onCaptureStarted(const CaptureResultExtras& resultExtras,
                               int64_t timestamp) override;
         binder::Status onResultReceived(const CameraMetadata& metadata,
-                              const CaptureResultExtras& resultExtras) override;
+                              const CaptureResultExtras& resultExtras,
+                              const std::vector<PhysicalCaptureResultInfo>& physicalResultInfos) override;
         binder::Status onPrepared(int streamId) override;
         binder::Status onRequestQueueEmpty() override;
         binder::Status onRepeatingRequestError(int64_t lastFrameNumber,
@@ -96,7 +99,7 @@
     // device goes into fatal error state after this
     void setCameraDeviceErrorLocked(camera_status_t error);
 
-    void disconnectLocked(); // disconnect from camera service
+    void disconnectLocked(sp<ACameraCaptureSession>& session); // disconnect from camera service
 
     camera_status_t stopRepeatingLocked();
 
@@ -122,7 +125,9 @@
             /*out*/int* captureSequenceId,
             bool isRepeating);
 
-    static camera_status_t allocateCaptureRequest(
+    camera_status_t updateOutputConfigurationLocked(ACaptureSessionOutput *output);
+
+    camera_status_t allocateCaptureRequest(
             const ACaptureRequest* request, sp<CaptureRequest>& outReq);
 
     static ACaptureRequest* allocateACaptureRequest(sp<CaptureRequest>& req);
@@ -136,7 +141,11 @@
     // For capture session to notify its end of life
     void notifySessionEndOfLifeLocked(ACameraCaptureSession* session);
 
-    camera_status_t configureStreamsLocked(const ACaptureSessionOutputContainer* outputs);
+    camera_status_t configureStreamsLocked(const ACaptureSessionOutputContainer* outputs,
+           const ACaptureRequest* sessionParameters);
+
+    // Input message will be posted and cleared after this returns
+    void postSessionMsgAndCleanup(sp<AMessage>& msg);
 
     static camera_status_t getIGBPfromAnw(
             ANativeWindow* anw, sp<IGraphicBufferProducer>& out);
@@ -185,7 +194,9 @@
         kWhatCaptureFail,      // onCaptureFailed
         kWhatCaptureSeqEnd,    // onCaptureSequenceCompleted
         kWhatCaptureSeqAbort,  // onCaptureSequenceAborted
-        kWhatCaptureBufferLost // onCaptureBufferLost
+        kWhatCaptureBufferLost,// onCaptureBufferLost
+        // Internal cleanup
+        kWhatCleanUpSessions   // Cleanup cached sp<ACameraCaptureSession>
     };
     static const char* kContextKey;
     static const char* kDeviceKey;
@@ -199,10 +210,16 @@
     static const char* kSequenceIdKey;
     static const char* kFrameNumberKey;
     static const char* kAnwKey;
+
     class CallbackHandler : public AHandler {
       public:
-        CallbackHandler() {}
         void onMessageReceived(const sp<AMessage> &msg) override;
+
+      private:
+        // This handler will cache all capture session sp until kWhatCleanUpSessions
+        // is processed. This is used to guarantee the last session reference is always
+        // being removed in callback thread without holding camera device lock
+        Vector<sp<ACameraCaptureSession>> mCachedSessions;
     };
     sp<CallbackHandler> mHandler;
 
@@ -210,7 +227,7 @@
      * Capture session related members *
      ***********************************/
     // The current active session
-    ACameraCaptureSession* mCurrentSession = nullptr;
+    wp<ACameraCaptureSession> mCurrentSession;
     bool mFlushing = false;
 
     int mNextSessionId = 0;
@@ -295,9 +312,10 @@
 
     camera_status_t createCaptureSession(
             const ACaptureSessionOutputContainer*       outputs,
+            const ACaptureRequest* sessionParameters,
             const ACameraCaptureSession_stateCallbacks* callbacks,
             /*out*/ACameraCaptureSession** session) {
-        return mDevice->createCaptureSession(outputs, callbacks, session);
+        return mDevice->createCaptureSession(outputs, sessionParameters, callbacks, session);
     }
 
     /***********************

diff --git a/camera/ndk/impl/ACameraManager.cpp b/camera/ndk/impl/ACameraManager.cpp
index 3f64bcc..c59d0e7 100644
--- a/camera/ndk/impl/ACameraManager.cpp
+++ b/camera/ndk/impl/ACameraManager.cpp

@@ -221,7 +221,7 @@
     mCallbacks.erase(cb);
 }
 
-void CameraManagerGlobal::getCameraIdList(std::vector<String8> *cameraIds) {
+void CameraManagerGlobal::getCameraIdList(std::vector<String8>* cameraIds) {
     // Ensure that we have initialized/refreshed the list of available devices
     auto cs = getCameraService();
     Mutex::Autolock _l(mLock);
@@ -340,6 +340,9 @@
         msg->setString(kCameraIdKey, AString(cameraId));
         msg->post();
     }
+    if (status == hardware::ICameraServiceListener::STATUS_NOT_PRESENT) {
+        mDeviceStatusMap.erase(cameraId);
+    }
 }
 
 } // namespace android

diff --git a/camera/ndk/impl/ACameraManager.h b/camera/ndk/impl/ACameraManager.h
index 4a172f3..cc42f77 100644
--- a/camera/ndk/impl/ACameraManager.h
+++ b/camera/ndk/impl/ACameraManager.h

@@ -19,6 +19,7 @@
 
 #include <camera/NdkCameraManager.h>
 
+#include <android-base/parseint.h>
 #include <android/hardware/ICameraService.h>
 #include <android/hardware/BnCameraServiceListener.h>
 #include <camera/CameraMetadata.h>
@@ -140,8 +141,29 @@
     static bool validStatus(int32_t status);
     static bool isStatusAvailable(int32_t status);
 
+    // The sort logic must match the logic in
+    // libcameraservice/common/CameraProviderManager.cpp::getAPI1CompatibleCameraDeviceIds
+    struct CameraIdComparator {
+        bool operator()(const String8& a, const String8& b) const {
+            uint32_t aUint = 0, bUint = 0;
+            bool aIsUint = base::ParseUint(a.c_str(), &aUint);
+            bool bIsUint = base::ParseUint(b.c_str(), &bUint);
+
+            // Uint device IDs first
+            if (aIsUint && bIsUint) {
+                return aUint < bUint;
+            } else if (aIsUint) {
+                return true;
+            } else if (bIsUint) {
+                return false;
+            }
+            // Simple string compare if both id are not uint
+            return a < b;
+        }
+    };
+
     // Map camera_id -> status
-    std::map<String8, int32_t> mDeviceStatusMap;
+    std::map<String8, int32_t, CameraIdComparator> mDeviceStatusMap;
 
     // For the singleton instance
     static Mutex sLock;

diff --git a/camera/ndk/impl/ACameraMetadata.cpp b/camera/ndk/impl/ACameraMetadata.cpp
index 7b33c32..fc00a2d 100644
--- a/camera/ndk/impl/ACameraMetadata.cpp
+++ b/camera/ndk/impl/ACameraMetadata.cpp

@@ -235,7 +235,7 @@
 }
 
 const CameraMetadata&
-ACameraMetadata::getInternalData() {
+ACameraMetadata::getInternalData() const {
     return mData;
 }
 
@@ -305,6 +305,7 @@
         case ACAMERA_STATISTICS_FACE_DETECT_MODE:
         case ACAMERA_STATISTICS_HOT_PIXEL_MAP_MODE:
         case ACAMERA_STATISTICS_LENS_SHADING_MAP_MODE:
+        case ACAMERA_STATISTICS_OIS_DATA_MODE:
         case ACAMERA_TONEMAP_CURVE_BLUE:
         case ACAMERA_TONEMAP_CURVE_GREEN:
         case ACAMERA_TONEMAP_CURVE_RED:
@@ -312,6 +313,7 @@
         case ACAMERA_TONEMAP_GAMMA:
         case ACAMERA_TONEMAP_PRESET_CURVE:
         case ACAMERA_BLACK_LEVEL_LOCK:
+        case ACAMERA_DISTORTION_CORRECTION_MODE:
             return true;
         default:
             return false;

diff --git a/camera/ndk/impl/ACameraMetadata.h b/camera/ndk/impl/ACameraMetadata.h
index 143efc7..0fd7efa 100644
--- a/camera/ndk/impl/ACameraMetadata.h
+++ b/camera/ndk/impl/ACameraMetadata.h

@@ -64,7 +64,7 @@
     void filterUnsupportedFeatures(); // Hide features not yet supported by NDK
     void filterStreamConfigurations(); // Hide input streams, translate hal format to NDK formats
 
-    const CameraMetadata& getInternalData();
+    const CameraMetadata& getInternalData() const;
 
     template<typename INTERNAL_T, typename NDK_T>
     camera_status_t updateImpl(uint32_t tag, uint32_t count, const NDK_T* data) {

diff --git a/camera/ndk/impl/ACaptureRequest.h b/camera/ndk/impl/ACaptureRequest.h
index e5b453e..06b2cc3 100644
--- a/camera/ndk/impl/ACaptureRequest.h
+++ b/camera/ndk/impl/ACaptureRequest.h

@@ -45,8 +45,19 @@
 };
 
 struct ACaptureRequest {
+    camera_status_t setContext(void* ctx) {
+        context = ctx;
+        return ACAMERA_OK;
+    }
+
+    camera_status_t getContext(void** ctx) const {
+        *ctx = context;
+        return ACAMERA_OK;
+    }
+
     ACameraMetadata*      settings;
     ACameraOutputTargets* targets;
+    void*                 context;
 };
 
 #endif // _ACAPTURE_REQUEST_H

diff --git a/camera/ndk/include/camera/NdkCameraCaptureSession.h b/camera/ndk/include/camera/NdkCameraCaptureSession.h
index d96f538..51cef8c 100644
--- a/camera/ndk/include/camera/NdkCameraCaptureSession.h
+++ b/camera/ndk/include/camera/NdkCameraCaptureSession.h

@@ -33,6 +33,7 @@
  * Do not #include files that aren't part of the NDK.
  */
 #include <sys/cdefs.h>
+#include <stdbool.h>
 
 #include <android/native_window.h>
 #include "NdkCameraError.h"
@@ -591,6 +592,54 @@
 
 #endif /* __ANDROID_API__ >= 24 */
 
+#if __ANDROID_API__ >= 28
+
+typedef struct ACaptureSessionOutput ACaptureSessionOutput;
+
+/**
+ * Update shared ACaptureSessionOutput.
+ *
+ * <p>A shared ACaptureSessionOutput (see {@link ACaptureSessionSharedOutput_create}) that
+ * was modified via calls to {@link ACaptureSessionSharedOutput_add} or
+ * {@link ACaptureSessionSharedOutput_remove} must be updated by calling this method before its
+ * changes take effect. After the update call returns  with {@link ACAMERA_OK}, any newly added
+ * native windows can be used as a target in subsequent capture requests.</p>
+ *
+ * <p>Native windows that get removed must not be part of any active repeating or single/burst
+ * request or have any pending results. Consider updating repeating requests via
+ * {@link ACaptureSessionOutput_setRepeatingRequest} and then wait for the last frame number
+ * when the sequence completes
+ * {@link ACameraCaptureSession_captureCallback#onCaptureSequenceCompleted}.</p>
+ *
+ * <p>Native windows that get added must not be part of any other registered ACaptureSessionOutput
+ * and must be compatible. Compatible windows must have matching format, rotation and
+ * consumer usage.</p>
+ *
+ * <p>A shared ACameraCaptureSession can support up to 4 additional native windows.</p>
+ *
+ * @param session the capture session of interest
+ * @param output the modified output configuration
+ *
+ * @return <ul><li>
+ *             {@link ACAMERA_OK} if the method succeeds.</li>
+ *         <li>{@link ACAMERA_ERROR_INVALID_PARAMETER} if session or output is NULL; or output
+ *             contains invalid native windows; or if an attempt was made to add
+ *             a native window to a different output configuration; or new native window is not
+ *             compatible; or any removed native window still has pending requests;</li>
+ *         <li>{@link ACAMERA_ERROR_INVALID_OPERATION} if output configuration is not shared (see
+ *             {@link ACaptureSessionSharedOutput_create};  or the number of additional
+ *             native windows goes beyond the supported limit.</li>
+ *         <li>{@link ACAMERA_ERROR_SESSION_CLOSED} if the capture session has been closed</li>
+ *         <li>{@link ACAMERA_ERROR_CAMERA_DISCONNECTED} if the camera device is closed</li>
+ *         <li>{@link ACAMERA_ERROR_CAMERA_DEVICE} if the camera device encounters fatal error</li>
+ *         <li>{@link ACAMERA_ERROR_CAMERA_SERVICE} if the camera service encounters fatal
+ *             error</li>
+ *         <li>{@link ACAMERA_ERROR_UNKNOWN} if the method fails for some other reasons</li></ul>
+ */
+camera_status_t ACameraCaptureSession_updateSharedOutput(ACameraCaptureSession* session,
+        ACaptureSessionOutput* output);
+#endif /* __ANDROID_API__ >= 28 */
+
 __END_DECLS
 
 #endif /* _NDK_CAMERA_CAPTURE_SESSION_H */

diff --git a/camera/ndk/include/camera/NdkCameraDevice.h b/camera/ndk/include/camera/NdkCameraDevice.h
index 9b7f6f4..92dad1c 100644
--- a/camera/ndk/include/camera/NdkCameraDevice.h
+++ b/camera/ndk/include/camera/NdkCameraDevice.h

@@ -90,18 +90,18 @@
 };
 
 /**
- * Camera device state callbacks to be used in {@link ACameraDevice_stateCallbacks}.
+ * Camera device state callbacks to be used in {@link ACameraDevice_StateCallbacks}.
  *
- * @param context The optional context in {@link ACameraDevice_stateCallbacks} will be
+ * @param context The optional context in {@link ACameraDevice_StateCallbacks} will be
  *                passed to this callback.
  * @param device The {@link ACameraDevice} that is being disconnected.
  */
 typedef void (*ACameraDevice_StateCallback)(void* context, ACameraDevice* device);
 
 /**
- * Camera device error state callbacks to be used in {@link ACameraDevice_stateCallbacks}.
+ * Camera device error state callbacks to be used in {@link ACameraDevice_StateCallbacks}.
  *
- * @param context The optional context in {@link ACameraDevice_stateCallbacks} will be
+ * @param context The optional context in {@link ACameraDevice_StateCallbacks} will be
  *                passed to this callback.
  * @param device The {@link ACameraDevice} that is being disconnected.
  * @param error The error code describes the cause of this error callback. See the folowing
@@ -150,7 +150,12 @@
      *
      */
     ACameraDevice_ErrorStateCallback  onError;
-} ACameraDevice_stateCallbacks;
+} ACameraDevice_StateCallbacks;
+
+/**
+ * For backward compatiblity.
+ */
+typedef ACameraDevice_StateCallbacks ACameraDevice_stateCallbacks;
 
 /**
  * Close the connection and free this ACameraDevice synchronously. Access to the ACameraDevice
@@ -251,6 +256,7 @@
      * @see ACameraDevice_createCaptureRequest
      */
     TEMPLATE_MANUAL = 6,
+
 } ACameraDevice_request_template;
 
 /**
@@ -637,6 +643,14 @@
  * target combinations with sizes outside of these guarantees, but this can only be tested for
  * by attempting to create a session with such targets.</p>
  *
+ * <p>Exception on 176x144 (QCIF) resolution:
+ * Camera devices usually have a fixed capability for downscaling from larger resolution to
+ * smaller, and the QCIF resolution sometimes cannot be fully supported due to this
+ * limitation on devices with high-resolution image sensors. Therefore, trying to configure a
+ * QCIF resolution stream together with any other stream larger than 1920x1080 resolution
+ * (either width or height) might not be supported, and capture session creation will fail if it
+ * is not.</p>
+ *
  * @param device the camera device of interest.
  * @param outputs the {@link ACaptureSessionOutputContainer} describes all output streams.
  * @param callbacks the {@link ACameraCaptureSession_stateCallbacks capture session state callbacks}.
@@ -661,9 +675,102 @@
 
 #endif /* __ANDROID_API__ >= 24 */
 
+#if __ANDROID_API__ >= 28
+
+/**
+ * Create a shared ACaptureSessionOutput object.
+ *
+ * <p>The ACaptureSessionOutput is used in {@link ACaptureSessionOutputContainer_add} method to add
+ * an output {@link ANativeWindow} to ACaptureSessionOutputContainer. Use
+ * {@link ACaptureSessionOutput_free} to free the object and its memory after application no longer
+ * needs the {@link ACaptureSessionOutput}. A shared ACaptureSessionOutput can be further modified
+ * via {@link ACaptureSessionSharedOutput_add} or {@link ACaptureSessionSharedOutput_remove} and
+ * must be updated via {@link ACameraCaptureSession_updateSharedOutput}.</p>
+ *
+ * @param anw the {@link ANativeWindow} to be associated with the {@link ACaptureSessionOutput}
+ * @param output the output {@link ACaptureSessionOutput} will be stored here if the
+ *                  method call succeeds.
+ *
+ * @return <ul>
+ *         <li>{@link ACAMERA_OK} if the method call succeeds. The created container will be
+ *                                filled in the output argument.</li>
+ *         <li>{@link ACAMERA_ERROR_INVALID_PARAMETER} if anw or output is NULL.</li></ul>
+ *
+ * @see ACaptureSessionOutputContainer_add
+ */
+camera_status_t ACaptureSessionSharedOutput_create(
+        ANativeWindow* anw, /*out*/ACaptureSessionOutput** output);
+
+/**
+ * Add a native window to shared ACaptureSessionOutput.
+ *
+ * The ACaptureSessionOutput must be created via {@link ACaptureSessionSharedOutput_create}.
+ *
+ * @param output  the shared ACaptureSessionOutput to be extended.
+ * @param anw The new native window.
+ *
+ * @return <ul>
+ *         <li>{@link ACAMERA_OK} if the method call succeeds.</li>
+ *         <li>{@link ACAMERA_ERROR_INVALID_PARAMETER} if anw or output is NULL; or output is not
+ *             shared see {@link ACaptureSessionSharedOutput_create}; or anw matches with the native
+ *             window associated with ACaptureSessionOutput; or anw is already present inside
+ *             ACaptureSessionOutput.</li></ul>
+ */
+camera_status_t ACaptureSessionSharedOutput_add(ACaptureSessionOutput *output, ANativeWindow *anw);
+
+/**
+ * Remove a native window from shared ACaptureSessionOutput.
+ *
+ * @param output the {@link ACaptureSessionOutput} to be modified.
+ * @param anw The native window to be removed.
+ *
+ * @return <ul>
+ *         <li>{@link ACAMERA_OK} if the method call succeeds.</li>
+ *         <li>{@link ACAMERA_ERROR_INVALID_PARAMETER} if anw or output is NULL; or output is not
+ *             shared see {@link ACaptureSessionSharedOutput_create}; or anw matches with the native
+ *             window associated with ACaptureSessionOutput; or anw is not present inside
+ *             ACaptureSessionOutput.</li></ul>
+ */
+camera_status_t ACaptureSessionSharedOutput_remove(ACaptureSessionOutput *output,
+        ANativeWindow* anw);
+
+/**
+ * Create a new camera capture session similar to {@link ACameraDevice_createCaptureSession}. This
+ * function allows clients to pass additional session parameters during session initialization. For
+ * further information about session parameters see {@link ACAMERA_REQUEST_AVAILABLE_SESSION_KEYS}.
+ *
+ * @param device the camera device of interest.
+ * @param outputs the {@link ACaptureSessionOutputContainer} describes all output streams.
+ * @param sessionParameters An optional capture request that contains the initial values of session
+ *                          parameters advertised in
+ *                          {@link ACAMERA_REQUEST_AVAILABLE_SESSION_KEYS}.
+ * @param callbacks the {@link ACameraCaptureSession_stateCallbacks}
+ *                  capture session state callbacks.
+ * @param session the created {@link ACameraCaptureSession} will be filled here if the method call
+ *                succeeds.
+ *
+ * @return <ul>
+ *         <li>{@link ACAMERA_OK} if the method call succeeds. The created capture session will be
+ *                                filled in session argument.</li>
+ *         <li>{@link ACAMERA_ERROR_INVALID_PARAMETER} if any of device, outputs, callbacks or
+ *                                session is NULL.</li>
+ *         <li>{@link ACAMERA_ERROR_CAMERA_DISCONNECTED} if the camera device is closed.</li>
+ *         <li>{@link ACAMERA_ERROR_CAMERA_DEVICE} if the camera device encounters fatal error.</li>
+ *         <li>{@link ACAMERA_ERROR_CAMERA_SERVICE} if the camera service encounters fatal error.
+ *         </li>
+ *         <li>{@link ACAMERA_ERROR_UNKNOWN} if the method fails for some other reasons.</li></ul>
+ */
+camera_status_t ACameraDevice_createCaptureSessionWithSessionParameters(
+        ACameraDevice* device,
+        const ACaptureSessionOutputContainer* outputs,
+        const ACaptureRequest* sessionParameters,
+        const ACameraCaptureSession_stateCallbacks* callbacks,
+        /*out*/ACameraCaptureSession** session);
+
+#endif /* __ANDROID_API__ >= 28 */
+
 __END_DECLS
 
 #endif /* _NDK_CAMERA_DEVICE_H */
 
 /** @} */
-

diff --git a/camera/ndk/include/camera/NdkCameraManager.h b/camera/ndk/include/camera/NdkCameraManager.h
index 5b5c98b..e5b3ad8 100644
--- a/camera/ndk/include/camera/NdkCameraManager.h
+++ b/camera/ndk/include/camera/NdkCameraManager.h

@@ -232,18 +232,18 @@
  * priority when accessing the camera, and this method will succeed even if the camera device is
  * in use by another camera API client. Any lower-priority application that loses control of the
  * camera in this way will receive an
- * {@link ACameraDevice_stateCallbacks#onDisconnected} callback.</p>
+ * {@link ACameraDevice_StateCallbacks#onDisconnected} callback.</p>
  *
  * <p>Once the camera is successfully opened,the ACameraDevice can then be set up
  * for operation by calling {@link ACameraDevice_createCaptureSession} and
  * {@link ACameraDevice_createCaptureRequest}.</p>
  *
  * <p>If the camera becomes disconnected after this function call returns,
- * {@link ACameraDevice_stateCallbacks#onDisconnected} with a
+ * {@link ACameraDevice_StateCallbacks#onDisconnected} with a
  * ACameraDevice in the disconnected state will be called.</p>
  *
  * <p>If the camera runs into error after this function call returns,
- * {@link ACameraDevice_stateCallbacks#onError} with a
+ * {@link ACameraDevice_StateCallbacks#onError} with a
  * ACameraDevice in the error state will be called.</p>
  *
  * @param manager the {@link ACameraManager} of interest.

diff --git a/camera/ndk/include/camera/NdkCameraMetadataTags.h b/camera/ndk/include/camera/NdkCameraMetadataTags.h
index 629d75a..bee1a46 100644
--- a/camera/ndk/include/camera/NdkCameraMetadataTags.h
+++ b/camera/ndk/include/camera/NdkCameraMetadataTags.h

@@ -69,6 +69,8 @@
     ACAMERA_SYNC,
     ACAMERA_REPROCESS,
     ACAMERA_DEPTH,
+    ACAMERA_LOGICAL_MULTI_CAMERA,
+    ACAMERA_DISTORTION_CORRECTION,
     ACAMERA_SECTION_COUNT,
 
     ACAMERA_VENDOR = 0x8000
@@ -104,6 +106,12 @@
     ACAMERA_SYNC_START             = ACAMERA_SYNC              << 16,
     ACAMERA_REPROCESS_START        = ACAMERA_REPROCESS         << 16,
     ACAMERA_DEPTH_START            = ACAMERA_DEPTH             << 16,
+    ACAMERA_LOGICAL_MULTI_CAMERA_START
+                                   = ACAMERA_LOGICAL_MULTI_CAMERA
+                                                                << 16,
+    ACAMERA_DISTORTION_CORRECTION_START
+                                   = ACAMERA_DISTORTION_CORRECTION
+                                                                << 16,
     ACAMERA_VENDOR_START           = ACAMERA_VENDOR            << 16
 } acamera_metadata_section_start_t;
 
@@ -471,15 +479,26 @@
      * Otherwise will always be present.</p>
      * <p>The maximum number of regions supported by the device is determined by the value
      * of android.control.maxRegionsAe.</p>
-     * <p>The data representation is int[5 * area_count].
-     * Every five elements represent a metering region of (xmin, ymin, xmax, ymax, weight).
-     * The rectangle is defined to be inclusive on xmin and ymin, but exclusive on xmax and
-     * ymax.</p>
-     * <p>The coordinate system is based on the active pixel array,
-     * with (0,0) being the top-left pixel in the active pixel array, and
+     * <p>For devices not supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system always follows that of ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with (0,0) being
+     * the top-left pixel in the active pixel array, and
      * (ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.width - 1,
-     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.height - 1) being the
-     * bottom-right pixel in the active pixel array.</p>
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.height - 1) being the bottom-right pixel in the
+     * active pixel array.</p>
+     * <p>For devices supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system depends on the mode being set.
+     * When the distortion correction mode is OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE, with
+     * <code>(0, 0)</code> being the top-left pixel of the pre-correction active array, and
+     * (ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE.width - 1,
+     * ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE.height - 1) being the bottom-right
+     * pixel in the pre-correction active pixel array.
+     * When the distortion correction mode is not OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with
+     * <code>(0, 0)</code> being the top-left pixel of the active array, and
+     * (ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.width - 1,
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.height - 1) being the bottom-right pixel in the
+     * active pixel array.</p>
      * <p>The weight must be within <code>[0, 1000]</code>, and represents a weight
      * for every pixel in the area. This means that a large metering area
      * with the same weight as a smaller area will have more effect in
@@ -495,9 +514,15 @@
      * region and output only the intersection rectangle as the metering region in the result
      * metadata.  If the region is entirely outside the crop region, it will be ignored and
      * not reported in the result metadata.</p>
+     * <p>The data representation is <code>int[5 * area_count]</code>.
+     * Every five elements represent a metering region of <code>(xmin, ymin, xmax, ymax, weight)</code>.
+     * The rectangle is defined to be inclusive on xmin and ymin, but exclusive on xmax and
+     * ymax.</p>
      *
+     * @see ACAMERA_DISTORTION_CORRECTION_MODE
      * @see ACAMERA_SCALER_CROP_REGION
      * @see ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE
+     * @see ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE
      */
     ACAMERA_CONTROL_AE_REGIONS =                                // int32[5*area_count]
             ACAMERA_CONTROL_START + 4,
@@ -633,15 +658,26 @@
      * Otherwise will always be present.</p>
      * <p>The maximum number of focus areas supported by the device is determined by the value
      * of android.control.maxRegionsAf.</p>
-     * <p>The data representation is int[5 * area_count].
-     * Every five elements represent a metering region of (xmin, ymin, xmax, ymax, weight).
-     * The rectangle is defined to be inclusive on xmin and ymin, but exclusive on xmax and
-     * ymax.</p>
-     * <p>The coordinate system is based on the active pixel array,
-     * with (0,0) being the top-left pixel in the active pixel array, and
+     * <p>For devices not supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system always follows that of ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with (0,0) being
+     * the top-left pixel in the active pixel array, and
      * (ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.width - 1,
-     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.height - 1) being the
-     * bottom-right pixel in the active pixel array.</p>
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.height - 1) being the bottom-right pixel in the
+     * active pixel array.</p>
+     * <p>For devices supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system depends on the mode being set.
+     * When the distortion correction mode is OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE, with
+     * <code>(0, 0)</code> being the top-left pixel of the pre-correction active array, and
+     * (ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE.width - 1,
+     * ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE.height - 1) being the bottom-right
+     * pixel in the pre-correction active pixel array.
+     * When the distortion correction mode is not OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with
+     * <code>(0, 0)</code> being the top-left pixel of the active array, and
+     * (ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.width - 1,
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.height - 1) being the bottom-right pixel in the
+     * active pixel array.</p>
      * <p>The weight must be within <code>[0, 1000]</code>, and represents a weight
      * for every pixel in the area. This means that a large metering area
      * with the same weight as a smaller area will have more effect in
@@ -651,15 +687,22 @@
      * is used, all non-zero weights will have the same effect. A region with 0 weight is
      * ignored.</p>
      * <p>If all regions have 0 weight, then no specific metering area needs to be used by the
-     * camera device.</p>
+     * camera device. The capture result will either be a zero weight region as well, or
+     * the region selected by the camera device as the focus area of interest.</p>
      * <p>If the metering region is outside the used ACAMERA_SCALER_CROP_REGION returned in
      * capture result metadata, the camera device will ignore the sections outside the crop
      * region and output only the intersection rectangle as the metering region in the result
      * metadata. If the region is entirely outside the crop region, it will be ignored and
      * not reported in the result metadata.</p>
+     * <p>The data representation is <code>int[5 * area_count]</code>.
+     * Every five elements represent a metering region of <code>(xmin, ymin, xmax, ymax, weight)</code>.
+     * The rectangle is defined to be inclusive on xmin and ymin, but exclusive on xmax and
+     * ymax.</p>
      *
+     * @see ACAMERA_DISTORTION_CORRECTION_MODE
      * @see ACAMERA_SCALER_CROP_REGION
      * @see ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE
+     * @see ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE
      */
     ACAMERA_CONTROL_AF_REGIONS =                                // int32[5*area_count]
             ACAMERA_CONTROL_START + 8,
@@ -791,15 +834,26 @@
      * Otherwise will always be present.</p>
      * <p>The maximum number of regions supported by the device is determined by the value
      * of android.control.maxRegionsAwb.</p>
-     * <p>The data representation is int[5 * area_count].
-     * Every five elements represent a metering region of (xmin, ymin, xmax, ymax, weight).
-     * The rectangle is defined to be inclusive on xmin and ymin, but exclusive on xmax and
-     * ymax.</p>
-     * <p>The coordinate system is based on the active pixel array,
-     * with (0,0) being the top-left pixel in the active pixel array, and
+     * <p>For devices not supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system always follows that of ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with (0,0) being
+     * the top-left pixel in the active pixel array, and
      * (ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.width - 1,
-     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.height - 1) being the
-     * bottom-right pixel in the active pixel array.</p>
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.height - 1) being the bottom-right pixel in the
+     * active pixel array.</p>
+     * <p>For devices supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system depends on the mode being set.
+     * When the distortion correction mode is OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE, with
+     * <code>(0, 0)</code> being the top-left pixel of the pre-correction active array, and
+     * (ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE.width - 1,
+     * ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE.height - 1) being the bottom-right
+     * pixel in the pre-correction active pixel array.
+     * When the distortion correction mode is not OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with
+     * <code>(0, 0)</code> being the top-left pixel of the active array, and
+     * (ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.width - 1,
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.height - 1) being the bottom-right pixel in the
+     * active pixel array.</p>
      * <p>The weight must range from 0 to 1000, and represents a weight
      * for every pixel in the area. This means that a large metering area
      * with the same weight as a smaller area will have more effect in
@@ -815,9 +869,15 @@
      * region and output only the intersection rectangle as the metering region in the result
      * metadata.  If the region is entirely outside the crop region, it will be ignored and
      * not reported in the result metadata.</p>
+     * <p>The data representation is <code>int[5 * area_count]</code>.
+     * Every five elements represent a metering region of <code>(xmin, ymin, xmax, ymax, weight)</code>.
+     * The rectangle is defined to be inclusive on xmin and ymin, but exclusive on xmax and
+     * ymax.</p>
      *
+     * @see ACAMERA_DISTORTION_CORRECTION_MODE
      * @see ACAMERA_SCALER_CROP_REGION
      * @see ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE
+     * @see ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE
      */
     ACAMERA_CONTROL_AWB_REGIONS =                               // int32[5*area_count]
             ACAMERA_CONTROL_START + 12,
@@ -837,10 +897,13 @@
      *
      * <p>This control (except for MANUAL) is only effective if
      * <code>ACAMERA_CONTROL_MODE != OFF</code> and any 3A routine is active.</p>
-     * <p>ZERO_SHUTTER_LAG will be supported if ACAMERA_REQUEST_AVAILABLE_CAPABILITIES
-     * contains PRIVATE_REPROCESSING or YUV_REPROCESSING. MANUAL will be supported if
-     * ACAMERA_REQUEST_AVAILABLE_CAPABILITIES contains MANUAL_SENSOR. Other intent values are
-     * always supported.</p>
+     * <p>All intents are supported by all devices, except that:
+     *   * ZERO_SHUTTER_LAG will be supported if ACAMERA_REQUEST_AVAILABLE_CAPABILITIES contains
+     * PRIVATE_REPROCESSING or YUV_REPROCESSING.
+     *   * MANUAL will be supported if ACAMERA_REQUEST_AVAILABLE_CAPABILITIES contains
+     * MANUAL_SENSOR.
+     *   * MOTION_TRACKING will be supported if ACAMERA_REQUEST_AVAILABLE_CAPABILITIES contains
+     * MOTION_TRACKING.</p>
      *
      * @see ACAMERA_CONTROL_MODE
      * @see ACAMERA_REQUEST_AVAILABLE_CAPABILITIES
@@ -885,11 +948,10 @@
      * <p>When set to AUTO, the individual algorithm controls in
      * ACAMERA_CONTROL_* are in effect, such as ACAMERA_CONTROL_AF_MODE.</p>
      * <p>When set to USE_SCENE_MODE, the individual controls in
-     * ACAMERA_CONTROL_* are mostly disabled, and the camera device implements
-     * one of the scene mode settings (such as ACTION, SUNSET, or PARTY)
-     * as it wishes. The camera device scene mode 3A settings are provided by
-     * capture results {@link ACameraMetadata} from
-     * {@link ACameraCaptureSession_captureCallback_result}.</p>
+     * ACAMERA_CONTROL_* are mostly disabled, and the camera device
+     * implements one of the scene mode settings (such as ACTION,
+     * SUNSET, or PARTY) as it wishes. The camera device scene mode
+     * 3A settings are provided by {@link ACameraCaptureSession_captureCallback_result capture results}.</p>
      * <p>When set to OFF_KEEP_STATE, it is similar to OFF mode, the only difference
      * is that this frame will not be used by camera device background 3A statistics
      * update, as if this frame is never captured. This mode can be used in the scenario
@@ -1043,20 +1105,18 @@
      * <p>For constant-framerate recording, for each normal
      * <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html">CamcorderProfile</a>, that is, a
      * <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html">CamcorderProfile</a> that has
-     * <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html#quality">quality</a>
-     * in the range [
-     * <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html#QUALITY_LOW">QUALITY_LOW</a>,
-     * <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html#QUALITY_2160P">QUALITY_2160P</a>],
-     * if the profile is supported by the device and has
-     * <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html#videoFrameRate">videoFrameRate</a>
-     * <code>x</code>, this list will always include (<code>x</code>,<code>x</code>).</p>
+     * <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html#quality">quality</a> in
+     * the range [<a href="https://developer.android.com/reference/android/media/CamcorderProfile.html#QUALITY_LOW">QUALITY_LOW</a>,
+     * <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html#QUALITY_2160P">QUALITY_2160P</a>], if the profile is
+     * supported by the device and has
+     * <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html#videoFrameRate">videoFrameRate</a> <code>x</code>, this list will
+     * always include (<code>x</code>,<code>x</code>).</p>
      * </li>
      * <li>
      * <p>Also, a camera device must either not support any
      * <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html">CamcorderProfile</a>,
      * or support at least one
-     * normal <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html">CamcorderProfile</a>
-     * that has
+     * normal <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html">CamcorderProfile</a> that has
      * <a href="https://developer.android.com/reference/android/media/CamcorderProfile.html#videoFrameRate">videoFrameRate</a> <code>x</code> &gt;= 24.</p>
      * </li>
      * </ul>
@@ -1282,7 +1342,7 @@
      * <p>State       | Transition Cause | New State | Notes
      * :------------:|:----------------:|:---------:|:-----------------------:
      * INACTIVE      |                  | INACTIVE  | Camera device auto exposure algorithm is disabled</p>
-     * <p>When ACAMERA_CONTROL_AE_MODE is AE_MODE_ON_*:</p>
+     * <p>When ACAMERA_CONTROL_AE_MODE is AE_MODE_ON*:</p>
      * <p>State        | Transition Cause                             | New State      | Notes
      * :-------------:|:--------------------------------------------:|:--------------:|:-----------------:
      * INACTIVE       | Camera device initiates AE scan              | SEARCHING      | Values changing
@@ -1303,10 +1363,13 @@
      * LOCKED         | aeLock is ON and aePrecaptureTrigger is CANCEL| LOCKED        | Precapture trigger is ignored when AE is already locked
      * Any state (excluding LOCKED) | ACAMERA_CONTROL_AE_PRECAPTURE_TRIGGER is START | PRECAPTURE     | Start AE precapture metering sequence
      * Any state (excluding LOCKED) | ACAMERA_CONTROL_AE_PRECAPTURE_TRIGGER is CANCEL| INACTIVE       | Currently active precapture metering sequence is canceled</p>
+     * <p>If the camera device supports AE external flash mode (ON_EXTERNAL_FLASH is included in
+     * ACAMERA_CONTROL_AE_AVAILABLE_MODES), ACAMERA_CONTROL_AE_STATE must be FLASH_REQUIRED after
+     * the camera device finishes AE scan and it's too dark without flash.</p>
      * <p>For the above table, the camera device may skip reporting any state changes that happen
      * without application intervention (i.e. mode switch, trigger, locking). Any state that
      * can be skipped in that manner is called a transient state.</p>
-     * <p>For example, for above AE modes (AE_MODE_ON_*), in addition to the state transitions
+     * <p>For example, for above AE modes (AE_MODE_ON*), in addition to the state transitions
      * listed in above table, it is also legal for the camera device to skip one or more
      * transient states between two results. See below table for examples:</p>
      * <p>State        | Transition Cause                                            | New State      | Notes
@@ -1319,9 +1382,11 @@
      * CONVERGED      | Camera device finished AE scan                              | FLASH_REQUIRED | Converged but too dark w/o flash after a new scan, transient states are skipped by camera device.
      * FLASH_REQUIRED | Camera device finished AE scan                              | CONVERGED      | Converged after a new scan, transient states are skipped by camera device.</p>
      *
+     * @see ACAMERA_CONTROL_AE_AVAILABLE_MODES
      * @see ACAMERA_CONTROL_AE_LOCK
      * @see ACAMERA_CONTROL_AE_MODE
      * @see ACAMERA_CONTROL_AE_PRECAPTURE_TRIGGER
+     * @see ACAMERA_CONTROL_AE_STATE
      * @see ACAMERA_CONTROL_MODE
      * @see ACAMERA_CONTROL_SCENE_MODE
      */
@@ -1619,13 +1684,13 @@
      * compared to previous regular requests. enableZsl does not affect requests with other
      * capture intents.</p>
      * <p>For example, when requests are submitted in the following order:
-     *   Request A: enableZsl is <code>true</code>, ACAMERA_CONTROL_CAPTURE_INTENT is PREVIEW
-     *   Request B: enableZsl is <code>true</code>, ACAMERA_CONTROL_CAPTURE_INTENT is STILL_CAPTURE</p>
+     *   Request A: enableZsl is ON, ACAMERA_CONTROL_CAPTURE_INTENT is PREVIEW
+     *   Request B: enableZsl is ON, ACAMERA_CONTROL_CAPTURE_INTENT is STILL_CAPTURE</p>
      * <p>The output images for request B may have contents captured before the output images for
      * request A, and the result metadata for request B may be older than the result metadata for
      * request A.</p>
-     * <p>Note that when enableZsl is <code>true</code>, it is not guaranteed to get output images captured in the
-     * past for requests with STILL_CAPTURE capture intent.</p>
+     * <p>Note that when enableZsl is <code>true</code>, it is not guaranteed to get output images captured in
+     * the past for requests with STILL_CAPTURE capture intent.</p>
      * <p>For applications targeting SDK versions O and newer, the value of enableZsl in
      * TEMPLATE_STILL_CAPTURE template may be <code>true</code>. The value in other templates is always
      * <code>false</code> if present.</p>
@@ -1638,6 +1703,26 @@
      */
     ACAMERA_CONTROL_ENABLE_ZSL =                                // byte (acamera_metadata_enum_android_control_enable_zsl_t)
             ACAMERA_CONTROL_START + 41,
+    /**
+     * <p>Whether a significant scene change is detected within the currently-set AF
+     * region(s).</p>
+     *
+     * <p>Type: byte (acamera_metadata_enum_android_control_af_scene_change_t)</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraCaptureSession_captureCallback_result callbacks</li>
+     * </ul></p>
+     *
+     * <p>When the camera focus routine detects a change in the scene it is looking at,
+     * such as a large shift in camera viewpoint, significant motion in the scene, or a
+     * significant illumination change, this value will be set to DETECTED for a single capture
+     * result. Otherwise the value will be NOT_DETECTED. The threshold for detection is similar
+     * to what would trigger a new passive focus scan to begin in CONTINUOUS autofocus modes.</p>
+     * <p>This key will be available if the camera device advertises this key via {@link ACAMERA_REQUEST_AVAILABLE_RESULT_KEYS }.</p>
+     */
+    ACAMERA_CONTROL_AF_SCENE_CHANGE =                           // byte (acamera_metadata_enum_android_control_af_scene_change_t)
+            ACAMERA_CONTROL_START + 42,
     ACAMERA_CONTROL_END,
 
     /**
@@ -1879,8 +1964,8 @@
      * the thumbnail data will also be rotated.</p>
      * <p>Note that this orientation is relative to the orientation of the camera sensor, given
      * by ACAMERA_SENSOR_ORIENTATION.</p>
-     * <p>To translate from the device orientation given by the Android sensor APIs, the following
-     * sample code may be used:</p>
+     * <p>To translate from the device orientation given by the Android sensor APIs for camera
+     * sensors which are not EXTERNAL, the following sample code may be used:</p>
      * <pre><code>private int getJpegOrientation(CameraCharacteristics c, int deviceOrientation) {
      *     if (deviceOrientation == android.view.OrientationEventListener.ORIENTATION_UNKNOWN) return 0;
      *     int sensorOrientation = c.get(CameraCharacteristics.SENSOR_ORIENTATION);
@@ -1899,6 +1984,8 @@
      *     return jpegOrientation;
      * }
      * </code></pre>
+     * <p>For EXTERNAL cameras the sensor orientation will always be set to 0 and the facing will
+     * also be set to EXTERNAL. The above code is not relevant in such case.</p>
      *
      * @see ACAMERA_SENSOR_ORIENTATION
      */
@@ -1959,14 +2046,14 @@
      * <p>When an ACAMERA_JPEG_ORIENTATION of non-zero degree is requested,
      * the camera device will handle thumbnail rotation in one of the following ways:</p>
      * <ul>
-     * <li>Set the
-     *   <a href="https://developer.android.com/reference/android/media/ExifInterface.html#TAG_ORIENTATION">EXIF orientation flag</a>
+     * <li>Set the <a href="https://developer.android.com/reference/android/media/ExifInterface.html#TAG_ORIENTATION">EXIF orientation flag</a>
      *   and keep jpeg and thumbnail image data unrotated.</li>
      * <li>Rotate the jpeg and thumbnail image data and not set
-     *   <a href="https://developer.android.com/reference/android/media/ExifInterface.html#TAG_ORIENTATION">EXIF orientation flag</a>.
-     *   In this case, LIMITED or FULL hardware level devices will report rotated thumnail size
-     *   in capture result, so the width and height will be interchanged if 90 or 270 degree
-     *   orientation is requested. LEGACY device will always report unrotated thumbnail size.</li>
+     *   <a href="https://developer.android.com/reference/android/media/ExifInterface.html#TAG_ORIENTATION">EXIF orientation flag</a>. In this
+     *   case, LIMITED or FULL hardware level devices will report rotated thumnail size in
+     *   capture result, so the width and height will be interchanged if 90 or 270 degree
+     *   orientation is requested. LEGACY device will always report unrotated thumbnail
+     *   size.</li>
      * </ul>
      *
      * @see ACAMERA_JPEG_ORIENTATION
@@ -2216,37 +2303,36 @@
      * </ul></p>
      *
      * <p>The position of the camera device's lens optical center,
-     * as a three-dimensional vector <code>(x,y,z)</code>, relative to the
-     * optical center of the largest camera device facing in the
-     * same direction as this camera, in the
-     * <a href="https://developer.android.com/reference/android/hardware/SensorEvent.html">Android sensor coordinate axes</a>.
-     * Note that only the axis definitions are shared with
-     * the sensor coordinate system, but not the origin.</p>
-     * <p>If this device is the largest or only camera device with a
-     * given facing, then this position will be <code>(0, 0, 0)</code>; a
-     * camera device with a lens optical center located 3 cm from
-     * the main sensor along the +X axis (to the right from the
-     * user's perspective) will report <code>(0.03, 0, 0)</code>.</p>
-     * <p>To transform a pixel coordinates between two cameras
-     * facing the same direction, first the source camera
-     * ACAMERA_LENS_RADIAL_DISTORTION must be corrected for.  Then
-     * the source camera ACAMERA_LENS_INTRINSIC_CALIBRATION needs
-     * to be applied, followed by the ACAMERA_LENS_POSE_ROTATION
-     * of the source camera, the translation of the source camera
-     * relative to the destination camera, the
-     * ACAMERA_LENS_POSE_ROTATION of the destination camera, and
-     * finally the inverse of ACAMERA_LENS_INTRINSIC_CALIBRATION
-     * of the destination camera. This obtains a
-     * radial-distortion-free coordinate in the destination
-     * camera pixel coordinates.</p>
-     * <p>To compare this against a real image from the destination
-     * camera, the destination camera image then needs to be
-     * corrected for radial distortion before comparison or
-     * sampling.</p>
+     * as a three-dimensional vector <code>(x,y,z)</code>.</p>
+     * <p>Prior to Android P, or when ACAMERA_LENS_POSE_REFERENCE is PRIMARY_CAMERA, this position
+     * is relative to the optical center of the largest camera device facing in the same
+     * direction as this camera, in the <a href="https://developer.android.com/reference/android/hardware/SensorEvent.html">Android sensor
+     * coordinate axes</a>. Note that only the axis definitions are shared with the sensor
+     * coordinate system, but not the origin.</p>
+     * <p>If this device is the largest or only camera device with a given facing, then this
+     * position will be <code>(0, 0, 0)</code>; a camera device with a lens optical center located 3 cm
+     * from the main sensor along the +X axis (to the right from the user's perspective) will
+     * report <code>(0.03, 0, 0)</code>.  Note that this means that, for many computer vision
+     * applications, the position needs to be negated to convert it to a translation from the
+     * camera to the origin.</p>
+     * <p>To transform a pixel coordinates between two cameras facing the same direction, first
+     * the source camera ACAMERA_LENS_DISTORTION must be corrected for.  Then the source
+     * camera ACAMERA_LENS_INTRINSIC_CALIBRATION needs to be applied, followed by the
+     * ACAMERA_LENS_POSE_ROTATION of the source camera, the translation of the source camera
+     * relative to the destination camera, the ACAMERA_LENS_POSE_ROTATION of the destination
+     * camera, and finally the inverse of ACAMERA_LENS_INTRINSIC_CALIBRATION of the destination
+     * camera. This obtains a radial-distortion-free coordinate in the destination camera pixel
+     * coordinates.</p>
+     * <p>To compare this against a real image from the destination camera, the destination camera
+     * image then needs to be corrected for radial distortion before comparison or sampling.</p>
+     * <p>When ACAMERA_LENS_POSE_REFERENCE is GYROSCOPE, then this position is relative to
+     * the center of the primary gyroscope on the device. The axis definitions are the same as
+     * with PRIMARY_CAMERA.</p>
      *
+     * @see ACAMERA_LENS_DISTORTION
      * @see ACAMERA_LENS_INTRINSIC_CALIBRATION
+     * @see ACAMERA_LENS_POSE_REFERENCE
      * @see ACAMERA_LENS_POSE_ROTATION
-     * @see ACAMERA_LENS_RADIAL_DISTORTION
      */
     ACAMERA_LENS_POSE_TRANSLATION =                             // float[3]
             ACAMERA_LENS_START + 7,
@@ -2335,13 +2421,15 @@
      * </code></pre>
      * <p>which can then be combined with the camera pose rotation
      * <code>R</code> and translation <code>t</code> (ACAMERA_LENS_POSE_ROTATION and
-     * ACAMERA_LENS_POSE_TRANSLATION, respective) to calculate the
+     * ACAMERA_LENS_POSE_TRANSLATION, respectively) to calculate the
      * complete transform from world coordinates to pixel
      * coordinates:</p>
-     * <pre><code>P = [ K 0   * [ R t
-     *      0 1 ]     0 1 ]
+     * <pre><code>P = [ K 0   * [ R -Rt
+     *      0 1 ]      0 1 ]
      * </code></pre>
-     * <p>and with <code>p_w</code> being a point in the world coordinate system
+     * <p>(Note the negation of poseTranslation when mapping from camera
+     * to world coordinates, and multiplication by the rotation).</p>
+     * <p>With <code>p_w</code> being a point in the world coordinate system
      * and <code>p_s</code> being a point in the camera active pixel array
      * coordinate system, and with the mapping including the
      * homogeneous division by z:</p>
@@ -2356,27 +2444,57 @@
      * where <code>(0,0)</code> is the top-left of the
      * preCorrectionActiveArraySize rectangle. Once the pose and
      * intrinsic calibration transforms have been applied to a
-     * world point, then the ACAMERA_LENS_RADIAL_DISTORTION
+     * world point, then the ACAMERA_LENS_DISTORTION
      * transform needs to be applied, and the result adjusted to
      * be in the ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE coordinate
      * system (where <code>(0, 0)</code> is the top-left of the
      * activeArraySize rectangle), to determine the final pixel
      * coordinate of the world point for processed (non-RAW)
      * output buffers.</p>
+     * <p>For camera devices, the center of pixel <code>(x,y)</code> is located at
+     * coordinate <code>(x + 0.5, y + 0.5)</code>.  So on a device with a
+     * precorrection active array of size <code>(10,10)</code>, the valid pixel
+     * indices go from <code>(0,0)-(9,9)</code>, and an perfectly-built camera would
+     * have an optical center at the exact center of the pixel grid, at
+     * coordinates <code>(5.0, 5.0)</code>, which is the top-left corner of pixel
+     * <code>(5,5)</code>.</p>
      *
+     * @see ACAMERA_LENS_DISTORTION
      * @see ACAMERA_LENS_POSE_ROTATION
      * @see ACAMERA_LENS_POSE_TRANSLATION
-     * @see ACAMERA_LENS_RADIAL_DISTORTION
      * @see ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE
      * @see ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE
      */
     ACAMERA_LENS_INTRINSIC_CALIBRATION =                        // float[5]
             ACAMERA_LENS_START + 10,
+    ACAMERA_LENS_RADIAL_DISTORTION =                            // Deprecated! DO NOT USE
+            ACAMERA_LENS_START + 11,
+    /**
+     * <p>The origin for ACAMERA_LENS_POSE_TRANSLATION.</p>
+     *
+     * @see ACAMERA_LENS_POSE_TRANSLATION
+     *
+     * <p>Type: byte (acamera_metadata_enum_android_lens_pose_reference_t)</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraManager_getCameraCharacteristics</li>
+     * </ul></p>
+     *
+     * <p>Different calibration methods and use cases can produce better or worse results
+     * depending on the selected coordinate origin.</p>
+     */
+    ACAMERA_LENS_POSE_REFERENCE =                               // byte (acamera_metadata_enum_android_lens_pose_reference_t)
+            ACAMERA_LENS_START + 12,
     /**
      * <p>The correction coefficients to correct for this camera device's
      * radial and tangential lens distortion.</p>
+     * <p>Replaces the deprecated ACAMERA_LENS_RADIAL_DISTORTION field, which was
+     * inconsistently defined.</p>
      *
-     * <p>Type: float[6]</p>
+     * @see ACAMERA_LENS_RADIAL_DISTORTION
+     *
+     * <p>Type: float[5]</p>
      *
      * <p>This tag may appear in:
      * <ul>
@@ -2384,13 +2502,13 @@
      *   <li>ACameraMetadata from ACameraCaptureSession_captureCallback_result callbacks</li>
      * </ul></p>
      *
-     * <p>Four radial distortion coefficients <code>[kappa_0, kappa_1, kappa_2,
+     * <p>Three radial distortion coefficients <code>[kappa_1, kappa_2,
      * kappa_3]</code> and two tangential distortion coefficients
      * <code>[kappa_4, kappa_5]</code> that can be used to correct the
      * lens's geometric distortion with the mapping equations:</p>
-     * <pre><code> x_c = x_i * ( kappa_0 + kappa_1 * r^2 + kappa_2 * r^4 + kappa_3 * r^6 ) +
+     * <pre><code> x_c = x_i * ( 1 + kappa_1 * r^2 + kappa_2 * r^4 + kappa_3 * r^6 ) +
      *        kappa_4 * (2 * x_i * y_i) + kappa_5 * ( r^2 + 2 * x_i^2 )
-     *  y_c = y_i * ( kappa_0 + kappa_1 * r^2 + kappa_2 * r^4 + kappa_3 * r^6 ) +
+     *  y_c = y_i * ( 1 + kappa_1 * r^2 + kappa_2 * r^4 + kappa_3 * r^6 ) +
      *        kappa_5 * (2 * x_i * y_i) + kappa_4 * ( r^2 + 2 * y_i^2 )
      * </code></pre>
      * <p>Here, <code>[x_c, y_c]</code> are the coordinates to sample in the
@@ -2398,23 +2516,21 @@
      * corrected image at the coordinate <code>[x_i, y_i]</code>:</p>
      * <pre><code> correctedImage(x_i, y_i) = sample_at(x_c, y_c, inputImage)
      * </code></pre>
-     * <p>The pixel coordinates are defined in a normalized
-     * coordinate system related to the
-     * ACAMERA_LENS_INTRINSIC_CALIBRATION calibration fields.
-     * Both <code>[x_i, y_i]</code> and <code>[x_c, y_c]</code> have <code>(0,0)</code> at the
-     * lens optical center <code>[c_x, c_y]</code>. The maximum magnitudes
-     * of both x and y coordinates are normalized to be 1 at the
-     * edge further from the optical center, so the range
-     * for both dimensions is <code>-1 &lt;= x &lt;= 1</code>.</p>
+     * <p>The pixel coordinates are defined in a coordinate system
+     * related to the ACAMERA_LENS_INTRINSIC_CALIBRATION
+     * calibration fields; see that entry for details of the mapping stages.
+     * Both <code>[x_i, y_i]</code> and <code>[x_c, y_c]</code>
+     * have <code>(0,0)</code> at the lens optical center <code>[c_x, c_y]</code>, and
+     * the range of the coordinates depends on the focal length
+     * terms of the intrinsic calibration.</p>
      * <p>Finally, <code>r</code> represents the radial distance from the
-     * optical center, <code>r^2 = x_i^2 + y_i^2</code>, and its magnitude
-     * is therefore no larger than <code>|r| &lt;= sqrt(2)</code>.</p>
+     * optical center, <code>r^2 = x_i^2 + y_i^2</code>.</p>
      * <p>The distortion model used is the Brown-Conrady model.</p>
      *
      * @see ACAMERA_LENS_INTRINSIC_CALIBRATION
      */
-    ACAMERA_LENS_RADIAL_DISTORTION =                            // float[6]
-            ACAMERA_LENS_START + 11,
+    ACAMERA_LENS_DISTORTION =                                   // float[5]
+            ACAMERA_LENS_START + 13,
     ACAMERA_LENS_END,
 
     /**
@@ -2662,11 +2778,12 @@
      * into the 3 stream types as below:</p>
      * <ul>
      * <li>Processed (but stalling): any non-RAW format with a stallDurations &gt; 0.
-     *   Typically {@link AIMAGE_FORMAT_JPEG} format.</li>
-     * <li>Raw formats: {@link AIMAGE_FORMAT_RAW16}, {@link AIMAGE_FORMAT_RAW10}, or
-     *   {@link AIMAGE_FORMAT_RAW12}.</li>
-     * <li>Processed (but not-stalling): any non-RAW format without a stall duration.
-     *   Typically {@link AIMAGE_FORMAT_YUV_420_888}.</li>
+     *   Typically {@link AIMAGE_FORMAT_JPEG JPEG format}.</li>
+     * <li>Raw formats: {@link AIMAGE_FORMAT_RAW16 RAW_SENSOR}, {@link AIMAGE_FORMAT_RAW10 RAW10}, or
+     *   {@link AIMAGE_FORMAT_RAW12 RAW12}.</li>
+     * <li>Processed (but not-stalling): any non-RAW format without a stall duration.  Typically
+     *   {@link AIMAGE_FORMAT_YUV_420_888 YUV_420_888},
+     *   <a href="https://developer.android.com/reference/android/graphics/ImageFormat.html#NV21">NV21</a>, or <a href="https://developer.android.com/reference/android/graphics/ImageFormat.html#YV12">YV12</a>.</li>
      * </ul>
      *
      * @see ACAMERA_SCALER_AVAILABLE_STREAM_CONFIGURATIONS
@@ -2787,7 +2904,7 @@
             ACAMERA_REQUEST_START + 12,
     /**
      * <p>A list of all keys that the camera device has available
-     * to use with {@link ACaptureRequest}.</p>
+     * to use with {@link ACaptureRequest }.</p>
      *
      * <p>Type: int32[n]</p>
      *
@@ -2809,9 +2926,7 @@
     ACAMERA_REQUEST_AVAILABLE_REQUEST_KEYS =                    // int32[n]
             ACAMERA_REQUEST_START + 13,
     /**
-     * <p>A list of all keys that the camera device has available
-     * to query with {@link ACameraMetadata} from
-     * {@link ACameraCaptureSession_captureCallback_result}.</p>
+     * <p>A list of all keys that the camera device has available to use with {@link ACameraCaptureSession_captureCallback_result }.</p>
      *
      * <p>Type: int32[n]</p>
      *
@@ -2842,9 +2957,7 @@
     ACAMERA_REQUEST_AVAILABLE_RESULT_KEYS =                     // int32[n]
             ACAMERA_REQUEST_START + 14,
     /**
-     * <p>A list of all keys that the camera device has available
-     * to query with {@link ACameraMetadata} from
-     * {@link ACameraManager_getCameraCharacteristics}.</p>
+     * <p>A list of all keys that the camera device has available to use with {@link ACameraManager_getCameraCharacteristics }.</p>
      *
      * <p>Type: int32[n]</p>
      *
@@ -2862,6 +2975,59 @@
      */
     ACAMERA_REQUEST_AVAILABLE_CHARACTERISTICS_KEYS =            // int32[n]
             ACAMERA_REQUEST_START + 15,
+    /**
+     * <p>A subset of the available request keys that the camera device
+     * can pass as part of the capture session initialization.</p>
+     *
+     * <p>Type: int32[n]</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraManager_getCameraCharacteristics</li>
+     * </ul></p>
+     *
+     * <p>This is a subset of ACAMERA_REQUEST_AVAILABLE_REQUEST_KEYS which
+     * contains a list of keys that are difficult to apply per-frame and
+     * can result in unexpected delays when modified during the capture session
+     * lifetime. Typical examples include parameters that require a
+     * time-consuming hardware re-configuration or internal camera pipeline
+     * change. For performance reasons we advise clients to pass their initial
+     * values as part of
+     * {@link ACameraDevice_createCaptureSessionWithSessionParameters }.
+     * Once the camera capture session is enabled it is also recommended to avoid
+     * changing them from their initial values set in
+     * {@link ACameraDevice_createCaptureSessionWithSessionParameters }.
+     * Control over session parameters can still be exerted in capture requests
+     * but clients should be aware and expect delays during their application.
+     * An example usage scenario could look like this:</p>
+     * <ul>
+     * <li>The camera client starts by quering the session parameter key list via
+     *   {@link ACameraManager_getCameraCharacteristics }.</li>
+     * <li>Before triggering the capture session create sequence, a capture request
+     *   must be built via
+     *   {@link ACameraDevice_createCaptureRequest }
+     *   using an appropriate template matching the particular use case.</li>
+     * <li>The client should go over the list of session parameters and check
+     *   whether some of the keys listed matches with the parameters that
+     *   they intend to modify as part of the first capture request.</li>
+     * <li>If there is no such match, the capture request can be  passed
+     *   unmodified to
+     *   {@link ACameraDevice_createCaptureSessionWithSessionParameters }.</li>
+     * <li>If matches do exist, the client should update the respective values
+     *   and pass the request to
+     *   {@link ACameraDevice_createCaptureSessionWithSessionParameters }.</li>
+     * <li>After the capture session initialization completes the session parameter
+     *   key list can continue to serve as reference when posting or updating
+     *   further requests. As mentioned above further changes to session
+     *   parameters should ideally be avoided, if updates are necessary
+     *   however clients could expect a delay/glitch during the
+     *   parameter switch.</li>
+     * </ul>
+     *
+     * @see ACAMERA_REQUEST_AVAILABLE_REQUEST_KEYS
+     */
+    ACAMERA_REQUEST_AVAILABLE_SESSION_KEYS =                    // int32[n]
+            ACAMERA_REQUEST_START + 16,
     ACAMERA_REQUEST_END,
 
     /**
@@ -2876,10 +3042,17 @@
      * </ul></p>
      *
      * <p>This control can be used to implement digital zoom.</p>
-     * <p>The data representation is int[4], which maps to (left, top, width, height).</p>
-     * <p>The crop region coordinate system is based off
-     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with <code>(0, 0)</code> being the
-     * top-left corner of the sensor active array.</p>
+     * <p>For devices not supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system always follows that of ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with <code>(0, 0)</code> being
+     * the top-left pixel of the active array.</p>
+     * <p>For devices supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system depends on the mode being set.
+     * When the distortion correction mode is OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE, with
+     * <code>(0, 0)</code> being the top-left pixel of the pre-correction active array.
+     * When the distortion correction mode is not OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with
+     * <code>(0, 0)</code> being the top-left pixel of the active array.</p>
      * <p>Output streams use this rectangle to produce their output,
      * cropping to a smaller region if necessary to maintain the
      * stream's aspect ratio, then scaling the sensor input to
@@ -2898,17 +3071,26 @@
      * outputs will crop horizontally (pillarbox), and 16:9
      * streams will match exactly. These additional crops will
      * be centered within the crop region.</p>
-     * <p>The width and height of the crop region cannot
-     * be set to be smaller than
+     * <p>If the coordinate system is ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, the width and height
+     * of the crop region cannot be set to be smaller than
      * <code>floor( activeArraySize.width / ACAMERA_SCALER_AVAILABLE_MAX_DIGITAL_ZOOM )</code> and
      * <code>floor( activeArraySize.height / ACAMERA_SCALER_AVAILABLE_MAX_DIGITAL_ZOOM )</code>, respectively.</p>
+     * <p>If the coordinate system is ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE, the width
+     * and height of the crop region cannot be set to be smaller than
+     * <code>floor( preCorrectionActiveArraySize.width / ACAMERA_SCALER_AVAILABLE_MAX_DIGITAL_ZOOM )</code>
+     * and
+     * <code>floor( preCorrectionActiveArraySize.height / ACAMERA_SCALER_AVAILABLE_MAX_DIGITAL_ZOOM )</code>,
+     * respectively.</p>
      * <p>The camera device may adjust the crop region to account
      * for rounding and other hardware requirements; the final
      * crop region used will be included in the output capture
      * result.</p>
+     * <p>The data representation is int[4], which maps to (left, top, width, height).</p>
      *
+     * @see ACAMERA_DISTORTION_CORRECTION_MODE
      * @see ACAMERA_SCALER_AVAILABLE_MAX_DIGITAL_ZOOM
      * @see ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE
+     * @see ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE
      */
     ACAMERA_SCALER_CROP_REGION =                                // int32[4]
             ACAMERA_SCALER_START,
@@ -2975,6 +3157,12 @@
      * IMPLEMENTATION_DEFINED | same as YUV_420_888                  | Any            |</p>
      * <p>Refer to ACAMERA_REQUEST_AVAILABLE_CAPABILITIES for additional
      * mandatory stream configurations on a per-capability basis.</p>
+     * <p>Exception on 176x144 (QCIF) resolution: camera devices usually have a fixed capability for
+     * downscaling from larger resolution to smaller, and the QCIF resolution sometimes is not
+     * fully supported due to this limitation on devices with high-resolution image sensors.
+     * Therefore, trying to configure a QCIF resolution stream together with any other
+     * stream larger than 1920x1080 resolution (either width or height) might not be supported,
+     * and capture session creation will fail if it is not.</p>
      *
      * @see ACAMERA_INFO_SUPPORTED_HARDWARE_LEVEL
      * @see ACAMERA_REQUEST_AVAILABLE_CAPABILITIES
@@ -3061,13 +3249,14 @@
      * ignored).</p>
      * <p>The following formats may always have a stall duration:</p>
      * <ul>
-     * <li>{@link AIMAGE_FORMAT_JPEG}</li>
-     * <li>{@link AIMAGE_FORMAT_RAW16}</li>
+     * <li>{@link AIMAGE_FORMAT_JPEG }</li>
+     * <li>{@link AIMAGE_FORMAT_RAW16 }</li>
      * </ul>
      * <p>The following formats will never have a stall duration:</p>
      * <ul>
-     * <li>{@link AIMAGE_FORMAT_YUV_420_888}</li>
-     * <li>{@link AIMAGE_FORMAT_RAW10}</li>
+     * <li>{@link AIMAGE_FORMAT_YUV_420_888 }</li>
+     * <li>{@link AIMAGE_FORMAT_RAW10 }</li>
+     * <li>{@link AIMAGE_FORMAT_RAW12 }</li>
      * </ul>
      * <p>All other formats may or may not have an allowed stall duration on
      * a per-capability basis; refer to ACAMERA_REQUEST_AVAILABLE_CAPABILITIES
@@ -3177,39 +3366,29 @@
      * can run concurrently to the rest of the camera pipeline, but
      * cannot process more than 1 capture at a time.</li>
      * </ul>
-     * <p>The necessary information for the application, given the model above,
-     * is provided via
-     * {@link ACAMERA_SCALER_AVAILABLE_MIN_FRAME_DURATIONS}.
-     * These are used to determine the maximum frame rate / minimum frame
-     * duration that is possible for a given stream configuration.</p>
+     * <p>The necessary information for the application, given the model above, is provided via
+     * {@link ACAMERA_SCALER_AVAILABLE_MIN_FRAME_DURATIONS }.
+     * These are used to determine the maximum frame rate / minimum frame duration that is
+     * possible for a given stream configuration.</p>
      * <p>Specifically, the application can use the following rules to
      * determine the minimum frame duration it can request from the camera
      * device:</p>
      * <ol>
-     * <li>Let the set of currently configured input/output streams
-     * be called <code>S</code>.</li>
-     * <li>Find the minimum frame durations for each stream in <code>S</code>, by looking
-     * it up in {@link ACAMERA_SCALER_AVAILABLE_MIN_FRAME_DURATIONS}
-     * (with its respective size/format). Let this set of frame durations be
-     * called <code>F</code>.</li>
-     * <li>For any given request <code>R</code>, the minimum frame duration allowed
-     * for <code>R</code> is the maximum out of all values in <code>F</code>. Let the streams
-     * used in <code>R</code> be called <code>S_r</code>.</li>
+     * <li>Let the set of currently configured input/output streams be called <code>S</code>.</li>
+     * <li>Find the minimum frame durations for each stream in <code>S</code>, by looking it up in {@link ACAMERA_SCALER_AVAILABLE_MIN_FRAME_DURATIONS }
+     * (with its respective size/format). Let this set of frame durations be called <code>F</code>.</li>
+     * <li>For any given request <code>R</code>, the minimum frame duration allowed for <code>R</code> is the maximum
+     * out of all values in <code>F</code>. Let the streams used in <code>R</code> be called <code>S_r</code>.</li>
      * </ol>
-     * <p>If none of the streams in <code>S_r</code> have a stall time (listed in {@link
-     * ACAMERA_SCALER_AVAILABLE_STALL_DURATIONS}
-     * using its respective size/format), then the frame duration in <code>F</code>
-     * determines the steady state frame rate that the application will get
-     * if it uses <code>R</code> as a repeating request. Let this special kind of
-     * request be called <code>Rsimple</code>.</p>
-     * <p>A repeating request <code>Rsimple</code> can be <em>occasionally</em> interleaved
-     * by a single capture of a new request <code>Rstall</code> (which has at least
-     * one in-use stream with a non-0 stall time) and if <code>Rstall</code> has the
-     * same minimum frame duration this will not cause a frame rate loss
-     * if all buffers from the previous <code>Rstall</code> have already been
-     * delivered.</p>
-     * <p>For more details about stalling, see
-     * {@link ACAMERA_SCALER_AVAILABLE_STALL_DURATIONS}.</p>
+     * <p>If none of the streams in <code>S_r</code> have a stall time (listed in {@link ACAMERA_SCALER_AVAILABLE_STALL_DURATIONS }
+     * using its respective size/format), then the frame duration in <code>F</code> determines the steady
+     * state frame rate that the application will get if it uses <code>R</code> as a repeating request. Let
+     * this special kind of request be called <code>Rsimple</code>.</p>
+     * <p>A repeating request <code>Rsimple</code> can be <em>occasionally</em> interleaved by a single capture of a
+     * new request <code>Rstall</code> (which has at least one in-use stream with a non-0 stall time) and if
+     * <code>Rstall</code> has the same minimum frame duration this will not cause a frame rate loss if all
+     * buffers from the previous <code>Rstall</code> have already been delivered.</p>
+     * <p>For more details about stalling, see {@link ACAMERA_SCALER_AVAILABLE_STALL_DURATIONS }.</p>
      * <p>This control is only effective if ACAMERA_CONTROL_AE_MODE or ACAMERA_CONTROL_MODE is set to
      * OFF; otherwise the auto-exposure algorithm will override this value.</p>
      *
@@ -3567,14 +3746,12 @@
      * timestamps for other captures from the same camera device, but are
      * not guaranteed to be comparable to any other time source.</p>
      * <p>When ACAMERA_SENSOR_INFO_TIMESTAMP_SOURCE <code>==</code> REALTIME, the
-     * timestamps measure time in the same timebase as
-     * <a href="https://developer.android.com/reference/android/os/SystemClock.html#elapsedRealtimeNanos">elapsedRealtimeNanos</a>
-     * (or CLOCK_BOOTTIME), and they can
+     * timestamps measure time in the same timebase as <a href="https://developer.android.com/reference/android/os/SystemClock.html#elapsedRealtimeNanos">SystemClock#elapsedRealtimeNanos</a>, and they can
      * be compared to other timestamps from other subsystems that
      * are using that base.</p>
      * <p>For reprocessing, the timestamp will match the start of exposure of
-     * the input image, i.e. {@link CaptureResult#SENSOR_TIMESTAMP the
-     * timestamp} in the TotalCaptureResult that was used to create the
+     * the input image, i.e. <a href="https://developer.android.com/reference/CaptureResult.html#SENSOR_TIMESTAMP">the
+     * timestamp</a> in the TotalCaptureResult that was used to create the
      * reprocess capture request.</p>
      *
      * @see ACAMERA_SENSOR_INFO_TIMESTAMP_SOURCE
@@ -3775,7 +3952,6 @@
      * optically shielded pixel areas. By blocking light, these pixels
      * provides a reliable black reference for black level compensation
      * in active array region.</p>
-     * <p>The data representation is int[4], which maps to (left, top, width, height).</p>
      * <p>This key provides a list of disjoint rectangles specifying the
      * regions of optically shielded (with metal shield) black pixel
      * regions if the camera device is capable of reading out these black
@@ -3785,6 +3961,7 @@
      * black level of each captured raw images.</p>
      * <p>When this key is reported, the ACAMERA_SENSOR_DYNAMIC_BLACK_LEVEL and
      * ACAMERA_SENSOR_DYNAMIC_WHITE_LEVEL will also be reported.</p>
+     * <p>The data representation is <code>int[4]</code>, which maps to <code>(left, top, width, height)</code>.</p>
      *
      * @see ACAMERA_SENSOR_BLACK_LEVEL_PATTERN
      * @see ACAMERA_SENSOR_DYNAMIC_BLACK_LEVEL
@@ -3825,9 +4002,8 @@
      * layout key (see ACAMERA_SENSOR_INFO_COLOR_FILTER_ARRANGEMENT), i.e. the
      * nth value given corresponds to the black level offset for the nth
      * color channel listed in the CFA.</p>
-     * <p>This key will be available if ACAMERA_SENSOR_OPTICAL_BLACK_REGIONS is
-     * available or the camera device advertises this key via
-     * {@link ACAMERA_REQUEST_AVAILABLE_RESULT_KEYS}.</p>
+     * <p>This key will be available if ACAMERA_SENSOR_OPTICAL_BLACK_REGIONS is available or the
+     * camera device advertises this key via {@link ACAMERA_REQUEST_AVAILABLE_RESULT_KEYS }.</p>
      *
      * @see ACAMERA_SENSOR_BLACK_LEVEL_PATTERN
      * @see ACAMERA_SENSOR_INFO_COLOR_FILTER_ARRANGEMENT
@@ -3853,7 +4029,7 @@
      * estimated white level for each frame.</p>
      * <p>This key will be available if ACAMERA_SENSOR_OPTICAL_BLACK_REGIONS is
      * available or the camera device advertises this key via
-     * {@link ACAMERA_REQUEST_AVAILABLE_RESULT_KEYS}.</p>
+     * {@link ACAMERA_REQUEST_AVAILABLE_RESULT_KEYS }.</p>
      *
      * @see ACAMERA_SENSOR_BLACK_LEVEL_PATTERN
      * @see ACAMERA_SENSOR_INFO_WHITE_LEVEL
@@ -3882,16 +4058,28 @@
      * <p>This rectangle is defined relative to the full pixel array; (0,0) is the top-left of
      * the full pixel array, and the size of the full pixel array is given by
      * ACAMERA_SENSOR_INFO_PIXEL_ARRAY_SIZE.</p>
-     * <p>The data representation is int[4], which maps to (left, top, width, height).</p>
      * <p>The coordinate system for most other keys that list pixel coordinates, including
      * ACAMERA_SCALER_CROP_REGION, is defined relative to the active array rectangle given in
      * this field, with <code>(0, 0)</code> being the top-left of this rectangle.</p>
      * <p>The active array may be smaller than the full pixel array, since the full array may
-     * include black calibration pixels or other inactive regions, and geometric correction
-     * resulting in scaling or cropping may have been applied.</p>
+     * include black calibration pixels or other inactive regions.</p>
+     * <p>For devices that do not support ACAMERA_DISTORTION_CORRECTION_MODE control, the active
+     * array must be the same as ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE.</p>
+     * <p>For devices that support ACAMERA_DISTORTION_CORRECTION_MODE control, the active array must
+     * be enclosed by ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE. The difference between
+     * pre-correction active array and active array accounts for scaling or cropping caused
+     * by lens geometric distortion correction.</p>
+     * <p>In general, application should always refer to active array size for controls like
+     * metering regions or crop region. Two exceptions are when the application is dealing with
+     * RAW image buffers (RAW_SENSOR, RAW10, RAW12 etc), or when application explicitly set
+     * ACAMERA_DISTORTION_CORRECTION_MODE to OFF. In these cases, application should refer
+     * to ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE.</p>
+     * <p>The data representation is <code>int[4]</code>, which maps to <code>(left, top, width, height)</code>.</p>
      *
+     * @see ACAMERA_DISTORTION_CORRECTION_MODE
      * @see ACAMERA_SCALER_CROP_REGION
      * @see ACAMERA_SENSOR_INFO_PIXEL_ARRAY_SIZE
+     * @see ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE
      */
     ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE =                     // int32[4]
             ACAMERA_SENSOR_INFO_START,
@@ -3960,8 +4148,7 @@
      * <p>Attempting to use frame durations beyond the maximum will result in the frame
      * duration being clipped to the maximum. See that control for a full definition of frame
      * durations.</p>
-     * <p>Refer to {@link
-     * ACAMERA_SCALER_AVAILABLE_MIN_FRAME_DURATIONS}
+     * <p>Refer to {@link ACAMERA_SCALER_AVAILABLE_MIN_FRAME_DURATIONS }
      * for the minimum frame duration values.</p>
      */
     ACAMERA_SENSOR_INFO_MAX_FRAME_DURATION =                    // int64
@@ -4000,9 +4187,9 @@
      * the raw buffers produced by this sensor.</p>
      * <p>If a camera device supports raw sensor formats, either this or
      * ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE is the maximum dimensions for the raw
-     * output formats listed in ACAMERA_SCALER_AVAILABLE_STREAM_CONFIGURATIONS (this depends on
-     * whether or not the image sensor returns buffers containing pixels that are not
-     * part of the active array region for blacklevel calibration or other purposes).</p>
+     * output formats listed in {@link ACAMERA_SCALER_AVAILABLE_STREAM_CONFIGURATIONS }
+     * (this depends on whether or not the image sensor returns buffers containing pixels that
+     * are not part of the active array region for blacklevel calibration or other purposes).</p>
      * <p>Some parts of the full pixel array may not receive light from the scene,
      * or be otherwise inactive.  The ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE key
      * defines the rectangle of active pixels that will be included in processed image
@@ -4092,7 +4279,6 @@
      *   <li>ACameraMetadata from ACameraManager_getCameraCharacteristics</li>
      * </ul></p>
      *
-     * <p>The data representation is int[4], which maps to (left, top, width, height).</p>
      * <p>This is the rectangle representing the size of the active region of the sensor (i.e.
      * the region that actually receives light from the scene) before any geometric correction
      * has been applied, and should be treated as the active region rectangle for any of the
@@ -4133,18 +4319,19 @@
      * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.</p>
      * <p>The currently supported fields that correct for geometric distortion are:</p>
      * <ol>
-     * <li>ACAMERA_LENS_RADIAL_DISTORTION.</li>
+     * <li>ACAMERA_LENS_DISTORTION.</li>
      * </ol>
-     * <p>If all of the geometric distortion fields are no-ops, this rectangle will be the same
-     * as the post-distortion-corrected rectangle given in
-     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.</p>
+     * <p>If the camera device doesn't support geometric distortion correction, or all of the
+     * geometric distortion fields are no-ops, this rectangle will be the same as the
+     * post-distortion-corrected rectangle given in ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.</p>
      * <p>This rectangle is defined relative to the full pixel array; (0,0) is the top-left of
      * the full pixel array, and the size of the full pixel array is given by
      * ACAMERA_SENSOR_INFO_PIXEL_ARRAY_SIZE.</p>
      * <p>The pre-correction active array may be smaller than the full pixel array, since the
      * full array may include black calibration pixels or other inactive regions.</p>
+     * <p>The data representation is <code>int[4]</code>, which maps to <code>(left, top, width, height)</code>.</p>
      *
-     * @see ACAMERA_LENS_RADIAL_DISTORTION
+     * @see ACAMERA_LENS_DISTORTION
      * @see ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE
      * @see ACAMERA_SENSOR_INFO_PIXEL_ARRAY_SIZE
      * @see ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE
@@ -4282,11 +4469,22 @@
      *   <li>ACameraMetadata from ACameraCaptureSession_captureCallback_result callbacks</li>
      * </ul></p>
      *
-     * <p>The coordinate system is that of ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with
+     * <p>For devices not supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system always follows that of ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with <code>(0, 0)</code> being
+     * the top-left pixel of the active array.</p>
+     * <p>For devices supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system depends on the mode being set.
+     * When the distortion correction mode is OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE, with
+     * <code>(0, 0)</code> being the top-left pixel of the pre-correction active array.
+     * When the distortion correction mode is not OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with
      * <code>(0, 0)</code> being the top-left pixel of the active array.</p>
      * <p>Only available if ACAMERA_STATISTICS_FACE_DETECT_MODE == FULL</p>
      *
+     * @see ACAMERA_DISTORTION_CORRECTION_MODE
      * @see ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE
+     * @see ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE
      * @see ACAMERA_STATISTICS_FACE_DETECT_MODE
      */
     ACAMERA_STATISTICS_FACE_LANDMARKS =                         // int32[n*6]
@@ -4302,12 +4500,23 @@
      *   <li>ACameraMetadata from ACameraCaptureSession_captureCallback_result callbacks</li>
      * </ul></p>
      *
-     * <p>The data representation is int[4], which maps to (left, top, width, height).</p>
-     * <p>The coordinate system is that of ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with
+     * <p>For devices not supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system always follows that of ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with <code>(0, 0)</code> being
+     * the top-left pixel of the active array.</p>
+     * <p>For devices supporting ACAMERA_DISTORTION_CORRECTION_MODE control, the coordinate
+     * system depends on the mode being set.
+     * When the distortion correction mode is OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE, with
+     * <code>(0, 0)</code> being the top-left pixel of the pre-correction active array.
+     * When the distortion correction mode is not OFF, the coordinate system follows
+     * ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE, with
      * <code>(0, 0)</code> being the top-left pixel of the active array.</p>
-     * <p>Only available if ACAMERA_STATISTICS_FACE_DETECT_MODE != OFF</p>
+     * <p>Only available if ACAMERA_STATISTICS_FACE_DETECT_MODE != OFF
+     * The data representation is <code>int[4]</code>, which maps to <code>(left, top, right, bottom)</code>.</p>
      *
+     * @see ACAMERA_DISTORTION_CORRECTION_MODE
      * @see ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE
+     * @see ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE
      * @see ACAMERA_STATISTICS_FACE_DETECT_MODE
      */
     ACAMERA_STATISTICS_FACE_RECTANGLES =                        // int32[n*4]
@@ -4483,6 +4692,94 @@
      */
     ACAMERA_STATISTICS_LENS_SHADING_MAP_MODE =                  // byte (acamera_metadata_enum_android_statistics_lens_shading_map_mode_t)
             ACAMERA_STATISTICS_START + 16,
+    /**
+     * <p>A control for selecting whether optical stabilization (OIS) position
+     * information is included in output result metadata.</p>
+     *
+     * <p>Type: byte (acamera_metadata_enum_android_statistics_ois_data_mode_t)</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraCaptureSession_captureCallback_result callbacks</li>
+     *   <li>ACaptureRequest</li>
+     * </ul></p>
+     *
+     * <p>Since optical image stabilization generally involves motion much faster than the duration
+     * of individualq image exposure, multiple OIS samples can be included for a single capture
+     * result. For example, if the OIS reporting operates at 200 Hz, a typical camera operating
+     * at 30fps may have 6-7 OIS samples per capture result. This information can be combined
+     * with the rolling shutter skew to account for lens motion during image exposure in
+     * post-processing algorithms.</p>
+     */
+    ACAMERA_STATISTICS_OIS_DATA_MODE =                          // byte (acamera_metadata_enum_android_statistics_ois_data_mode_t)
+            ACAMERA_STATISTICS_START + 17,
+    /**
+     * <p>An array of timestamps of OIS samples, in nanoseconds.</p>
+     *
+     * <p>Type: int64[n]</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraCaptureSession_captureCallback_result callbacks</li>
+     * </ul></p>
+     *
+     * <p>The array contains the timestamps of OIS samples. The timestamps are in the same
+     * timebase as and comparable to ACAMERA_SENSOR_TIMESTAMP.</p>
+     *
+     * @see ACAMERA_SENSOR_TIMESTAMP
+     */
+    ACAMERA_STATISTICS_OIS_TIMESTAMPS =                         // int64[n]
+            ACAMERA_STATISTICS_START + 18,
+    /**
+     * <p>An array of shifts of OIS samples, in x direction.</p>
+     *
+     * <p>Type: float[n]</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraCaptureSession_captureCallback_result callbacks</li>
+     * </ul></p>
+     *
+     * <p>The array contains the amount of shifts in x direction, in pixels, based on OIS samples.
+     * A positive value is a shift from left to right in the pre-correction active array
+     * coordinate system. For example, if the optical center is (1000, 500) in pre-correction
+     * active array coordinates, a shift of (3, 0) puts the new optical center at (1003, 500).</p>
+     * <p>The number of shifts must match the number of timestamps in
+     * ACAMERA_STATISTICS_OIS_TIMESTAMPS.</p>
+     * <p>The OIS samples are not affected by whether lens distortion correction is enabled (on
+     * supporting devices). They are always reported in pre-correction active array coordinates,
+     * since the scaling of OIS shifts would depend on the specific spot on the sensor the shift
+     * is needed.</p>
+     *
+     * @see ACAMERA_STATISTICS_OIS_TIMESTAMPS
+     */
+    ACAMERA_STATISTICS_OIS_X_SHIFTS =                           // float[n]
+            ACAMERA_STATISTICS_START + 19,
+    /**
+     * <p>An array of shifts of OIS samples, in y direction.</p>
+     *
+     * <p>Type: float[n]</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraCaptureSession_captureCallback_result callbacks</li>
+     * </ul></p>
+     *
+     * <p>The array contains the amount of shifts in y direction, in pixels, based on OIS samples.
+     * A positive value is a shift from top to bottom in pre-correction active array coordinate
+     * system. For example, if the optical center is (1000, 500) in active array coordinates, a
+     * shift of (0, 5) puts the new optical center at (1000, 505).</p>
+     * <p>The number of shifts must match the number of timestamps in
+     * ACAMERA_STATISTICS_OIS_TIMESTAMPS.</p>
+     * <p>The OIS samples are not affected by whether lens distortion correction is enabled (on
+     * supporting devices). They are always reported in pre-correction active array coordinates,
+     * since the scaling of OIS shifts would depend on the specific spot on the sensor the shift
+     * is needed.</p>
+     *
+     * @see ACAMERA_STATISTICS_OIS_TIMESTAMPS
+     */
+    ACAMERA_STATISTICS_OIS_Y_SHIFTS =                           // float[n]
+            ACAMERA_STATISTICS_START + 20,
     ACAMERA_STATISTICS_END,
 
     /**
@@ -4555,6 +4852,24 @@
      */
     ACAMERA_STATISTICS_INFO_AVAILABLE_LENS_SHADING_MAP_MODES =  // byte[n]
             ACAMERA_STATISTICS_INFO_START + 7,
+    /**
+     * <p>List of OIS data output modes for ACAMERA_STATISTICS_OIS_DATA_MODE that
+     * are supported by this camera device.</p>
+     *
+     * @see ACAMERA_STATISTICS_OIS_DATA_MODE
+     *
+     * <p>Type: byte[n]</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraManager_getCameraCharacteristics</li>
+     * </ul></p>
+     *
+     * <p>If no OIS data output is available for this camera device, this key will
+     * contain only OFF.</p>
+     */
+    ACAMERA_STATISTICS_INFO_AVAILABLE_OIS_DATA_MODES =          // byte[n]
+            ACAMERA_STATISTICS_INFO_START + 8,
     ACAMERA_STATISTICS_INFO_END,
 
     /**
@@ -4627,6 +4942,8 @@
      * of points can be less than max (that is, the request doesn't have to
      * always provide a curve with number of points equivalent to
      * ACAMERA_TONEMAP_MAX_CURVE_POINTS).</p>
+     * <p>For devices with MONOCHROME capability, only red channel is used. Green and blue channels
+     * are ignored.</p>
      * <p>A few examples, and their corresponding graphical mappings; these
      * only specify the red channel and the precision is limited to 4
      * digits, for conciseness.</p>
@@ -4809,12 +5126,26 @@
      * the following code snippet can be used:</p>
      * <pre><code>// Returns true if the device supports the required hardware level, or better.
      * boolean isHardwareLevelSupported(CameraCharacteristics c, int requiredLevel) {
+     *     final int[] sortedHwLevels = {
+     *         CameraCharacteristics.INFO_SUPPORTED_HARDWARE_LEVEL_LEGACY,
+     *         CameraCharacteristics.INFO_SUPPORTED_HARDWARE_LEVEL_EXTERNAL,
+     *         CameraCharacteristics.INFO_SUPPORTED_HARDWARE_LEVEL_LIMITED,
+     *         CameraCharacteristics.INFO_SUPPORTED_HARDWARE_LEVEL_FULL,
+     *         CameraCharacteristics.INFO_SUPPORTED_HARDWARE_LEVEL_3
+     *     };
      *     int deviceLevel = c.get(CameraCharacteristics.INFO_SUPPORTED_HARDWARE_LEVEL);
-     *     if (deviceLevel == CameraCharacteristics.INFO_SUPPORTED_HARDWARE_LEVEL_LEGACY) {
-     *         return requiredLevel == deviceLevel;
+     *     if (requiredLevel == deviceLevel) {
+     *         return true;
      *     }
-     *     // deviceLevel is not LEGACY, can use numerical sort
-     *     return requiredLevel &lt;= deviceLevel;
+     *
+     *     for (int sortedlevel : sortedHwLevels) {
+     *         if (sortedlevel == requiredLevel) {
+     *             return true;
+     *         } else if (sortedlevel == deviceLevel) {
+     *             return false;
+     *         }
+     *     }
+     *     return false; // Should never reach here
      * }
      * </code></pre>
      * <p>At a high level, the levels are:</p>
@@ -4828,11 +5159,13 @@
      *   post-processing settings, and image capture at a high rate.</li>
      * <li><code>LEVEL_3</code> devices additionally support YUV reprocessing and RAW image capture, along
      *   with additional output stream configurations.</li>
+     * <li><code>EXTERNAL</code> devices are similar to <code>LIMITED</code> devices with exceptions like some sensor or
+     *   lens information not reported or less stable framerates.</li>
      * </ul>
      * <p>See the individual level enums for full descriptions of the supported capabilities.  The
      * ACAMERA_REQUEST_AVAILABLE_CAPABILITIES entry describes the device's capabilities at a
      * finer-grain level, if needed. In addition, many controls have their available settings or
-     * ranges defined in individual metadata tag entries in this document.</p>
+     * ranges defined in individual entries from {@link ACameraManager_getCameraCharacteristics }.</p>
      * <p>Some features are not part of any particular hardware level or capability and must be
      * queried separately. These include:</p>
      * <ul>
@@ -4853,6 +5186,23 @@
      */
     ACAMERA_INFO_SUPPORTED_HARDWARE_LEVEL =                     // byte (acamera_metadata_enum_android_info_supported_hardware_level_t)
             ACAMERA_INFO_START,
+    /**
+     * <p>A short string for manufacturer version information about the camera device, such as
+     * ISP hardware, sensors, etc.</p>
+     *
+     * <p>Type: byte</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraManager_getCameraCharacteristics</li>
+     * </ul></p>
+     *
+     * <p>This can be used in <a href="https://developer.android.com/reference/android/media/ExifInterface.html#TAG_IMAGE_DESCRIPTION">TAG_IMAGE_DESCRIPTION</a>
+     * in jpeg EXIF. This key may be absent if no version information is available on the
+     * device.</p>
+     */
+    ACAMERA_INFO_VERSION =                                      // byte
+            ACAMERA_INFO_START + 1,
     ACAMERA_INFO_END,
 
     /**
@@ -5069,6 +5419,104 @@
             ACAMERA_DEPTH_START + 4,
     ACAMERA_DEPTH_END,
 
+    /**
+     * <p>The accuracy of frame timestamp synchronization between physical cameras</p>
+     *
+     * <p>Type: byte (acamera_metadata_enum_android_logical_multi_camera_sensor_sync_type_t)</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraManager_getCameraCharacteristics</li>
+     * </ul></p>
+     *
+     * <p>The accuracy of the frame timestamp synchronization determines the physical cameras'
+     * ability to start exposure at the same time. If the sensorSyncType is CALIBRATED,
+     * the physical camera sensors usually run in master-slave mode so that their shutter
+     * time is synchronized. For APPROXIMATE sensorSyncType, the camera sensors usually run in
+     * master-master mode, and there could be offset between their start of exposure.</p>
+     * <p>In both cases, all images generated for a particular capture request still carry the same
+     * timestamps, so that they can be used to look up the matching frame number and
+     * onCaptureStarted callback.</p>
+     */
+    ACAMERA_LOGICAL_MULTI_CAMERA_SENSOR_SYNC_TYPE =             // byte (acamera_metadata_enum_android_logical_multi_camera_sensor_sync_type_t)
+            ACAMERA_LOGICAL_MULTI_CAMERA_START + 1,
+    ACAMERA_LOGICAL_MULTI_CAMERA_END,
+
+    /**
+     * <p>Mode of operation for the lens distortion correction block.</p>
+     *
+     * <p>Type: byte (acamera_metadata_enum_android_distortion_correction_mode_t)</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraCaptureSession_captureCallback_result callbacks</li>
+     *   <li>ACaptureRequest</li>
+     * </ul></p>
+     *
+     * <p>The lens distortion correction block attempts to improve image quality by fixing
+     * radial, tangential, or other geometric aberrations in the camera device's optics.  If
+     * available, the ACAMERA_LENS_DISTORTION field documents the lens's distortion parameters.</p>
+     * <p>OFF means no distortion correction is done.</p>
+     * <p>FAST/HIGH_QUALITY both mean camera device determined distortion correction will be
+     * applied. HIGH_QUALITY mode indicates that the camera device will use the highest-quality
+     * correction algorithms, even if it slows down capture rate. FAST means the camera device
+     * will not slow down capture rate when applying correction. FAST may be the same as OFF if
+     * any correction at all would slow down capture rate.  Every output stream will have a
+     * similar amount of enhancement applied.</p>
+     * <p>The correction only applies to processed outputs such as YUV, JPEG, or DEPTH16; it is not
+     * applied to any RAW output.</p>
+     * <p>This control will be on by default on devices that support this control. Applications
+     * disabling distortion correction need to pay extra attention with the coordinate system of
+     * metering regions, crop region, and face rectangles. When distortion correction is OFF,
+     * metadata coordinates follow the coordinate system of
+     * ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE. When distortion is not OFF, metadata
+     * coordinates follow the coordinate system of ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE.  The
+     * camera device will map these metadata fields to match the corrected image produced by the
+     * camera device, for both capture requests and results.  However, this mapping is not very
+     * precise, since rectangles do not generally map to rectangles when corrected.  Only linear
+     * scaling between the active array and precorrection active array coordinates is
+     * performed. Applications that require precise correction of metadata need to undo that
+     * linear scaling, and apply a more complete correction that takes into the account the app's
+     * own requirements.</p>
+     * <p>The full list of metadata that is affected in this way by distortion correction is:</p>
+     * <ul>
+     * <li>ACAMERA_CONTROL_AF_REGIONS</li>
+     * <li>ACAMERA_CONTROL_AE_REGIONS</li>
+     * <li>ACAMERA_CONTROL_AWB_REGIONS</li>
+     * <li>ACAMERA_SCALER_CROP_REGION</li>
+     * <li>android.statistics.faces</li>
+     * </ul>
+     *
+     * @see ACAMERA_CONTROL_AE_REGIONS
+     * @see ACAMERA_CONTROL_AF_REGIONS
+     * @see ACAMERA_CONTROL_AWB_REGIONS
+     * @see ACAMERA_LENS_DISTORTION
+     * @see ACAMERA_SCALER_CROP_REGION
+     * @see ACAMERA_SENSOR_INFO_ACTIVE_ARRAY_SIZE
+     * @see ACAMERA_SENSOR_INFO_PRE_CORRECTION_ACTIVE_ARRAY_SIZE
+     */
+    ACAMERA_DISTORTION_CORRECTION_MODE =                        // byte (acamera_metadata_enum_android_distortion_correction_mode_t)
+            ACAMERA_DISTORTION_CORRECTION_START,
+    /**
+     * <p>List of distortion correction modes for ACAMERA_DISTORTION_CORRECTION_MODE that are
+     * supported by this camera device.</p>
+     *
+     * @see ACAMERA_DISTORTION_CORRECTION_MODE
+     *
+     * <p>Type: byte[n]</p>
+     *
+     * <p>This tag may appear in:
+     * <ul>
+     *   <li>ACameraMetadata from ACameraManager_getCameraCharacteristics</li>
+     * </ul></p>
+     *
+     * <p>No device is required to support this API; such devices will always list only 'OFF'.
+     * All devices that support this API will list both FAST and HIGH_QUALITY.</p>
+     */
+    ACAMERA_DISTORTION_CORRECTION_AVAILABLE_MODES =             // byte[n]
+            ACAMERA_DISTORTION_CORRECTION_START + 1,
+    ACAMERA_DISTORTION_CORRECTION_END,
+
 } acamera_metadata_tag_t;
 
 /**
@@ -5282,6 +5730,21 @@
      */
     ACAMERA_CONTROL_AE_MODE_ON_AUTO_FLASH_REDEYE                     = 4,
 
+    /**
+     * <p>An external flash has been turned on.</p>
+     * <p>It informs the camera device that an external flash has been turned on, and that
+     * metering (and continuous focus if active) should be quickly recaculated to account
+     * for the external flash. Otherwise, this mode acts like ON.</p>
+     * <p>When the external flash is turned off, AE mode should be changed to one of the
+     * other available AE modes.</p>
+     * <p>If the camera device supports AE external flash mode, ACAMERA_CONTROL_AE_STATE must
+     * be FLASH_REQUIRED after the camera device finishes AE scan and it's too dark without
+     * flash.</p>
+     *
+     * @see ACAMERA_CONTROL_AE_STATE
+     */
+    ACAMERA_CONTROL_AE_MODE_ON_EXTERNAL_FLASH                        = 5,
+
 } acamera_metadata_enum_android_control_ae_mode_t;
 
 // ACAMERA_CONTROL_AE_PRECAPTURE_TRIGGER
@@ -5645,6 +6108,15 @@
      */
     ACAMERA_CONTROL_CAPTURE_INTENT_MANUAL                            = 6,
 
+    /**
+     * <p>This request is for a motion tracking use case, where
+     * the application will use camera and inertial sensor data to
+     * locate and track objects in the world.</p>
+     * <p>The camera device auto-exposure routine will limit the exposure time
+     * of the camera to no more than 20 milliseconds, to minimize motion blur.</p>
+     */
+    ACAMERA_CONTROL_CAPTURE_INTENT_MOTION_TRACKING                   = 7,
+
 } acamera_metadata_enum_android_control_capture_intent_t;
 
 // ACAMERA_CONTROL_EFFECT_MODE
@@ -6135,6 +6607,20 @@
 
 } acamera_metadata_enum_android_control_enable_zsl_t;
 
+// ACAMERA_CONTROL_AF_SCENE_CHANGE
+typedef enum acamera_metadata_enum_acamera_control_af_scene_change {
+    /**
+     * <p>Scene change is not detected within the AF region(s).</p>
+     */
+    ACAMERA_CONTROL_AF_SCENE_CHANGE_NOT_DETECTED                     = 0,
+
+    /**
+     * <p>Scene change is detected within the AF region(s).</p>
+     */
+    ACAMERA_CONTROL_AF_SCENE_CHANGE_DETECTED                         = 1,
+
+} acamera_metadata_enum_android_control_af_scene_change_t;
+
 
 
 // ACAMERA_EDGE_MODE
@@ -6157,13 +6643,13 @@
     ACAMERA_EDGE_MODE_HIGH_QUALITY                                   = 2,
 
     /**
-     * <p>Edge enhancement is applied at different levels for different output streams,
-     * based on resolution. Streams at maximum recording resolution (see {@link
-     * ACameraDevice_createCaptureSession}) or below have
-     * edge enhancement applied, while higher-resolution streams have no edge enhancement
-     * applied. The level of edge enhancement for low-resolution streams is tuned so that
-     * frame rate is not impacted, and the quality is equal to or better than FAST (since it
-     * is only applied to lower-resolution outputs, quality may improve from FAST).</p>
+     * <p>Edge enhancement is applied at different
+     * levels for different output streams, based on resolution. Streams at maximum recording
+     * resolution (see {@link ACameraDevice_createCaptureSession })
+     * or below have edge enhancement applied, while higher-resolution streams have no edge
+     * enhancement applied. The level of edge enhancement for low-resolution streams is tuned
+     * so that frame rate is not impacted, and the quality is equal to or better than FAST
+     * (since it is only applied to lower-resolution outputs, quality may improve from FAST).</p>
      * <p>This mode is intended to be used by applications operating in a zero-shutter-lag mode
      * with YUV or PRIVATE reprocessing, where the application continuously captures
      * high-resolution intermediate buffers into a circular buffer, from which a final image is
@@ -6342,6 +6828,27 @@
 
 } acamera_metadata_enum_android_lens_state_t;
 
+// ACAMERA_LENS_POSE_REFERENCE
+typedef enum acamera_metadata_enum_acamera_lens_pose_reference {
+    /**
+     * <p>The value of ACAMERA_LENS_POSE_TRANSLATION is relative to the optical center of
+     * the largest camera device facing the same direction as this camera.</p>
+     * <p>This is the default value for API levels before Android P.</p>
+     *
+     * @see ACAMERA_LENS_POSE_TRANSLATION
+     */
+    ACAMERA_LENS_POSE_REFERENCE_PRIMARY_CAMERA                       = 0,
+
+    /**
+     * <p>The value of ACAMERA_LENS_POSE_TRANSLATION is relative to the position of the
+     * primary gyroscope of this Android device.</p>
+     *
+     * @see ACAMERA_LENS_POSE_TRANSLATION
+     */
+    ACAMERA_LENS_POSE_REFERENCE_GYROSCOPE                            = 1,
+
+} acamera_metadata_enum_android_lens_pose_reference_t;
+
 
 // ACAMERA_LENS_INFO_FOCUS_DISTANCE_CALIBRATION
 typedef enum acamera_metadata_enum_acamera_lens_info_focus_distance_calibration {
@@ -6412,13 +6919,12 @@
 
     /**
      * <p>Noise reduction is applied at different levels for different output streams,
-     * based on resolution. Streams at maximum recording resolution (see {@link
-     * ACameraDevice_createCaptureSession}) or below have noise
-     * reduction applied, while higher-resolution streams have MINIMAL (if supported) or no
-     * noise reduction applied (if MINIMAL is not supported.) The degree of noise reduction
-     * for low-resolution streams is tuned so that frame rate is not impacted, and the quality
-     * is equal to or better than FAST (since it is only applied to lower-resolution outputs,
-     * quality may improve from FAST).</p>
+     * based on resolution. Streams at maximum recording resolution (see {@link ACameraDevice_createCaptureSession })
+     * or below have noise reduction applied, while higher-resolution streams have MINIMAL (if
+     * supported) or no noise reduction applied (if MINIMAL is not supported.) The degree of
+     * noise reduction for low-resolution streams is tuned so that frame rate is not impacted,
+     * and the quality is equal to or better than FAST (since it is only applied to
+     * lower-resolution outputs, quality may improve from FAST).</p>
      * <p>This mode is intended to be used by applications operating in a zero-shutter-lag mode
      * with YUV or PRIVATE reprocessing, where the application continuously captures
      * high-resolution intermediate buffers into a circular buffer, from which a final image is
@@ -6635,18 +7141,16 @@
      * to FAST. Additionally, maximum-resolution images can be captured at &gt;= 10 frames
      * per second.  Here, 'high resolution' means at least 8 megapixels, or the maximum
      * resolution of the device, whichever is smaller.</p>
-     * <p>More specifically, this means that at least one output {@link
-     * AIMAGE_FORMAT_YUV_420_888} size listed in
-     * {@link ACAMERA_SCALER_AVAILABLE_STREAM_CONFIGURATIONS} is larger or equal to the
-     * 'high resolution' defined above, and can be captured at at least 20 fps.
-     * For the largest {@link AIMAGE_FORMAT_YUV_420_888} size listed in
-     * {@link ACAMERA_SCALER_AVAILABLE_STREAM_CONFIGURATIONS}, camera device can capture this
-     * size for at least 10 frames per second.
-     * Also the ACAMERA_CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES entry lists at least one FPS range
-     * where the minimum FPS is &gt;= 1 / minimumFrameDuration for the largest YUV_420_888 size.</p>
-     * <p>If the device supports the {@link AIMAGE_FORMAT_RAW10}, {@link
-     * AIMAGE_FORMAT_RAW12}, then those can also be captured at the same rate
-     * as the maximum-size YUV_420_888 resolution is.</p>
+     * <p>More specifically, this means that at least one output {@link AIMAGE_FORMAT_YUV_420_888 } size listed in
+     * {@link ACAMERA_SCALER_AVAILABLE_STREAM_CONFIGURATIONS }
+     * is larger or equal to the 'high resolution' defined above, and can be captured at at
+     * least 20 fps.  For the largest {@link AIMAGE_FORMAT_YUV_420_888 } size listed in
+     * {@link ACAMERA_SCALER_AVAILABLE_STREAM_CONFIGURATIONS },
+     * camera device can capture this size for at least 10 frames per second.  Also the
+     * ACAMERA_CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES entry lists at least one FPS range where
+     * the minimum FPS is &gt;= 1 / minimumFrameDuration for the largest YUV_420_888 size.</p>
+     * <p>If the device supports the {@link AIMAGE_FORMAT_RAW10 }, {@link AIMAGE_FORMAT_RAW12 }, then those can also be
+     * captured at the same rate as the maximum-size YUV_420_888 resolution is.</p>
      * <p>In addition, the ACAMERA_SYNC_MAX_LATENCY field is guaranted to have a value between 0
      * and 4, inclusive. ACAMERA_CONTROL_AE_LOCK_AVAILABLE and ACAMERA_CONTROL_AWB_LOCK_AVAILABLE
      * are also guaranteed to be <code>true</code> so burst capture with these two locks ON yields
@@ -6663,42 +7167,114 @@
      * <p>The camera device can produce depth measurements from its field of view.</p>
      * <p>This capability requires the camera device to support the following:</p>
      * <ul>
-     * <li>{@link AIMAGE_FORMAT_DEPTH16} is supported as an output format.</li>
-     * <li>{@link AIMAGE_FORMAT_DEPTH_POINT_CLOUD} is optionally supported as an
-     *   output format.</li>
-     * <li>This camera device, and all camera devices with the same ACAMERA_LENS_FACING,
-     *   will list the following calibration entries in {@link ACameraMetadata} from both
-     *   {@link ACameraManager_getCameraCharacteristics} and
-     *   {@link ACameraCaptureSession_captureCallback_result}:<ul>
+     * <li>{@link AIMAGE_FORMAT_DEPTH16 } is supported as
+     *   an output format.</li>
+     * <li>{@link AIMAGE_FORMAT_DEPTH_POINT_CLOUD } is
+     *   optionally supported as an output format.</li>
+     * <li>This camera device, and all camera devices with the same ACAMERA_LENS_FACING, will
+     *   list the following calibration metadata entries in both {@link ACameraManager_getCameraCharacteristics }
+     *   and {@link ACameraCaptureSession_captureCallback_result }:<ul>
      * <li>ACAMERA_LENS_POSE_TRANSLATION</li>
      * <li>ACAMERA_LENS_POSE_ROTATION</li>
      * <li>ACAMERA_LENS_INTRINSIC_CALIBRATION</li>
-     * <li>ACAMERA_LENS_RADIAL_DISTORTION</li>
+     * <li>ACAMERA_LENS_DISTORTION</li>
      * </ul>
      * </li>
      * <li>The ACAMERA_DEPTH_DEPTH_IS_EXCLUSIVE entry is listed by this device.</li>
+     * <li>As of Android P, the ACAMERA_LENS_POSE_REFERENCE entry is listed by this device.</li>
      * <li>A LIMITED camera with only the DEPTH_OUTPUT capability does not have to support
      *   normal YUV_420_888, JPEG, and PRIV-format outputs. It only has to support the DEPTH16
      *   format.</li>
      * </ul>
      * <p>Generally, depth output operates at a slower frame rate than standard color capture,
      * so the DEPTH16 and DEPTH_POINT_CLOUD formats will commonly have a stall duration that
-     * should be accounted for (see
-     * {@link ACAMERA_DEPTH_AVAILABLE_DEPTH_STALL_DURATIONS}).
+     * should be accounted for (see {@link ACAMERA_DEPTH_AVAILABLE_DEPTH_STALL_DURATIONS }).
      * On a device that supports both depth and color-based output, to enable smooth preview,
      * using a repeating burst is recommended, where a depth-output target is only included
      * once every N frames, where N is the ratio between preview output rate and depth output
      * rate, including depth stall time.</p>
      *
      * @see ACAMERA_DEPTH_DEPTH_IS_EXCLUSIVE
+     * @see ACAMERA_LENS_DISTORTION
      * @see ACAMERA_LENS_FACING
      * @see ACAMERA_LENS_INTRINSIC_CALIBRATION
+     * @see ACAMERA_LENS_POSE_REFERENCE
      * @see ACAMERA_LENS_POSE_ROTATION
      * @see ACAMERA_LENS_POSE_TRANSLATION
-     * @see ACAMERA_LENS_RADIAL_DISTORTION
      */
     ACAMERA_REQUEST_AVAILABLE_CAPABILITIES_DEPTH_OUTPUT              = 8,
 
+    /**
+     * <p>The camera device supports the MOTION_TRACKING value for
+     * ACAMERA_CONTROL_CAPTURE_INTENT, which limits maximum exposure time to 20 ms.</p>
+     * <p>This limits the motion blur of capture images, resulting in better image tracking
+     * results for use cases such as image stabilization or augmented reality.</p>
+     *
+     * @see ACAMERA_CONTROL_CAPTURE_INTENT
+     */
+    ACAMERA_REQUEST_AVAILABLE_CAPABILITIES_MOTION_TRACKING           = 10,
+
+    /**
+     * <p>The camera device is a logical camera backed by two or more physical cameras that are
+     * also exposed to the application.</p>
+     * <p>Camera application shouldn't assume that there are at most 1 rear camera and 1 front
+     * camera in the system. For an application that switches between front and back cameras,
+     * the recommendation is to switch between the first rear camera and the first front
+     * camera in the list of supported camera devices.</p>
+     * <p>This capability requires the camera device to support the following:</p>
+     * <ul>
+     * <li>The IDs of underlying physical cameras are returned via
+     *   <a href="https://developer.android.com/reference/android/hardware/camera2/CameraCharacteristics.html#getPhysicalCameraIds">CameraCharacteristics#getPhysicalCameraIds</a>.</li>
+     * <li>This camera device must list static metadata
+     *   ACAMERA_LOGICAL_MULTI_CAMERA_SENSOR_SYNC_TYPE in
+     *   <a href="https://developer.android.com/reference/android/hardware/camera2/CameraCharacteristics.html">CameraCharacteristics</a>.</li>
+     * <li>The underlying physical cameras' static metadata must list the following entries,
+     *   so that the application can correlate pixels from the physical streams:<ul>
+     * <li>ACAMERA_LENS_POSE_REFERENCE</li>
+     * <li>ACAMERA_LENS_POSE_ROTATION</li>
+     * <li>ACAMERA_LENS_POSE_TRANSLATION</li>
+     * <li>ACAMERA_LENS_INTRINSIC_CALIBRATION</li>
+     * <li>ACAMERA_LENS_DISTORTION</li>
+     * </ul>
+     * </li>
+     * <li>The SENSOR_INFO_TIMESTAMP_SOURCE of the logical device and physical devices must be
+     *   the same.</li>
+     * <li>The logical camera device must be LIMITED or higher device.</li>
+     * </ul>
+     * <p>Both the logical camera device and its underlying physical devices support the
+     * mandatory stream combinations required for their device levels.</p>
+     * <p>Additionally, for each guaranteed stream combination, the logical camera supports:</p>
+     * <ul>
+     * <li>For each guaranteed stream combination, the logical camera supports replacing one
+     *   logical {@link AIMAGE_FORMAT_YUV_420_888 YUV_420_888}
+     *   or raw stream with two physical streams of the same size and format, each from a
+     *   separate physical camera, given that the size and format are supported by both
+     *   physical cameras.</li>
+     * <li>If the logical camera doesn't advertise RAW capability, but the underlying physical
+     *   cameras do, the logical camera will support guaranteed stream combinations for RAW
+     *   capability, except that the RAW streams will be physical streams, each from a separate
+     *   physical camera. This is usually the case when the physical cameras have different
+     *   sensor sizes.</li>
+     * </ul>
+     * <p>Using physical streams in place of a logical stream of the same size and format will
+     * not slow down the frame rate of the capture, as long as the minimum frame duration
+     * of the physical and logical streams are the same.</p>
+     *
+     * @see ACAMERA_LENS_DISTORTION
+     * @see ACAMERA_LENS_INTRINSIC_CALIBRATION
+     * @see ACAMERA_LENS_POSE_REFERENCE
+     * @see ACAMERA_LENS_POSE_ROTATION
+     * @see ACAMERA_LENS_POSE_TRANSLATION
+     * @see ACAMERA_LOGICAL_MULTI_CAMERA_SENSOR_SYNC_TYPE
+     */
+    ACAMERA_REQUEST_AVAILABLE_CAPABILITIES_LOGICAL_MULTI_CAMERA      = 11,
+
+    /**
+     * <p>The camera device is a monochrome camera that doesn't contain a color filter array,
+     * and the pixel values on U and V planes are all 128.</p>
+     */
+    ACAMERA_REQUEST_AVAILABLE_CAPABILITIES_MONOCHROME                = 12,
+
 } acamera_metadata_enum_android_request_available_capabilities_t;
 
 
@@ -6918,8 +7494,8 @@
 
     /**
      * <p>Timestamps from ACAMERA_SENSOR_TIMESTAMP are in the same timebase as
-     * <a href="https://developer.android.com/reference/android/os/SystemClock.html#elapsedRealtimeNanos">elapsedRealtimeNanos</a>
-     * (or CLOCK_BOOTTIME), and they can be compared to other timestamps using that base.</p>
+     * <a href="https://developer.android.com/reference/android/os/SystemClock.html#elapsedRealtimeNanos">SystemClock#elapsedRealtimeNanos</a>,
+     * and they can be compared to other timestamps using that base.</p>
      *
      * @see ACAMERA_SENSOR_TIMESTAMP
      */
@@ -7030,6 +7606,26 @@
 
 } acamera_metadata_enum_android_statistics_lens_shading_map_mode_t;
 
+// ACAMERA_STATISTICS_OIS_DATA_MODE
+typedef enum acamera_metadata_enum_acamera_statistics_ois_data_mode {
+    /**
+     * <p>Do not include OIS data in the capture result.</p>
+     */
+    ACAMERA_STATISTICS_OIS_DATA_MODE_OFF                             = 0,
+
+    /**
+     * <p>Include OIS data in the capture result.</p>
+     * <p>ACAMERA_STATISTICS_OIS_TIMESTAMPS, ACAMERA_STATISTICS_OIS_X_SHIFTS,
+     * and ACAMERA_STATISTICS_OIS_Y_SHIFTS provide OIS data in the output result metadata.</p>
+     *
+     * @see ACAMERA_STATISTICS_OIS_TIMESTAMPS
+     * @see ACAMERA_STATISTICS_OIS_X_SHIFTS
+     * @see ACAMERA_STATISTICS_OIS_Y_SHIFTS
+     */
+    ACAMERA_STATISTICS_OIS_DATA_MODE_ON                              = 1,
+
+} acamera_metadata_enum_android_statistics_ois_data_mode_t;
+
 
 
 // ACAMERA_TONEMAP_MODE
@@ -7104,7 +7700,7 @@
      * <p>This camera device does not have enough capabilities to qualify as a <code>FULL</code> device or
      * better.</p>
      * <p>Only the stream configurations listed in the <code>LEGACY</code> and <code>LIMITED</code> tables in the
-     * {@link ACameraDevice_createCaptureSession} documentation are guaranteed to be supported.</p>
+     * {@link ACameraDevice_createCaptureSession createCaptureSession} documentation are guaranteed to be supported.</p>
      * <p>All <code>LIMITED</code> devices support the <code>BACKWARDS_COMPATIBLE</code> capability, indicating basic
      * support for color image capture. The only exception is that the device may
      * alternatively support only the <code>DEPTH_OUTPUT</code> capability, if it can only output depth
@@ -7130,7 +7726,7 @@
     /**
      * <p>This camera device is capable of supporting advanced imaging applications.</p>
      * <p>The stream configurations listed in the <code>FULL</code>, <code>LEGACY</code> and <code>LIMITED</code> tables in the
-     * {@link ACameraDevice_createCaptureSession} documentation are guaranteed to be supported.</p>
+     * {@link ACameraDevice_createCaptureSession createCaptureSession} documentation are guaranteed to be supported.</p>
      * <p>A <code>FULL</code> device will support below capabilities:</p>
      * <ul>
      * <li><code>BURST_CAPTURE</code> capability (ACAMERA_REQUEST_AVAILABLE_CAPABILITIES contains
@@ -7157,8 +7753,7 @@
 
     /**
      * <p>This camera device is running in backward compatibility mode.</p>
-     * <p>Only the stream configurations listed in the <code>LEGACY</code> table in the {@link
-     * ACameraDevice_createCaptureSession} documentation are supported.</p>
+     * <p>Only the stream configurations listed in the <code>LEGACY</code> table in the {@link ACameraDevice_createCaptureSession createCaptureSession} documentation are supported.</p>
      * <p>A <code>LEGACY</code> device does not support per-frame control, manual sensor control, manual
      * post-processing, arbitrary cropping regions, and has relaxed performance constraints.
      * No additional capabilities beyond <code>BACKWARD_COMPATIBLE</code> will ever be listed by a
@@ -7179,9 +7774,7 @@
      * <p>This camera device is capable of YUV reprocessing and RAW data capture, in addition to
      * FULL-level capabilities.</p>
      * <p>The stream configurations listed in the <code>LEVEL_3</code>, <code>RAW</code>, <code>FULL</code>, <code>LEGACY</code> and
-     * <code>LIMITED</code> tables in the {@link
-     * ACameraDevice_createCaptureSession}
-     * documentation are guaranteed to be supported.</p>
+     * <code>LIMITED</code> tables in the {@link ACameraDevice_createCaptureSession createCaptureSession} documentation are guaranteed to be supported.</p>
      * <p>The following additional capabilities are guaranteed to be supported:</p>
      * <ul>
      * <li><code>YUV_REPROCESSING</code> capability (ACAMERA_REQUEST_AVAILABLE_CAPABILITIES contains
@@ -7194,6 +7787,37 @@
      */
     ACAMERA_INFO_SUPPORTED_HARDWARE_LEVEL_3                          = 3,
 
+    /**
+     * <p>This camera device is backed by an external camera connected to this Android device.</p>
+     * <p>The device has capability identical to a LIMITED level device, with the following
+     * exceptions:</p>
+     * <ul>
+     * <li>The device may not report lens/sensor related information such as<ul>
+     * <li>ACAMERA_LENS_FOCAL_LENGTH</li>
+     * <li>ACAMERA_LENS_INFO_HYPERFOCAL_DISTANCE</li>
+     * <li>ACAMERA_SENSOR_INFO_PHYSICAL_SIZE</li>
+     * <li>ACAMERA_SENSOR_INFO_WHITE_LEVEL</li>
+     * <li>ACAMERA_SENSOR_BLACK_LEVEL_PATTERN</li>
+     * <li>ACAMERA_SENSOR_INFO_COLOR_FILTER_ARRANGEMENT</li>
+     * <li>ACAMERA_SENSOR_ROLLING_SHUTTER_SKEW</li>
+     * </ul>
+     * </li>
+     * <li>The device will report 0 for ACAMERA_SENSOR_ORIENTATION</li>
+     * <li>The device has less guarantee on stable framerate, as the framerate partly depends
+     *   on the external camera being used.</li>
+     * </ul>
+     *
+     * @see ACAMERA_LENS_FOCAL_LENGTH
+     * @see ACAMERA_LENS_INFO_HYPERFOCAL_DISTANCE
+     * @see ACAMERA_SENSOR_BLACK_LEVEL_PATTERN
+     * @see ACAMERA_SENSOR_INFO_COLOR_FILTER_ARRANGEMENT
+     * @see ACAMERA_SENSOR_INFO_PHYSICAL_SIZE
+     * @see ACAMERA_SENSOR_INFO_WHITE_LEVEL
+     * @see ACAMERA_SENSOR_ORIENTATION
+     * @see ACAMERA_SENSOR_ROLLING_SHUTTER_SKEW
+     */
+    ACAMERA_INFO_SUPPORTED_HARDWARE_LEVEL_EXTERNAL                   = 4,
+
 } acamera_metadata_enum_android_info_supported_hardware_level_t;
 
 
@@ -7281,6 +7905,48 @@
 } acamera_metadata_enum_android_depth_depth_is_exclusive_t;
 
 
+// ACAMERA_LOGICAL_MULTI_CAMERA_SENSOR_SYNC_TYPE
+typedef enum acamera_metadata_enum_acamera_logical_multi_camera_sensor_sync_type {
+    /**
+     * <p>A software mechanism is used to synchronize between the physical cameras. As a result,
+     * the timestamp of an image from a physical stream is only an approximation of the
+     * image sensor start-of-exposure time.</p>
+     */
+    ACAMERA_LOGICAL_MULTI_CAMERA_SENSOR_SYNC_TYPE_APPROXIMATE        = 0,
+
+    /**
+     * <p>The camera device supports frame timestamp synchronization at the hardware level,
+     * and the timestamp of a physical stream image accurately reflects its
+     * start-of-exposure time.</p>
+     */
+    ACAMERA_LOGICAL_MULTI_CAMERA_SENSOR_SYNC_TYPE_CALIBRATED         = 1,
+
+} acamera_metadata_enum_android_logical_multi_camera_sensor_sync_type_t;
+
+
+// ACAMERA_DISTORTION_CORRECTION_MODE
+typedef enum acamera_metadata_enum_acamera_distortion_correction_mode {
+    /**
+     * <p>No distortion correction is applied.</p>
+     */
+    ACAMERA_DISTORTION_CORRECTION_MODE_OFF                           = 0,
+
+    /**
+     * <p>Lens distortion correction is applied without reducing frame rate
+     * relative to sensor output. It may be the same as OFF if distortion correction would
+     * reduce frame rate relative to sensor.</p>
+     */
+    ACAMERA_DISTORTION_CORRECTION_MODE_FAST                          = 1,
+
+    /**
+     * <p>High-quality distortion correction is applied, at the cost of
+     * possibly reduced frame rate relative to sensor output.</p>
+     */
+    ACAMERA_DISTORTION_CORRECTION_MODE_HIGH_QUALITY                  = 2,
+
+} acamera_metadata_enum_android_distortion_correction_mode_t;
+
+
 #endif /* __ANDROID_API__ >= 24 */
 
 __END_DECLS

diff --git a/camera/ndk/include/camera/NdkCaptureRequest.h b/camera/ndk/include/camera/NdkCaptureRequest.h
index c62ba2c..4961ce3 100644
--- a/camera/ndk/include/camera/NdkCaptureRequest.h
+++ b/camera/ndk/include/camera/NdkCaptureRequest.h

@@ -305,6 +305,58 @@
 
 #endif /* __ANDROID_API__ >= 24 */
 
+#if __ANDROID_API__ >= 28
+
+/**
+ * Associate an arbitrary user context pointer to the {@link ACaptureRequest}
+ *
+ * This method is useful for user to identify the capture request in capture session callbacks.
+ * The context is NULL for newly created request.
+ * {@link ACameraOutputTarget_free} will not free the context. Also calling this method twice
+ * will not cause the previous context be freed.
+ * Also note that calling this method after the request has been sent to capture session will not
+ * change the context pointer in the capture callbacks.
+ *
+ * @param request the {@link ACaptureRequest} of interest.
+ * @param context the user context pointer to be associated with this capture request.
+ *
+ * @return <ul>
+ *         <li>{@link ACAMERA_OK} if the method call succeeds.</li>
+ *         <li>{@link ACAMERA_ERROR_INVALID_PARAMETER} if request is NULL.</li></ul>
+ */
+camera_status_t ACaptureRequest_setUserContext(
+        ACaptureRequest* request, void* context);
+
+/**
+ * Get the user context pointer of the {@link ACaptureRequest}
+ *
+ * This method is useful for user to identify the capture request in capture session callbacks.
+ * The context is NULL for newly created request.
+ *
+ * @param request the {@link ACaptureRequest} of interest.
+ * @param context the user context pointer of this capture request.
+ *
+ * @return <ul>
+ *         <li>{@link ACAMERA_OK} if the method call succeeds.</li>
+ *         <li>{@link ACAMERA_ERROR_INVALID_PARAMETER} if request is NULL.</li></ul>
+ */
+camera_status_t ACaptureRequest_getUserContext(
+        const ACaptureRequest* request, /*out*/void** context);
+
+/**
+ * Create a copy of input {@link ACaptureRequest}.
+ *
+ * <p>The returned ACaptureRequest must be freed by the application by {@link ACaptureRequest_free}
+ * after application is done using it.</p>
+ *
+ * @param src the input {@link ACaptureRequest} to be copied.
+ *
+ * @return a valid ACaptureRequest pointer or NULL if the input request cannot be copied.
+ */
+ACaptureRequest* ACaptureRequest_copy(const ACaptureRequest* src);
+
+#endif /* __ANDROID_API__ >= 28 */
+
 __END_DECLS
 
 #endif /* _NDK_CAPTURE_REQUEST_H */

diff --git a/camera/ndk/libcamera2ndk.map.txt b/camera/ndk/libcamera2ndk.map.txt
index 41bb22b..d179aa0 100644
--- a/camera/ndk/libcamera2ndk.map.txt
+++ b/camera/ndk/libcamera2ndk.map.txt

@@ -6,9 +6,11 @@
     ACameraCaptureSession_getDevice;
     ACameraCaptureSession_setRepeatingRequest;
     ACameraCaptureSession_stopRepeating;
+    ACameraCaptureSession_updateSharedOutput;
     ACameraDevice_close;
     ACameraDevice_createCaptureRequest;
     ACameraDevice_createCaptureSession;
+    ACameraDevice_createCaptureSessionWithSessionParameters;
     ACameraDevice_getId;
     ACameraManager_create;
     ACameraManager_delete;
@@ -25,9 +27,11 @@
     ACameraOutputTarget_create;
     ACameraOutputTarget_free;
     ACaptureRequest_addTarget;
+    ACaptureRequest_copy;
     ACaptureRequest_free;
     ACaptureRequest_getAllTags;
     ACaptureRequest_getConstEntry;
+    ACaptureRequest_getUserContext;
     ACaptureRequest_removeTarget;
     ACaptureRequest_setEntry_double;
     ACaptureRequest_setEntry_float;
@@ -35,11 +39,15 @@
     ACaptureRequest_setEntry_i64;
     ACaptureRequest_setEntry_rational;
     ACaptureRequest_setEntry_u8;
+    ACaptureRequest_setUserContext;
     ACaptureSessionOutputContainer_add;
     ACaptureSessionOutputContainer_create;
     ACaptureSessionOutputContainer_free;
     ACaptureSessionOutputContainer_remove;
     ACaptureSessionOutput_create;
+    ACaptureSessionSharedOutput_create;
+    ACaptureSessionSharedOutput_add;
+    ACaptureSessionSharedOutput_remove;
     ACaptureSessionOutput_free;
   local:
     *;

diff --git a/camera/tests/CameraBinderTests.cpp b/camera/tests/CameraBinderTests.cpp
index 51d9214..1de7013 100644
--- a/camera/tests/CameraBinderTests.cpp
+++ b/camera/tests/CameraBinderTests.cpp

@@ -198,9 +198,11 @@
 
 
     virtual binder::Status onResultReceived(const CameraMetadata& metadata,
-            const CaptureResultExtras& resultExtras) {
+            const CaptureResultExtras& resultExtras,
+            const std::vector<PhysicalCaptureResultInfo>& physicalResultInfos) {
         (void) metadata;
         (void) resultExtras;
+        (void) physicalResultInfos;
         Mutex::Autolock l(mLock);
         mLastStatus = SENT_RESULT;
         mStatusesHit.push_back(mLastStatus);
@@ -317,6 +319,9 @@
     EXPECT_TRUE(res.isOk()) << res;
 
     EXPECT_EQ(numCameras, static_cast<const int>(statuses.size()));
+    for (const auto &it : statuses) {
+        listener->onStatusChanged(it.status, String16(it.cameraId));
+    }
 
     for (int32_t i = 0; i < numCameras; i++) {
         String16 cameraId = String16(String8::format("%d", i));
@@ -421,6 +426,9 @@
         serviceListener = new TestCameraServiceListener();
         std::vector<hardware::CameraStatus> statuses;
         service->addListener(serviceListener, &statuses);
+        for (const auto &it : statuses) {
+            serviceListener->onStatusChanged(it.status, String16(it.cameraId));
+        }
         service->getNumberOfCameras(hardware::ICameraService::CAMERA_TYPE_BACKWARD_COMPATIBLE,
                 &numCameras);
     }
@@ -439,8 +447,9 @@
     ASSERT_NOT_NULL(service);
     EXPECT_TRUE(serviceListener->waitForNumCameras(numCameras));
     for (int32_t i = 0; i < numCameras; i++) {
+        String8 cameraId8 = String8::format("%d", i);
         // Make sure we're available, or skip device tests otherwise
-        String16 cameraId(String8::format("%d",i));
+        String16 cameraId(cameraId8);
         int32_t s = serviceListener->getStatus(cameraId);
         EXPECT_EQ(hardware::ICameraServiceListener::STATUS_PRESENT, s);
         if (s != hardware::ICameraServiceListener::STATUS_PRESENT) {
@@ -476,7 +485,8 @@
         res = device->createStream(output, &streamId);
         EXPECT_TRUE(res.isOk()) << res;
         EXPECT_LE(0, streamId);
-        res = device->endConfigure(/*isConstrainedHighSpeed*/ false);
+        CameraMetadata sessionParams;
+        res = device->endConfigure(/*isConstrainedHighSpeed*/ false, sessionParams);
         EXPECT_TRUE(res.isOk()) << res;
         EXPECT_FALSE(callbacks->hadError());
 
@@ -487,7 +497,7 @@
         EXPECT_TRUE(res.isOk()) << res;
 
         hardware::camera2::CaptureRequest request;
-        request.mMetadata = requestTemplate;
+        request.mPhysicalCameraSettings.push_back({cameraId8.string(), requestTemplate});
         request.mSurfaceList.add(surface);
         request.mIsReprocess = false;
         int64_t lastFrameNumber = 0;
@@ -514,7 +524,7 @@
                 /*out*/&requestTemplate);
         EXPECT_TRUE(res.isOk()) << res;
         hardware::camera2::CaptureRequest request2;
-        request2.mMetadata = requestTemplate;
+        request2.mPhysicalCameraSettings.push_back({cameraId8.string(), requestTemplate});
         request2.mSurfaceList.add(surface);
         request2.mIsReprocess = false;
         callbacks->clearStatus();
@@ -547,10 +557,10 @@
         EXPECT_TRUE(res.isOk()) << res;
         android::hardware::camera2::CaptureRequest request3;
         android::hardware::camera2::CaptureRequest request4;
-        request3.mMetadata = requestTemplate;
+        request3.mPhysicalCameraSettings.push_back({cameraId8.string(), requestTemplate});
         request3.mSurfaceList.add(surface);
         request3.mIsReprocess = false;
-        request4.mMetadata = requestTemplate2;
+        request4.mPhysicalCameraSettings.push_back({cameraId8.string(), requestTemplate2});
         request4.mSurfaceList.add(surface);
         request4.mIsReprocess = false;
         std::vector<hardware::camera2::CaptureRequest> requestList;
@@ -574,7 +584,7 @@
         EXPECT_TRUE(res.isOk()) << res;
         res = device->deleteStream(streamId);
         EXPECT_TRUE(res.isOk()) << res;
-        res = device->endConfigure(/*isConstrainedHighSpeed*/ false);
+        res = device->endConfigure(/*isConstrainedHighSpeed*/ false, sessionParams);
         EXPECT_TRUE(res.isOk()) << res;
 
         sleep(/*second*/1); // allow some time for errors to show up, if any
@@ -584,3 +594,62 @@
     }
 
 };
+
+TEST_F(CameraClientBinderTest, CheckBinderCaptureRequest) {
+    sp<CaptureRequest> requestOriginal, requestParceled;
+    sp<IGraphicBufferProducer> gbProducer;
+    sp<IGraphicBufferConsumer> gbConsumer;
+    BufferQueue::createBufferQueue(&gbProducer, &gbConsumer);
+    sp<Surface> surface(new Surface(gbProducer, /*controlledByApp*/false));
+    Vector<sp<Surface>> surfaceList;
+    surfaceList.push_back(surface);
+    std::string physicalDeviceId1 = "0";
+    std::string physicalDeviceId2 = "1";
+    CameraMetadata physicalDeviceSettings1, physicalDeviceSettings2;
+    uint8_t intent1 = ANDROID_CONTROL_CAPTURE_INTENT_PREVIEW;
+    uint8_t intent2 = ANDROID_CONTROL_CAPTURE_INTENT_VIDEO_RECORD;
+    EXPECT_EQ(OK, physicalDeviceSettings1.update(ANDROID_CONTROL_CAPTURE_INTENT, &intent1, 1));
+    EXPECT_EQ(OK, physicalDeviceSettings2.update(ANDROID_CONTROL_CAPTURE_INTENT, &intent2, 1));
+
+    requestParceled = new CaptureRequest();
+    Parcel p;
+    EXPECT_TRUE(requestParceled->readFromParcel(&p) != OK);
+    p.writeInt32(0);
+    p.setDataPosition(0);
+    EXPECT_TRUE(requestParceled->readFromParcel(&p) != OK);
+    p.freeData();
+    p.writeInt32(-1);
+    p.setDataPosition(0);
+    EXPECT_TRUE(requestParceled->readFromParcel(&p) != OK);
+    p.freeData();
+    p.writeInt32(1);
+    p.setDataPosition(0);
+    EXPECT_TRUE(requestParceled->readFromParcel(&p) != OK);
+
+    requestOriginal = new CaptureRequest();
+    requestOriginal->mPhysicalCameraSettings.push_back({physicalDeviceId1,
+            physicalDeviceSettings1});
+    requestOriginal->mPhysicalCameraSettings.push_back({physicalDeviceId2,
+            physicalDeviceSettings2});
+    requestOriginal->mSurfaceList.push_back(surface);
+    requestOriginal->mIsReprocess = false;
+    requestOriginal->mSurfaceConverted = false;
+
+    p.freeData();
+    EXPECT_TRUE(requestOriginal->writeToParcel(&p) == OK);
+    p.setDataPosition(0);
+    EXPECT_TRUE(requestParceled->readFromParcel(&p) == OK);
+    EXPECT_EQ(requestParceled->mIsReprocess, false);
+    EXPECT_FALSE(requestParceled->mSurfaceList.empty());
+    EXPECT_EQ(2u, requestParceled->mPhysicalCameraSettings.size());
+    auto it = requestParceled->mPhysicalCameraSettings.begin();
+    EXPECT_EQ(physicalDeviceId1, it->id);
+    EXPECT_TRUE(it->settings.exists(ANDROID_CONTROL_CAPTURE_INTENT));
+    auto entry = it->settings.find(ANDROID_CONTROL_CAPTURE_INTENT);
+    EXPECT_EQ(entry.data.u8[0], intent1);
+    it++;
+    EXPECT_EQ(physicalDeviceId2, it->id);
+    EXPECT_TRUE(it->settings.exists(ANDROID_CONTROL_CAPTURE_INTENT));
+    entry = it->settings.find(ANDROID_CONTROL_CAPTURE_INTENT);
+    EXPECT_EQ(entry.data.u8[0], intent2);
+};

diff --git a/camera/tests/CameraZSLTests.cpp b/camera/tests/CameraZSLTests.cpp
index ecca354..02c6e2a 100644
--- a/camera/tests/CameraZSLTests.cpp
+++ b/camera/tests/CameraZSLTests.cpp

@@ -256,10 +256,10 @@
         ASSERT_TRUE(nullptr != surfaceControl.get());
         ASSERT_TRUE(surfaceControl->isValid());
 
-        SurfaceComposerClient::openGlobalTransaction();
-        ASSERT_EQ(NO_ERROR, surfaceControl->setLayer(0x7fffffff));
-        ASSERT_EQ(NO_ERROR, surfaceControl->show());
-        SurfaceComposerClient::closeGlobalTransaction();
+        SurfaceComposerClient::Transaction{}
+                .setLayer(surfaceControl, 0x7fffffff)
+                .show(surfaceControl)
+                .apply();
 
         previewSurface = surfaceControl->getSurface();
         ASSERT_TRUE(previewSurface != NULL);

diff --git a/camera/tests/VendorTagDescriptorTests.cpp b/camera/tests/VendorTagDescriptorTests.cpp
index 75cfb73..0ee358d 100644
--- a/camera/tests/VendorTagDescriptorTests.cpp
+++ b/camera/tests/VendorTagDescriptorTests.cpp

@@ -142,6 +142,7 @@
     EXPECT_EQ(OK, vDescOriginal->writeToParcel(&p));
     p.setDataPosition(0);
 
+    vDescParceled = new VendorTagDescriptor();
     ASSERT_EQ(OK, vDescParceled->readFromParcel(&p));
 
     // Ensure consistent tag count

diff --git a/cmds/screenrecord/Android.mk b/cmds/screenrecord/Android.mk
index 7aa684a..5e83ed6 100644
--- a/cmds/screenrecord/Android.mk
+++ b/cmds/screenrecord/Android.mk

@@ -25,8 +25,8 @@
 	Program.cpp
 
 LOCAL_SHARED_LIBRARIES := \
-	libstagefright libmedia libutils libbinder libstagefright_foundation \
-	libjpeg libgui libcutils liblog libEGL libGLESv2
+	libstagefright libmedia libmedia_omx libutils libbinder libstagefright_foundation \
+	libjpeg libui libgui libcutils liblog libEGL libGLESv2
 
 LOCAL_C_INCLUDES := \
 	frameworks/av/media/libstagefright \

diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index bc32bbe..d1859d1 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp

@@ -50,6 +50,7 @@
 #include <media/stagefright/MediaCodec.h>
 #include <media/stagefright/MediaErrors.h>
 #include <media/stagefright/MediaMuxer.h>
+#include <media/stagefright/PersistentSurface.h>
 #include <media/ICrypto.h>
 #include <media/MediaCodecBuffer.h>
 
@@ -70,9 +71,11 @@
 static bool gVerbose = false;           // chatty on stdout
 static bool gRotate = false;            // rotate 90 degrees
 static bool gMonotonicTime = false;     // use system monotonic time for timestamps
+static bool gPersistentSurface = false; // use persistent surface
 static enum {
     FORMAT_MP4, FORMAT_H264, FORMAT_FRAMES, FORMAT_RAW_FRAMES
 } gOutputFormat = FORMAT_MP4;           // data format for output
+static AString gCodecName = "";         // codec name override
 static bool gSizeSpecified = false;     // was size explicitly requested?
 static bool gWantInfoScreen = false;    // do we want initial info screen?
 static bool gWantFrameTime = false;     // do we want times on each frame?
@@ -132,18 +135,11 @@
                 strerror(errno));
         return err;
     }
+    signal(SIGPIPE, SIG_IGN);
     return NO_ERROR;
 }
 
 /*
- * Returns "true" if the device is rotated 90 degrees.
- */
-static bool isDeviceRotated(int orientation) {
-    return orientation != DISPLAY_ORIENTATION_0 &&
-            orientation != DISPLAY_ORIENTATION_180;
-}
-
-/*
  * Configures and starts the MediaCodec encoder.  Obtains an input surface
  * from the codec.
  */
@@ -154,6 +150,7 @@
     if (gVerbose) {
         printf("Configuring recorder for %dx%d %s at %.2fMbps\n",
                 gVideoWidth, gVideoHeight, kMimeTypeAvc, gBitRate / 1000000.0);
+        fflush(stdout);
     }
 
     sp<AMessage> format = new AMessage;
@@ -169,11 +166,21 @@
     looper->setName("screenrecord_looper");
     looper->start();
     ALOGV("Creating codec");
-    sp<MediaCodec> codec = MediaCodec::CreateByType(looper, kMimeTypeAvc, true);
-    if (codec == NULL) {
-        fprintf(stderr, "ERROR: unable to create %s codec instance\n",
-                kMimeTypeAvc);
-        return UNKNOWN_ERROR;
+    sp<MediaCodec> codec;
+    if (gCodecName.empty()) {
+        codec = MediaCodec::CreateByType(looper, kMimeTypeAvc, true);
+        if (codec == NULL) {
+            fprintf(stderr, "ERROR: unable to create %s codec instance\n",
+                    kMimeTypeAvc);
+            return UNKNOWN_ERROR;
+        }
+    } else {
+        codec = MediaCodec::CreateByComponentName(looper, gCodecName);
+        if (codec == NULL) {
+            fprintf(stderr, "ERROR: unable to create %s codec instance\n",
+                    gCodecName.c_str());
+            return UNKNOWN_ERROR;
+        }
     }
 
     err = codec->configure(format, NULL, NULL,
@@ -187,10 +194,18 @@
 
     ALOGV("Creating encoder input surface");
     sp<IGraphicBufferProducer> bufferProducer;
-    err = codec->createInputSurface(&bufferProducer);
+    if (gPersistentSurface) {
+        sp<PersistentSurface> surface = MediaCodec::CreatePersistentInputSurface();
+        bufferProducer = surface->getBufferProducer();
+        err = codec->setInputSurface(surface);
+    } else {
+        err = codec->createInputSurface(&bufferProducer);
+    }
     if (err != NO_ERROR) {
         fprintf(stderr,
-            "ERROR: unable to create encoder input surface (err=%d)\n", err);
+            "ERROR: unable to %s encoder input surface (err=%d)\n",
+            gPersistentSurface ? "set" : "create",
+            err);
         codec->release();
         return err;
     }
@@ -213,26 +228,17 @@
  * Sets the display projection, based on the display dimensions, video size,
  * and device orientation.
  */
-static status_t setDisplayProjection(const sp<IBinder>& dpy,
+static status_t setDisplayProjection(
+        SurfaceComposerClient::Transaction& t,
+        const sp<IBinder>& dpy,
         const DisplayInfo& mainDpyInfo) {
 
     // Set the region of the layer stack we're interested in, which in our
-    // case is "all of it".  If the app is rotated (so that the width of the
-    // app is based on the height of the display), reverse width/height.
-    bool deviceRotated = isDeviceRotated(mainDpyInfo.orientation);
-    uint32_t sourceWidth, sourceHeight;
-    if (!deviceRotated) {
-        sourceWidth = mainDpyInfo.w;
-        sourceHeight = mainDpyInfo.h;
-    } else {
-        ALOGV("using rotated width/height");
-        sourceHeight = mainDpyInfo.w;
-        sourceWidth = mainDpyInfo.h;
-    }
-    Rect layerStackRect(sourceWidth, sourceHeight);
+    // case is "all of it".
+    Rect layerStackRect(mainDpyInfo.w, mainDpyInfo.h);
 
     // We need to preserve the aspect ratio of the display.
-    float displayAspect = (float) sourceHeight / (float) sourceWidth;
+    float displayAspect = (float) mainDpyInfo.h / (float) mainDpyInfo.w;
 
 
     // Set the way we map the output onto the display surface (which will
@@ -273,13 +279,15 @@
         if (gRotate) {
             printf("Rotated content area is %ux%u at offset x=%d y=%d\n",
                     outHeight, outWidth, offY, offX);
+            fflush(stdout);
         } else {
             printf("Content area is %ux%u at offset x=%d y=%d\n",
                     outWidth, outHeight, offX, offY);
+            fflush(stdout);
         }
     }
 
-    SurfaceComposerClient::setDisplayProjection(dpy,
+    t.setDisplayProjection(dpy,
             gRotate ? DISPLAY_ORIENTATION_90 : DISPLAY_ORIENTATION_0,
             layerStackRect, displayRect);
     return NO_ERROR;
@@ -295,11 +303,11 @@
     sp<IBinder> dpy = SurfaceComposerClient::createDisplay(
             String8("ScreenRecorder"), false /*secure*/);
 
-    SurfaceComposerClient::openGlobalTransaction();
-    SurfaceComposerClient::setDisplaySurface(dpy, bufferProducer);
-    setDisplayProjection(dpy, mainDpyInfo);
-    SurfaceComposerClient::setDisplayLayerStack(dpy, 0);    // default stack
-    SurfaceComposerClient::closeGlobalTransaction();
+    SurfaceComposerClient::Transaction t;
+    t.setDisplaySurface(dpy, bufferProducer);
+    setDisplayProjection(t, dpy, mainDpyInfo);
+    t.setDisplayLayerStack(dpy, 0);    // default stack
+    t.apply();
 
     *pDisplayHandle = dpy;
 
@@ -307,6 +315,22 @@
 }
 
 /*
+ * Set the main display width and height to the actual width and height
+ */
+static status_t getActualDisplaySize(const sp<IBinder>& mainDpy, DisplayInfo* mainDpyInfo) {
+    Rect viewport;
+    status_t err = SurfaceComposerClient::getDisplayViewport(mainDpy, &viewport);
+    if (err != NO_ERROR) {
+        fprintf(stderr, "ERROR: unable to get display viewport\n");
+        return err;
+    }
+    mainDpyInfo->w = viewport.width();
+    mainDpyInfo->h = viewport.height();
+
+    return NO_ERROR;
+}
+
+/*
  * Runs the MediaCodec encoder, sending the output to the MediaMuxer.  The
  * input frames are coming from the virtual display as fast as SurfaceFlinger
  * wants to send them.
@@ -344,6 +368,7 @@
         if (systemTime(CLOCK_MONOTONIC) > endWhenNsec) {
             if (gVerbose) {
                 printf("Time limit reached\n");
+                fflush(stdout);
             }
             break;
         }
@@ -375,14 +400,22 @@
                     // useful stuff is hard to get at without a Dalvik VM.
                     err = SurfaceComposerClient::getDisplayInfo(mainDpy,
                             &mainDpyInfo);
-                    if (err != NO_ERROR) {
+                    if (err == NO_ERROR) {
+                        err = getActualDisplaySize(mainDpy, &mainDpyInfo);
+                        if (err != NO_ERROR) {
+                            fprintf(stderr, "ERROR: unable to set actual display size\n");
+                            return err;
+                        }
+
+                        if (orientation != mainDpyInfo.orientation) {
+                            ALOGD("orientation changed, now %d", mainDpyInfo.orientation);
+                            SurfaceComposerClient::Transaction t;
+                            setDisplayProjection(t, virtualDpy, mainDpyInfo);
+                            t.apply();
+                            orientation = mainDpyInfo.orientation;
+                        }
+                    } else {
                         ALOGW("getDisplayInfo(main) failed: %d", err);
-                    } else if (orientation != mainDpyInfo.orientation) {
-                        ALOGD("orientation changed, now %d", mainDpyInfo.orientation);
-                        SurfaceComposerClient::openGlobalTransaction();
-                        setDisplayProjection(virtualDpy, mainDpyInfo);
-                        SurfaceComposerClient::closeGlobalTransaction();
-                        orientation = mainDpyInfo.orientation;
                     }
                 }
 
@@ -481,6 +514,7 @@
         printf("Encoder stopping; recorded %u frames in %" PRId64 " seconds\n",
                 debugNumFrames, nanoseconds_to_seconds(
                         systemTime(CLOCK_MONOTONIC) - startWhenNsec));
+        fflush(stdout);
     }
     return NO_ERROR;
 }
@@ -523,6 +557,10 @@
     return rawFp;
 }
 
+static inline uint32_t floorToEven(uint32_t num) {
+    return num & ~1;
+}
+
 /*
  * Main "do work" start point.
  *
@@ -550,18 +588,26 @@
         fprintf(stderr, "ERROR: unable to get display characteristics\n");
         return err;
     }
+
+    err = getActualDisplaySize(mainDpy, &mainDpyInfo);
+    if (err != NO_ERROR) {
+        fprintf(stderr, "ERROR: unable to set actual display size\n");
+        return err;
+    }
+
     if (gVerbose) {
         printf("Main display is %dx%d @%.2ffps (orientation=%u)\n",
                 mainDpyInfo.w, mainDpyInfo.h, mainDpyInfo.fps,
                 mainDpyInfo.orientation);
+        fflush(stdout);
     }
 
-    bool rotated = isDeviceRotated(mainDpyInfo.orientation);
+    // Encoder can't take odd number as config
     if (gVideoWidth == 0) {
-        gVideoWidth = rotated ? mainDpyInfo.h : mainDpyInfo.w;
+        gVideoWidth = floorToEven(mainDpyInfo.w);
     }
     if (gVideoHeight == 0) {
-        gVideoHeight = rotated ? mainDpyInfo.w : mainDpyInfo.h;
+        gVideoHeight = floorToEven(mainDpyInfo.h);
     }
 
     // Configure and start the encoder.
@@ -621,6 +667,7 @@
         }
         if (gVerbose) {
             printf("Bugreport overlay created\n");
+            fflush(stdout);
         }
     } else {
         // Use the encoder's input surface as the virtual display surface.
@@ -713,6 +760,7 @@
 
         if (gVerbose) {
             printf("Stopping encoder and muxer\n");
+            fflush(stdout);
         }
     }
 
@@ -759,6 +807,7 @@
             printf(" %s", argv[i]);
         }
         putchar('\n');
+        fflush(stdout);
     }
 
     pid_t pid = fork();
@@ -896,7 +945,9 @@
         { "show-frame-time",    no_argument,        NULL, 'f' },
         { "rotate",             no_argument,        NULL, 'r' },
         { "output-format",      required_argument,  NULL, 'o' },
+        { "codec-name",         required_argument,  NULL, 'N' },
         { "monotonic-time",     no_argument,        NULL, 'm' },
+        { "persistent-surface", no_argument,        NULL, 'p' },
         { NULL,                 0,                  NULL, 0 }
     };
 
@@ -976,9 +1027,15 @@
                 return 2;
             }
             break;
+        case 'N':
+            gCodecName = optarg;
+            break;
         case 'm':
             gMonotonicTime = true;
             break;
+        case 'p':
+            gPersistentSurface = true;
+            break;
         default:
             if (ic != '?') {
                 fprintf(stderr, "getopt_long returned unexpected value 0x%x\n", ic);

diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index f647ffd..c7619af 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk

@@ -8,9 +8,9 @@
         SineSource.cpp
 
 LOCAL_SHARED_LIBRARIES := \
-        libstagefright libmedia libutils libbinder libstagefright_foundation \
-        libjpeg libgui libcutils liblog \
-        libhidlmemory \
+        libstagefright libmedia libmedia_omx libmediaextractor libutils libbinder \
+        libstagefright_foundation libjpeg libui libgui libcutils liblog \
+        libhidlbase \
         android.hardware.media.omx@1.0 \
 
 LOCAL_C_INCLUDES:= \
@@ -36,7 +36,8 @@
         record.cpp
 
 LOCAL_SHARED_LIBRARIES := \
-        libstagefright libmedia liblog libutils libbinder libstagefright_foundation
+        libstagefright libmedia libmediaextractor liblog libutils libbinder \
+        libstagefright_foundation
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/media/libstagefright \
@@ -60,7 +61,8 @@
         recordvideo.cpp
 
 LOCAL_SHARED_LIBRARIES := \
-        libstagefright libmedia liblog libutils libbinder libstagefright_foundation
+        libstagefright libmedia libmediaextractor liblog libutils libbinder \
+        libstagefright_foundation
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/media/libstagefright \
@@ -85,7 +87,8 @@
         audioloop.cpp
 
 LOCAL_SHARED_LIBRARIES := \
-        libstagefright libmedia liblog libutils libbinder libstagefright_foundation
+        libstagefright libmedia libmediaextractor liblog libutils libbinder \
+        libstagefright_foundation
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/media/libstagefright \
@@ -107,8 +110,8 @@
         stream.cpp    \
 
 LOCAL_SHARED_LIBRARIES := \
-        libstagefright liblog libutils libbinder libgui \
-        libstagefright_foundation libmedia libcutils
+        libstagefright liblog libutils libbinder libui libgui \
+        libstagefright_foundation libmedia libcutils libmediaextractor
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/media/libstagefright \
@@ -132,7 +135,7 @@
 
 LOCAL_SHARED_LIBRARIES := \
         libstagefright liblog libutils libbinder libstagefright_foundation \
-        libmedia libaudioclient libgui libcutils
+        libmedia libmedia_omx libaudioclient libui libgui libcutils
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/media/libstagefright \
@@ -163,6 +166,8 @@
         libbinder \
         libstagefright_foundation \
         libmedia \
+        libmedia_omx \
+        libui \
         libgui \
         libcutils \
         libRScpp \
@@ -199,7 +204,7 @@
 
 LOCAL_SHARED_LIBRARIES := \
         libstagefright liblog libutils libbinder libstagefright_foundation \
-        libcutils libc
+        libcutils libc libmediaextractor
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/media/libstagefright \

diff --git a/cmds/stagefright/SineSource.cpp b/cmds/stagefright/SineSource.cpp
index cad8caf..0ecc16c 100644
--- a/cmds/stagefright/SineSource.cpp
+++ b/cmds/stagefright/SineSource.cpp

@@ -4,6 +4,7 @@
 
 #include <media/stagefright/MediaBufferGroup.h>
 #include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaBuffer.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MetaData.h>
 
@@ -59,10 +60,10 @@
 }
 
 status_t SineSource::read(
-        MediaBuffer **out, const ReadOptions * /* options */) {
+        MediaBufferBase **out, const ReadOptions * /* options */) {
     *out = NULL;
 
-    MediaBuffer *buffer;
+    MediaBufferBase *buffer;
     status_t err = mGroup->acquire_buffer(&buffer);
 
     if (err != OK) {
@@ -88,7 +89,7 @@
         x += k;
     }
 
-    buffer->meta_data()->setInt64(
+    buffer->meta_data().setInt64(
             kKeyTime, ((int64_t)mPhase * 1000000) / mSampleRate);
 
     mPhase += numFramesPerBuffer;

diff --git a/cmds/stagefright/SineSource.h b/cmds/stagefright/SineSource.h
index be05661..1817291 100644
--- a/cmds/stagefright/SineSource.h
+++ b/cmds/stagefright/SineSource.h

@@ -2,7 +2,7 @@
 
 #define SINE_SOURCE_H_
 
-#include <media/stagefright/MediaSource.h>
+#include <media/MediaSource.h>
 #include <utils/Compat.h>
 
 namespace android {
@@ -18,7 +18,7 @@
     virtual sp<MetaData> getFormat();
 
     virtual status_t read(
-            MediaBuffer **out, const ReadOptions *options = NULL);
+            MediaBufferBase **out, const ReadOptions *options = NULL);
 
 protected:
     virtual ~SineSource();

diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index ed44b4d..d4f2e8d 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp

@@ -14,6 +14,10 @@
  * limitations under the License.
  */
 
+#define LOG_NDEBUG 0
+#define LOG_TAG "audioloop"
+#include <utils/Log.h>
+
 #include <sys/types.h>
 #include <sys/stat.h>
 #include <fcntl.h>
@@ -29,7 +33,6 @@
 #include <media/stagefright/AudioSource.h>
 #include <media/stagefright/MediaCodecSource.h>
 #include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MetaData.h>
 #include <media/stagefright/SimpleDecodingSource.h>
 #include "SineSource.h"
 
@@ -37,11 +40,13 @@
 
 static void usage(const char* name)
 {
-    fprintf(stderr, "Usage: %s [-d du.ration] [-m] [-w] [<output-file>]\n", name);
+    fprintf(stderr, "Usage: %s [-d du.ration] [-m] [-w] [-N name] [<output-file>]\n", name);
     fprintf(stderr, "Encodes either a sine wave or microphone input to AMR format\n");
     fprintf(stderr, "    -d    duration in seconds, default 5 seconds\n");
     fprintf(stderr, "    -m    use microphone for input, default sine source\n");
     fprintf(stderr, "    -w    use AMR wideband (default narrowband)\n");
+    fprintf(stderr, "    -N    name of the encoder; must be set with -M\n");
+    fprintf(stderr, "    -M    media type of the encoder; must be set with -N\n");
     fprintf(stderr, "    <output-file> output file for AMR encoding,"
             " if unspecified, decode to speaker.\n");
 }
@@ -54,8 +59,10 @@
     bool outputWBAMR = false;
     bool playToSpeaker = true;
     const char* fileOut = NULL;
+    AString name;
+    AString mediaType;
     int ch;
-    while ((ch = getopt(argc, argv, "d:mw")) != -1) {
+    while ((ch = getopt(argc, argv, "d:mwN:M:")) != -1) {
         switch (ch) {
         case 'd':
             duration = atoi(optarg);
@@ -66,6 +73,12 @@
         case 'w':
             outputWBAMR = true;
             break;
+        case 'N':
+            name.setTo(optarg);
+            break;
+        case 'M':
+            mediaType.setTo(optarg);
+            break;
         default:
             usage(argv[0]);
             return -1;
@@ -76,8 +89,18 @@
     if (argc == 1) {
         fileOut = argv[0];
     }
-    const int32_t kSampleRate = outputWBAMR ? 16000 : 8000;
-    const int32_t kBitRate = outputWBAMR ? 16000 : 8000;
+    if ((name.empty() && !mediaType.empty()) || (!name.empty() && mediaType.empty())) {
+        fprintf(stderr, "-N and -M must be set together\n");
+        usage(argv[0]);
+        return -1;
+    }
+    if (!name.empty() && fileOut != NULL) {
+        fprintf(stderr, "-N and -M cannot be used with <output file>\n");
+        usage(argv[0]);
+        return -1;
+    }
+    int32_t sampleRate = !name.empty() ? 44100 : outputWBAMR ? 16000 : 8000;
+    int32_t bitRate = sampleRate;
 
     android::ProcessState::self()->startThreadPool();
     sp<MediaSource> source;
@@ -87,22 +110,27 @@
         source = new AudioSource(
                 AUDIO_SOURCE_MIC,
                 String16(),
-                kSampleRate,
+                sampleRate,
                 channels);
     } else {
         // use a sine source at 500 hz.
-        source = new SineSource(kSampleRate, channels);
+        source = new SineSource(sampleRate, channels);
     }
 
     sp<AMessage> meta = new AMessage;
-    meta->setString(
-            "mime",
-            outputWBAMR ? MEDIA_MIMETYPE_AUDIO_AMR_WB
-                    : MEDIA_MIMETYPE_AUDIO_AMR_NB);
+    if (name.empty()) {
+        meta->setString(
+                "mime",
+                outputWBAMR ? MEDIA_MIMETYPE_AUDIO_AMR_WB
+                        : MEDIA_MIMETYPE_AUDIO_AMR_NB);
+    } else {
+        meta->setString("mime", mediaType);
+        meta->setString("testing-name", name);
+    }
 
     meta->setInt32("channel-count", channels);
-    meta->setInt32("sample-rate", kSampleRate);
-    meta->setInt32("bitrate", kBitRate);
+    meta->setInt32("sample-rate", sampleRate);
+    meta->setInt32("bitrate", bitRate);
     int32_t maxInputSize;
     if (source->getFormat()->findInt32(kKeyMaxInputSize, &maxInputSize)) {
         meta->setInt32("max-input-size", maxInputSize);
@@ -112,7 +140,7 @@
     looper->setName("audioloop");
     looper->start();
 
-    sp<IMediaSource> encoder = MediaCodecSource::Create(looper, meta, source);
+    sp<MediaSource> encoder = MediaCodecSource::Create(looper, meta, source);
 
     if (fileOut != NULL) {
         // target file specified, write encoded AMR output
@@ -128,19 +156,20 @@
         writer->stop();
     } else {
         // otherwise decode to speaker
-        sp<IMediaSource> decoder = SimpleDecodingSource::Create(encoder);
+        sp<MediaSource> decoder = SimpleDecodingSource::Create(encoder);
 
         if (playToSpeaker) {
-            AudioPlayer *player = new AudioPlayer(NULL);
-            player->setSource(decoder);
-            player->start();
+            AudioPlayer player(NULL);
+            player.setSource(decoder);
+            player.start();
             sleep(duration);
 
+ALOGI("Line: %d", __LINE__);
             decoder.clear(); // must clear |decoder| otherwise delete player will hang.
-            delete player; // there is no player->stop()...
+ALOGI("Line: %d", __LINE__);
         } else {
             CHECK_EQ(decoder->start(), (status_t)OK);
-            MediaBuffer* buffer;
+            MediaBufferBase* buffer;
             while (decoder->read(&buffer) == OK) {
                 // do something with buffer (save it eventually?)
                 // need to stop after some count though...
@@ -151,6 +180,7 @@
             }
             CHECK_EQ(decoder->stop(), (status_t)OK);
         }
+ALOGI("Line: %d", __LINE__);
     }
 
     return 0;

diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index 3108a67..6a58467 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp

@@ -430,10 +430,10 @@
         CHECK(control != NULL);
         CHECK(control->isValid());
 
-        SurfaceComposerClient::openGlobalTransaction();
-        CHECK_EQ(control->setLayer(INT_MAX), (status_t)OK);
-        CHECK_EQ(control->show(), (status_t)OK);
-        SurfaceComposerClient::closeGlobalTransaction();
+        SurfaceComposerClient::Transaction{}
+                 .setLayer(control, INT_MAX)
+                 .show(control)
+                 .apply();
 
         surface = control->getSurface();
         CHECK(surface != NULL);

diff --git a/cmds/stagefright/mediafilter.cpp b/cmds/stagefright/mediafilter.cpp
index f219e69..f24d2dd 100644
--- a/cmds/stagefright/mediafilter.cpp
+++ b/cmds/stagefright/mediafilter.cpp

@@ -764,10 +764,10 @@
         CHECK(control != NULL);
         CHECK(control->isValid());
 
-        SurfaceComposerClient::openGlobalTransaction();
-        CHECK_EQ((status_t)OK, control->setLayer(INT_MAX));
-        CHECK_EQ((status_t)OK, control->show());
-        SurfaceComposerClient::closeGlobalTransaction();
+        SurfaceComposerClient::Transaction{}
+                .setLayer(control, INT_MAX)
+                .show(control)
+                .apply();
 
         surface = control->getSurface();
         CHECK(surface != NULL);

diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index 94c2e96..44b0015 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp

@@ -17,6 +17,7 @@
 #include "SineSource.h"
 
 #include <binder/ProcessState.h>
+#include <media/MediaExtractor.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/ALooper.h>
 #include <media/stagefright/foundation/AMessage.h>
@@ -27,7 +28,7 @@
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MediaCodecSource.h>
 #include <media/stagefright/MetaData.h>
-#include <media/stagefright/MediaExtractor.h>
+#include <media/stagefright/MediaExtractorFactory.h>
 #include <media/stagefright/MPEG4Writer.h>
 #include <media/stagefright/SimpleDecodingSource.h>
 #include <media/MediaPlayerInterface.h>
@@ -120,7 +121,7 @@
     sp<MediaSource> source;
 
     sp<MediaExtractor> extractor =
-        MediaExtractor::Create(new FileSource(filename));
+        MediaExtractorFactory::Create(new FileSource(filename));
     if (extractor == NULL) {
         return NULL;
     }
@@ -320,14 +321,14 @@
     looper->setName("record");
     looper->start();
 
-    sp<IMediaSource> encoder =
+    sp<MediaSource> encoder =
         MediaCodecSource::Create(looper, encMeta, audioSource);
 
     encoder->start();
 
     int32_t n = 0;
     status_t err;
-    MediaBuffer *buffer;
+    MediaBufferBase *buffer;
     while ((err = encoder->read(&buffer)) == OK) {
         printf(".");
         fflush(stdout);

diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index 7a3c842..a63b9b9 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp

@@ -90,7 +90,7 @@
     }
 
     virtual status_t read(
-            MediaBuffer **buffer, const MediaSource::ReadOptions *options __unused) {
+            MediaBufferBase **buffer, const MediaSource::ReadOptions *options __unused) {
 
         if (mNumFramesOutput % 10 == 0) {
             fprintf(stderr, ".");
@@ -114,8 +114,8 @@
         x = x >= 0xa0 ? 0x60 : x + 1;
 #endif
         (*buffer)->set_range(0, mSize);
-        (*buffer)->meta_data()->clear();
-        (*buffer)->meta_data()->setInt64(
+        (*buffer)->meta_data().clear();
+        (*buffer)->meta_data().setInt64(
                 kKeyTime, (mNumFramesOutput * 1000000) / mFrameRate);
         ++mNumFramesOutput;
 
@@ -303,7 +303,7 @@
     looper->setName("recordvideo");
     looper->start();
 
-    sp<IMediaSource> encoder =
+    sp<MediaSource> encoder =
         MediaCodecSource::Create(
                 looper, enc_meta, source, NULL /* consumer */,
                 preferSoftwareCodec ? MediaCodecSource::FLAG_PREFER_SOFTWARE_CODEC : 0);

diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index d70282b..61fc897 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp

@@ -31,9 +31,11 @@
 
 #include <binder/IServiceManager.h>
 #include <binder/ProcessState.h>
+#include <media/DataSource.h>
+#include <media/MediaExtractor.h>
+#include <media/MediaSource.h>
 #include <media/ICrypto.h>
 #include <media/IMediaHTTPService.h>
-#include <media/IMediaCodecService.h>
 #include <media/IMediaPlayerService.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/foundation/ALooper.h>
@@ -41,14 +43,14 @@
 #include <media/stagefright/foundation/AUtils.h>
 #include "include/NuCachedSource2.h"
 #include <media/stagefright/AudioPlayer.h>
-#include <media/stagefright/DataSource.h>
+#include <media/stagefright/DataSourceFactory.h>
 #include <media/stagefright/JPEGSource.h>
+#include <media/stagefright/InterfaceUtils.h>
 #include <media/stagefright/MediaCodec.h>
 #include <media/stagefright/MediaCodecList.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/MediaExtractor.h>
-#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MediaExtractorFactory.h>
 #include <media/stagefright/MetaData.h>
 #include <media/stagefright/SimpleDecodingSource.h>
 #include <media/stagefright/Utils.h>
@@ -65,7 +67,6 @@
 #include <gui/SurfaceComposerClient.h>
 
 #include <android/hardware/media/omx/1.0/IOmx.h>
-#include <media/omx/1.0/WOmx.h>
 
 using namespace android;
 
@@ -79,6 +80,7 @@
 static bool gDisplayHistogram;
 static bool showProgress = true;
 static String8 gWriteMP4Filename;
+static String8 gComponentNameOverride;
 
 static sp<ANativeWindow> gSurface;
 
@@ -141,14 +143,14 @@
     }
 }
 
-static void dumpSource(const sp<IMediaSource> &source, const String8 &filename) {
+static void dumpSource(const sp<MediaSource> &source, const String8 &filename) {
     FILE *out = fopen(filename.string(), "wb");
 
     CHECK_EQ((status_t)OK, source->start());
 
     status_t err;
     for (;;) {
-        MediaBuffer *mbuf;
+        MediaBufferBase *mbuf;
         err = source->read(&mbuf);
 
         if (err == INFO_FORMAT_CHANGED) {
@@ -174,13 +176,13 @@
     out = NULL;
 }
 
-static void playSource(sp<IMediaSource> &source) {
+static void playSource(sp<MediaSource> &source) {
     sp<MetaData> meta = source->getFormat();
 
     const char *mime;
     CHECK(meta->findCString(kKeyMIMEType, &mime));
 
-    sp<IMediaSource> rawSource;
+    sp<MediaSource> rawSource;
     if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_RAW, mime)) {
         rawSource = source;
     } else {
@@ -192,7 +194,10 @@
             CHECK(!gPreferSoftwareCodec);
             flags |= MediaCodecList::kHardwareCodecsOnly;
         }
-        rawSource = SimpleDecodingSource::Create(source, flags, gSurface);
+        rawSource = SimpleDecodingSource::Create(
+                source, flags, gSurface,
+                gComponentNameOverride.isEmpty() ? nullptr : gComponentNameOverride.c_str(),
+                !gComponentNameOverride.isEmpty());
         if (rawSource == NULL) {
             return;
         }
@@ -229,7 +234,7 @@
         CHECK(meta->findInt64(kKeyDuration, &durationUs));
 
         status_t err;
-        MediaBuffer *buffer;
+        MediaBufferBase *buffer;
         MediaSource::ReadOptions options;
         int64_t seekTimeUs = -1;
         for (;;) {
@@ -248,7 +253,7 @@
                 shouldSeek = true;
             } else {
                 int64_t timestampUs;
-                CHECK(buffer->meta_data()->findInt64(kKeyTime, &timestampUs));
+                CHECK(buffer->meta_data().findInt64(kKeyTime, &timestampUs));
 
                 bool failed = false;
 
@@ -316,7 +321,7 @@
     while (numIterationsLeft-- > 0) {
         long numFrames = 0;
 
-        MediaBuffer *buffer;
+        MediaBufferBase *buffer;
 
         for (;;) {
             int64_t startDecodeUs = getNowUs();
@@ -404,14 +409,14 @@
 ////////////////////////////////////////////////////////////////////////////////
 
 struct DetectSyncSource : public MediaSource {
-    explicit DetectSyncSource(const sp<IMediaSource> &source);
+    explicit DetectSyncSource(const sp<MediaSource> &source);
 
     virtual status_t start(MetaData *params = NULL);
     virtual status_t stop();
     virtual sp<MetaData> getFormat();
 
     virtual status_t read(
-            MediaBuffer **buffer, const ReadOptions *options);
+            MediaBufferBase **buffer, const ReadOptions *options);
 
 private:
     enum StreamType {
@@ -421,14 +426,14 @@
         OTHER,
     };
 
-    sp<IMediaSource> mSource;
+    sp<MediaSource> mSource;
     StreamType mStreamType;
     bool mSawFirstIDRFrame;
 
     DISALLOW_EVIL_CONSTRUCTORS(DetectSyncSource);
 };
 
-DetectSyncSource::DetectSyncSource(const sp<IMediaSource> &source)
+DetectSyncSource::DetectSyncSource(const sp<MediaSource> &source)
     : mSource(source),
       mStreamType(OTHER),
       mSawFirstIDRFrame(false) {
@@ -460,7 +465,7 @@
     return mSource->getFormat();
 }
 
-static bool isIDRFrame(MediaBuffer *buffer) {
+static bool isIDRFrame(MediaBufferBase *buffer) {
     const uint8_t *data =
         (const uint8_t *)buffer->data() + buffer->range_offset();
     size_t size = buffer->range_length();
@@ -477,7 +482,7 @@
 }
 
 status_t DetectSyncSource::read(
-        MediaBuffer **buffer, const ReadOptions *options) {
+        MediaBufferBase **buffer, const ReadOptions *options) {
     for (;;) {
         status_t err = mSource->read(buffer, options);
 
@@ -487,12 +492,12 @@
 
         if (mStreamType == AVC) {
             bool isIDR = isIDRFrame(*buffer);
-            (*buffer)->meta_data()->setInt32(kKeyIsSyncFrame, isIDR);
+            (*buffer)->meta_data().setInt32(kKeyIsSyncFrame, isIDR);
             if (isIDR) {
                 mSawFirstIDRFrame = true;
             }
         } else {
-            (*buffer)->meta_data()->setInt32(kKeyIsSyncFrame, true);
+            (*buffer)->meta_data().setInt32(kKeyIsSyncFrame, true);
         }
 
         if (mStreamType != AVC || mSawFirstIDRFrame) {
@@ -510,7 +515,7 @@
 ////////////////////////////////////////////////////////////////////////////////
 
 static void writeSourcesToMP4(
-        Vector<sp<IMediaSource> > &sources, bool syncInfoPresent) {
+        Vector<sp<MediaSource> > &sources, bool syncInfoPresent) {
 #if 0
     sp<MPEG4Writer> writer =
         new MPEG4Writer(gWriteMP4Filename.string());
@@ -528,7 +533,7 @@
     writer->setMaxFileDuration(60000000ll);
 
     for (size_t i = 0; i < sources.size(); ++i) {
-        sp<IMediaSource> source = sources.editItemAt(i);
+        sp<MediaSource> source = sources.editItemAt(i);
 
         CHECK_EQ(writer->addSource(
                     syncInfoPresent ? source : new DetectSyncSource(source)),
@@ -545,7 +550,7 @@
     writer->stop();
 }
 
-static void performSeekTest(const sp<IMediaSource> &source) {
+static void performSeekTest(const sp<MediaSource> &source) {
     CHECK_EQ((status_t)OK, source->start());
 
     int64_t durationUs;
@@ -557,7 +562,7 @@
         options.setSeekTo(
                 seekTimeUs, MediaSource::ReadOptions::SEEK_PREVIOUS_SYNC);
 
-        MediaBuffer *buffer;
+        MediaBufferBase *buffer;
         status_t err;
         for (;;) {
             err = source->read(&buffer, &options);
@@ -586,7 +591,7 @@
 
         if (err == OK) {
             int64_t timeUs;
-            CHECK(buffer->meta_data()->findInt64(kKeyTime, &timeUs));
+            CHECK(buffer->meta_data().findInt64(kKeyTime, &timeUs));
 
             printf("%" PRId64 "\t%" PRId64 "\t%" PRId64 "\n",
                    seekTimeUs, timeUs, seekTimeUs - timeUs);
@@ -617,6 +622,7 @@
     fprintf(stderr, "       -o playback audio\n");
     fprintf(stderr, "       -w(rite) filename (write to .mp4 file)\n");
     fprintf(stderr, "       -k seek test\n");
+    fprintf(stderr, "       -N(ame) of the component\n");
     fprintf(stderr, "       -x display a histogram of decoding times/fps "
                     "(video only)\n");
     fprintf(stderr, "       -q don't show progress indicator\n");
@@ -702,7 +708,7 @@
     sp<ALooper> looper;
 
     int res;
-    while ((res = getopt(argc, argv, "haqn:lm:b:ptsrow:kxSTd:D:")) >= 0) {
+    while ((res = getopt(argc, argv, "haqn:lm:b:ptsrow:kN:xSTd:D:")) >= 0) {
         switch (res) {
             case 'a':
             {
@@ -731,6 +737,12 @@
                 break;
             }
 
+            case 'N':
+            {
+                gComponentNameOverride.setTo(optarg);
+                break;
+            }
+
             case 'l':
             {
                 listComponents = true;
@@ -881,7 +893,7 @@
                 VideoFrame *frame = (VideoFrame *)mem->pointer();
 
                 CHECK_EQ(writeJpegFile("/sdcard/out.jpg",
-                            (uint8_t *)frame + sizeof(VideoFrame),
+                            frame->getFlattenedData(),
                             frame->mWidth, frame->mHeight), 0);
             }
 
@@ -909,37 +921,24 @@
     }
 
     if (listComponents) {
-        sp<IOMX> omx;
-        if (property_get_bool("persist.media.treble_omx", true)) {
-            using namespace ::android::hardware::media::omx::V1_0;
-            sp<IOmx> tOmx = IOmx::getService();
+        using ::android::hardware::hidl_vec;
+        using ::android::hardware::hidl_string;
+        using namespace ::android::hardware::media::omx::V1_0;
+        sp<IOmx> omx = IOmx::getService();
+        CHECK(omx.get() != nullptr);
 
-            CHECK(tOmx.get() != NULL);
-
-            omx = new utils::LWOmx(tOmx);
-        } else {
-            sp<IServiceManager> sm = defaultServiceManager();
-            sp<IBinder> binder = sm->getService(String16("media.codec"));
-            sp<IMediaCodecService> service = interface_cast<IMediaCodecService>(binder);
-
-            CHECK(service.get() != NULL);
-
-            omx = service->getOMX();
-        }
-        CHECK(omx.get() != NULL);
-
-        List<IOMX::ComponentInfo> list;
-        omx->listNodes(&list);
-
-        for (List<IOMX::ComponentInfo>::iterator it = list.begin();
-             it != list.end(); ++it) {
-            printf("%s\t Roles: ", (*it).mName.string());
-            for (List<String8>::iterator itRoles = (*it).mRoles.begin() ;
-                    itRoles != (*it).mRoles.end() ; ++itRoles) {
-                printf("%s\t", (*itRoles).string());
-            }
-            printf("\n");
-        }
+        hidl_vec<IOmx::ComponentInfo> nodeList;
+        auto transStatus = omx->listNodes([](
+                const auto& status, const auto& nodeList) {
+                    CHECK(status == Status::OK);
+                    for (const auto& info : nodeList) {
+                        printf("%s\t Roles: ", info.mName.c_str());
+                        for (const auto& role : info.mRoles) {
+                            printf("%s\t", role.c_str());
+                        }
+                    }
+                });
+        CHECK(transStatus.isOk());
     }
 
     sp<SurfaceComposerClient> composerClient;
@@ -960,10 +959,10 @@
             CHECK(control != NULL);
             CHECK(control->isValid());
 
-            SurfaceComposerClient::openGlobalTransaction();
-            CHECK_EQ(control->setLayer(INT_MAX), (status_t)OK);
-            CHECK_EQ(control->show(), (status_t)OK);
-            SurfaceComposerClient::closeGlobalTransaction();
+            SurfaceComposerClient::Transaction{}
+                    .setLayer(control, INT_MAX)
+                    .show(control)
+                    .apply();
 
             gSurface = control->getSurface();
             CHECK(gSurface != NULL);
@@ -988,7 +987,7 @@
         const char *filename = argv[k];
 
         sp<DataSource> dataSource =
-            DataSource::CreateFromURI(NULL /* httpService */, filename);
+            DataSourceFactory::CreateFromURI(NULL /* httpService */, filename);
 
         if (strncasecmp(filename, "sine:", 5) && dataSource == NULL) {
             fprintf(stderr, "Unable to create data source.\n");
@@ -1002,8 +1001,8 @@
             isJPEG = true;
         }
 
-        Vector<sp<IMediaSource> > mediaSources;
-        sp<IMediaSource> mediaSource;
+        Vector<sp<MediaSource> > mediaSources;
+        sp<MediaSource> mediaSource;
 
         if (isJPEG) {
             mediaSource = new JPEGSource(dataSource);
@@ -1022,7 +1021,7 @@
                 mediaSources.push(mediaSource);
             }
         } else {
-            sp<IMediaExtractor> extractor = MediaExtractor::Create(dataSource);
+            sp<IMediaExtractor> extractor = MediaExtractorFactory::Create(dataSource);
 
             if (extractor == NULL) {
                 fprintf(stderr, "could not create extractor.\n");
@@ -1049,7 +1048,8 @@
                 bool haveAudio = false;
                 bool haveVideo = false;
                 for (size_t i = 0; i < numTracks; ++i) {
-                    sp<IMediaSource> source = extractor->getTrack(i);
+                    sp<MediaSource> source = CreateMediaSourceFromIMediaSource(
+                            extractor->getTrack(i));
                     if (source == nullptr) {
                         fprintf(stderr, "skip NULL track %zu, track count %zu.\n", i, numTracks);
                         continue;
@@ -1084,7 +1084,7 @@
                             i, MediaExtractor::kIncludeExtensiveMetaData);
 
                     if (meta == NULL) {
-                        break;
+                        continue;
                     }
                     const char *mime;
                     meta->findCString(kKeyMIMEType, &mime);
@@ -1115,7 +1115,7 @@
                            thumbTimeUs, thumbTimeUs / 1E6);
                 }
 
-                mediaSource = extractor->getTrack(i);
+                mediaSource = CreateMediaSourceFromIMediaSource(extractor->getTrack(i));
                 if (mediaSource == nullptr) {
                     fprintf(stderr, "skip NULL track %zu, total tracks %zu.\n", i, numTracks);
                     return -1;
@@ -1128,7 +1128,7 @@
         } else if (dumpStream) {
             dumpSource(mediaSource, dumpStreamFilename);
         } else if (dumpPCMStream) {
-            sp<IMediaSource> decSource = SimpleDecodingSource::Create(mediaSource);
+            sp<MediaSource> decSource = SimpleDecodingSource::Create(mediaSource);
             dumpSource(decSource, dumpStreamFilename);
         } else if (seekTest) {
             performSeekTest(mediaSource);

diff --git a/cmds/stagefright/stream.cpp b/cmds/stagefright/stream.cpp
index 2e1d240..b0199d8 100644
--- a/cmds/stagefright/stream.cpp
+++ b/cmds/stagefright/stream.cpp

@@ -21,15 +21,18 @@
 #include <binder/ProcessState.h>
 #include <cutils/properties.h> // for property_get
 
+#include <media/DataSource.h>
 #include <media/IMediaHTTPService.h>
 #include <media/IStreamSource.h>
+#include <media/MediaExtractor.h>
 #include <media/mediaplayer.h>
+#include <media/MediaSource.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/DataSource.h>
+#include <media/stagefright/DataSourceFactory.h>
+#include <media/stagefright/InterfaceUtils.h>
 #include <media/stagefright/MPEG2TSWriter.h>
-#include <media/stagefright/MediaExtractor.h>
-#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MediaExtractorFactory.h>
 #include <media/stagefright/MetaData.h>
 
 #include <binder/IServiceManager.h>
@@ -161,11 +164,11 @@
     : mCurrentBufferIndex(-1),
       mCurrentBufferOffset(0) {
     sp<DataSource> dataSource =
-        DataSource::CreateFromURI(NULL /* httpService */, filename);
+        DataSourceFactory::CreateFromURI(NULL /* httpService */, filename);
 
     CHECK(dataSource != NULL);
 
-    sp<IMediaExtractor> extractor = MediaExtractor::Create(dataSource);
+    sp<IMediaExtractor> extractor = MediaExtractorFactory::Create(dataSource);
     CHECK(extractor != NULL);
 
     mWriter = new MPEG2TSWriter(
@@ -182,7 +185,7 @@
             continue;
         }
 
-        sp<IMediaSource> track = extractor->getTrack(i);
+        sp<MediaSource> track = CreateMediaSourceFromIMediaSource(extractor->getTrack(i));
         if (track == nullptr) {
             fprintf(stderr, "skip NULL track %zu, total tracks %zu\n", i, numTracks);
             continue;
@@ -335,10 +338,10 @@
     CHECK(control != NULL);
     CHECK(control->isValid());
 
-    SurfaceComposerClient::openGlobalTransaction();
-    CHECK_EQ(control->setLayer(INT_MAX), (status_t)OK);
-    CHECK_EQ(control->show(), (status_t)OK);
-    SurfaceComposerClient::closeGlobalTransaction();
+    SurfaceComposerClient::Transaction{}
+            .setLayer(control, INT_MAX)
+            .show(control)
+            .apply();
 
     sp<Surface> surface = control->getSurface();
     CHECK(surface != NULL);

diff --git a/drm/common/Android.bp b/drm/common/Android.bp
index 0098c89..1552c3f 100644
--- a/drm/common/Android.bp
+++ b/drm/common/Android.bp

@@ -33,6 +33,8 @@
         "ReadWriteUtils.cpp",
     ],
 
+    cflags: ["-Wall", "-Werror"],
+
     static_libs: ["libbinder"],
 
     export_include_dirs: ["include"],

diff --git a/drm/libdrmframework/plugins/common/util/Android.bp b/drm/libdrmframework/plugins/common/util/Android.bp
index 0c0b6f2..7372eb7 100644
--- a/drm/libdrmframework/plugins/common/util/Android.bp
+++ b/drm/libdrmframework/plugins/common/util/Android.bp

@@ -19,5 +19,7 @@
 
     srcs: ["src/MimeTypeUtil.cpp"],
 
+    cflags: ["-Wall", "-Werror"],
+
     export_include_dirs: ["include"],
 }

diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.bp b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.bp
index 3f0f5f7..28a78aa 100644
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.bp
+++ b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.bp

@@ -21,6 +21,9 @@
         "-DUSE_64BIT_DRM_API",
         // The flag below turns on local debug printouts
         //"-DDRM_OMA_FL_ENGINE_DEBUG",
+        "-Wall",
+        "-Werror",
+        "-Wno-unused-variable",
     ],
 
     srcs: ["src/FwdLockEngine.cpp"],

diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
index 830def9..73eea89 100644
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
+++ b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp

@@ -502,8 +502,8 @@
         int retVal = FwdLockFile_CheckHeaderIntegrity(fileDesc);
         DecodeSession* decodeSession = new DecodeSession(fileDesc);
 
-        if (retVal && NULL != decodeSession) {
-            decodeSessionMap.addValue(decryptHandle->decryptId, decodeSession);
+        if (retVal && NULL != decodeSession &&
+            decodeSessionMap.addValue(decryptHandle->decryptId, decodeSession)) {
             const char *pmime= FwdLockFile_GetContentType(fileDesc);
             String8 contentType = String8(pmime == NULL ? "" : pmime);
             contentType.toLower();
@@ -513,7 +513,11 @@
             decryptHandle->decryptInfo = NULL;
             result = DRM_NO_ERROR;
         } else {
-            LOG_VERBOSE("FwdLockEngine::onOpenDecryptSession Integrity Check failed for the fd");
+            if (retVal && NULL != decodeSession) {
+              LOG_VERBOSE("FwdLockEngine::onOpenDecryptSession Integrity Check failed for the fd");
+            } else {
+              LOG_VERBOSE("FwdLockEngine::onOpenDecryptSession DecodeSesssion insertion failed");
+            }
             FwdLockFile_detach(fileDesc);
             delete decodeSession;
         }
@@ -631,7 +635,7 @@
     ssize_t size = -1;
 
     if (NULL != decryptHandle &&
-       decodeSessionMap.isCreated(decryptHandle->decryptId) &&
+        decodeSessionMap.isCreated(decryptHandle->decryptId) &&
         NULL != buffer &&
         numBytes > -1) {
         DecodeSession* session = decodeSessionMap.getValue(decryptHandle->decryptId);

diff --git a/drm/libdrmframework/plugins/forward-lock/internal-format/common/Android.bp b/drm/libdrmframework/plugins/forward-lock/internal-format/common/Android.bp
index 698f278..3be327a 100644
--- a/drm/libdrmframework/plugins/forward-lock/internal-format/common/Android.bp
+++ b/drm/libdrmframework/plugins/forward-lock/internal-format/common/Android.bp

@@ -19,6 +19,8 @@
 
     srcs: ["FwdLockGlue.c"],
 
+    cflags: ["-Wall", "-Werror"],
+
     shared_libs: ["libcrypto"],
 
     export_include_dirs: ["."],

diff --git a/drm/libdrmframework/plugins/forward-lock/internal-format/converter/Android.bp b/drm/libdrmframework/plugins/forward-lock/internal-format/converter/Android.bp
index 33f2fe0..d4e04b8 100644
--- a/drm/libdrmframework/plugins/forward-lock/internal-format/converter/Android.bp
+++ b/drm/libdrmframework/plugins/forward-lock/internal-format/converter/Android.bp

@@ -19,6 +19,8 @@
 
     srcs: ["FwdLockConv.c"],
 
+    cflags: ["-Wall", "-Werror"],
+
     shared_libs: ["libcrypto"],
     static_libs: ["libfwdlock-common"],
 

diff --git a/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/Android.bp b/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/Android.bp
index b6d7a06..0bf2737 100644
--- a/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/Android.bp
+++ b/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/Android.bp

@@ -19,6 +19,8 @@
 
     srcs: ["FwdLockFile.c"],
 
+    cflags: ["-Wall", "-Werror"],
+
     shared_libs: ["libcrypto"],
     static_libs: ["libfwdlock-common"],
 

diff --git a/drm/libdrmframework/plugins/passthru/Android.bp b/drm/libdrmframework/plugins/passthru/Android.bp
index 1dcf89c..05b6440 100644
--- a/drm/libdrmframework/plugins/passthru/Android.bp
+++ b/drm/libdrmframework/plugins/passthru/Android.bp

@@ -32,5 +32,7 @@
     cflags: [
         // Set the following flag to enable the decryption passthru flow
         //"-DENABLE_PASSTHRU_DECRYPTION",
+        "-Wall",
+        "-Werror",
     ],
 }

diff --git a/drm/libmediadrm/Android.bp b/drm/libmediadrm/Android.bp
index f906564..4991e50 100644
--- a/drm/libmediadrm/Android.bp
+++ b/drm/libmediadrm/Android.bp

@@ -2,10 +2,11 @@
 // libmediadrm
 //
 
-cc_library_shared {
+// TODO: change it back to cc_library_shared when MediaPlayer2 switches to
+// using NdkMediaDrm, instead of MediaDrm.java.
+cc_library {
     name: "libmediadrm",
 
-
     srcs: [
         "DrmPluginPath.cpp",
         "DrmSessionManager.cpp",
@@ -13,29 +14,26 @@
         "IDrm.cpp",
         "IDrmClient.cpp",
         "IMediaDrmService.cpp",
-        "PluginMetricsReporting.cpp",
         "SharedLibrary.cpp",
         "DrmHal.cpp",
         "CryptoHal.cpp",
-        "protos/plugin_metrics.proto",
     ],
 
-    proto: {
-        type: "lite",
-    },
-
     shared_libs: [
         "libbinder",
         "libcutils",
         "libdl",
         "liblog",
+        "libmediadrmmetrics_lite",
         "libmediametrics",
         "libmediautils",
+        "libprotobuf-cpp-lite",
         "libstagefright_foundation",
         "libutils",
         "android.hardware.drm@1.0",
+        "android.hardware.drm@1.1",
+        "libhidlallocatorutils",
         "libhidlbase",
-        "libhidlmemory",
         "libhidltransport",
     ],
 
@@ -44,3 +42,70 @@
         "-Wall",
     ],
 }
+
+// This is the version of the drm metrics configured for protobuf lite.
+cc_library_shared {
+    name: "libmediadrmmetrics_lite",
+    srcs: [
+        "DrmMetrics.cpp",
+        "PluginMetricsReporting.cpp",
+        "protos/metrics.proto",
+    ],
+
+    proto: {
+        export_proto_headers: true,
+        type: "lite",
+    },
+    shared_libs: [
+        "android.hardware.drm@1.0",
+        "android.hardware.drm@1.1",
+        "libbase",
+        "libbinder",
+        "libhidlbase",
+        "liblog",
+        "libmediametrics",
+        "libprotobuf-cpp-lite",
+        "libstagefright_foundation",
+        "libutils",
+    ],
+    cflags: [
+        // Suppress unused parameter and no error options. These cause problems
+        // with the when using the map type in a proto definition.
+        "-Wno-unused-parameter",
+        "-Wno-error",
+    ],
+}
+
+// This is the version of the drm metrics library configured for full protobuf.
+cc_library_shared {
+    name: "libmediadrmmetrics_full",
+    srcs: [
+        "DrmMetrics.cpp",
+        "PluginMetricsReporting.cpp",
+        "protos/metrics.proto",
+    ],
+
+    proto: {
+        export_proto_headers: true,
+        type: "full",
+    },
+    shared_libs: [
+        "android.hardware.drm@1.0",
+        "android.hardware.drm@1.1",
+        "libbase",
+        "libbinder",
+        "libhidlbase",
+        "liblog",
+        "libmediametrics",
+        "libprotobuf-cpp-full",
+        "libstagefright_foundation",
+        "libutils",
+    ],
+    cflags: [
+        // Suppress unused parameter and no error options. These cause problems
+        // when using the map type in a proto definition.
+        "-Wno-unused-parameter",
+        "-Wno-error",
+    ],
+}
+

diff --git a/drm/libmediadrm/CryptoHal.cpp b/drm/libmediadrm/CryptoHal.cpp
index 5dd2563..3035c5a 100644
--- a/drm/libmediadrm/CryptoHal.cpp
+++ b/drm/libmediadrm/CryptoHal.cpp

@@ -22,13 +22,14 @@
 #include <android/hidl/manager/1.0/IServiceManager.h>
 
 #include <binder/IMemory.h>
-#include <cutils/native_handle.h>
-#include <media/CryptoHal.h>
+#include <hidlmemory/FrameworkUtils.h>
 #include <media/hardware/CryptoAPI.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AString.h>
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/MediaErrors.h>
+#include <mediadrm/CryptoHal.h>
+
 
 using ::android::hardware::drm::V1_0::BufferType;
 using ::android::hardware::drm::V1_0::DestinationBuffer;
@@ -117,15 +118,24 @@
 
     auto manager = ::IServiceManager::getService();
     if (manager != NULL) {
-        manager->listByInterface(ICryptoFactory::descriptor,
+        manager->listByInterface(drm::V1_0::ICryptoFactory::descriptor,
                 [&factories](const hidl_vec<hidl_string> &registered) {
                     for (const auto &instance : registered) {
-                        auto factory = ICryptoFactory::getService(instance);
+                        auto factory = drm::V1_0::ICryptoFactory::getService(instance);
                         if (factory != NULL) {
+                            ALOGD("found drm@1.0 ICryptoFactory %s", instance.c_str());
                             factories.push_back(factory);
-                            ALOGI("makeCryptoFactories: factory instance %s is %s",
-                                    instance.c_str(),
-                                    factory->isRemote() ? "Remote" : "Not Remote");
+                        }
+                    }
+                }
+            );
+        manager->listByInterface(drm::V1_1::ICryptoFactory::descriptor,
+                [&factories](const hidl_vec<hidl_string> &registered) {
+                    for (const auto &instance : registered) {
+                        auto factory = drm::V1_1::ICryptoFactory::getService(instance);
+                        if (factory != NULL) {
+                            ALOGD("found drm@1.1 ICryptoFactory %s", instance.c_str());
+                            factories.push_back(factory);
                         }
                     }
                 }
@@ -136,7 +146,7 @@
         // must be in passthrough mode, load the default passthrough service
         auto passthrough = ICryptoFactory::getService();
         if (passthrough != NULL) {
-            ALOGI("makeCryptoFactories: using default crypto instance");
+            ALOGI("makeCryptoFactories: using default passthrough crypto instance");
             factories.push_back(passthrough);
         } else {
             ALOGE("Failed to find any crypto factories");
@@ -213,10 +223,14 @@
     Mutex::Autolock autoLock(mLock);
 
     if (mInitCheck != OK) {
-        return mInitCheck;
+        return false;
     }
 
-    return mPlugin->requiresSecureDecoderComponent(hidl_string(mime));
+    Return<bool> hResult = mPlugin->requiresSecureDecoderComponent(hidl_string(mime));
+    if (!hResult.isOk()) {
+        return false;
+    }
+    return hResult;
 }
 
 
@@ -227,26 +241,20 @@
  * are sent by providing an offset into the heap and a buffer size.
  */
 int32_t CryptoHal::setHeapBase(const sp<IMemoryHeap>& heap) {
+    using ::android::hardware::fromHeap;
+    using ::android::hardware::HidlMemory;
+
     if (heap == NULL) {
         ALOGE("setHeapBase(): heap is NULL");
         return -1;
     }
-    native_handle_t* nativeHandle = native_handle_create(1, 0);
-    if (!nativeHandle) {
-        ALOGE("setHeapBase(), failed to create native handle");
-        return -1;
-    }
 
     Mutex::Autolock autoLock(mLock);
 
     int32_t seqNum = mHeapSeqNum++;
-
-    int fd = heap->getHeapID();
-    nativeHandle->data[0] = fd;
-    auto hidlHandle = hidl_handle(nativeHandle);
-    auto hidlMemory = hidl_memory("ashmem", hidlHandle, heap->getSize());
+    sp<HidlMemory> hidlMemory = fromHeap(heap);
     mHeapBases.add(seqNum, HeapBase(mNextBufferId, heap->getSize()));
-    Return<void> hResult = mPlugin->setSharedBufferBase(hidlMemory, mNextBufferId++);
+    Return<void> hResult = mPlugin->setSharedBufferBase(*hidlMemory, mNextBufferId++);
     ALOGE_IF(!hResult.isOk(), "setSharedBufferBase(): remote call failed");
     return seqNum;
 }
@@ -254,7 +262,22 @@
 void CryptoHal::clearHeapBase(int32_t seqNum) {
     Mutex::Autolock autoLock(mLock);
 
-    mHeapBases.removeItem(seqNum);
+    /*
+     * Clear the remote shared memory mapping by setting the shared
+     * buffer base to a null hidl_memory.
+     *
+     * TODO: Add a releaseSharedBuffer method in a future DRM HAL
+     * API version to make this explicit.
+     */
+    ssize_t index = mHeapBases.indexOfKey(seqNum);
+    if (index >= 0) {
+        if (mPlugin != NULL) {
+            uint32_t bufferId = mHeapBases[index].getBufferId();
+            Return<void> hResult = mPlugin->setSharedBufferBase(hidl_memory(), bufferId);
+            ALOGE_IF(!hResult.isOk(), "setSharedBufferBase(): remote call failed");
+        }
+        mHeapBases.removeItem(seqNum);
+    }
 }
 
 status_t CryptoHal::toSharedBuffer(const sp<IMemory>& memory, int32_t seqNum, ::SharedBuffer* buffer) {

diff --git a/drm/libmediadrm/DrmHal.cpp b/drm/libmediadrm/DrmHal.cpp
index bc37557..cf08610 100644
--- a/drm/libmediadrm/DrmHal.cpp
+++ b/drm/libmediadrm/DrmHal.cpp

@@ -16,48 +16,81 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "DrmHal"
+#include <iomanip>
+
 #include <utils/Log.h>
 
 #include <binder/IPCThreadState.h>
 #include <binder/IServiceManager.h>
 
-#include <android/hardware/drm/1.0/IDrmFactory.h>
-#include <android/hardware/drm/1.0/IDrmPlugin.h>
 #include <android/hardware/drm/1.0/types.h>
 #include <android/hidl/manager/1.0/IServiceManager.h>
 #include <hidl/ServiceManagement.h>
 
-#include <media/DrmHal.h>
-#include <media/DrmSessionClientInterface.h>
-#include <media/DrmSessionManager.h>
+#include <media/EventMetric.h>
 #include <media/PluginMetricsReporting.h>
 #include <media/drm/DrmAPI.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AString.h>
+#include <media/stagefright/foundation/base64.h>
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/MediaErrors.h>
+#include <mediadrm/DrmHal.h>
+#include <mediadrm/DrmSessionClientInterface.h>
+#include <mediadrm/DrmSessionManager.h>
 
-using ::android::hardware::drm::V1_0::EventType;
-using ::android::hardware::drm::V1_0::IDrmFactory;
-using ::android::hardware::drm::V1_0::IDrmPlugin;
-using ::android::hardware::drm::V1_0::KeyedVector;
-using ::android::hardware::drm::V1_0::KeyRequestType;
-using ::android::hardware::drm::V1_0::KeyStatus;
-using ::android::hardware::drm::V1_0::KeyStatusType;
-using ::android::hardware::drm::V1_0::KeyType;
-using ::android::hardware::drm::V1_0::KeyValue;
-using ::android::hardware::drm::V1_0::SecureStop;
-using ::android::hardware::drm::V1_0::Status;
+using drm::V1_0::KeyedVector;
+using drm::V1_0::KeyStatusType;
+using drm::V1_0::KeyType;
+using drm::V1_0::KeyValue;
+using drm::V1_1::HdcpLevel;;
+using drm::V1_0::SecureStop;
+using drm::V1_1::SecureStopRelease;
+using drm::V1_0::SecureStopId;
+using drm::V1_1::SecurityLevel;
+using drm::V1_0::Status;
+using ::android::hardware::drm::V1_1::DrmMetricGroup;
 using ::android::hardware::hidl_array;
 using ::android::hardware::hidl_string;
 using ::android::hardware::hidl_vec;
 using ::android::hardware::Return;
 using ::android::hardware::Void;
 using ::android::hidl::manager::V1_0::IServiceManager;
+using ::android::os::PersistableBundle;
 using ::android::sp;
 
+namespace {
+
+// This constant corresponds to the PROPERTY_DEVICE_UNIQUE_ID constant
+// in the MediaDrm API.
+constexpr char kPropertyDeviceUniqueId[] = "deviceUniqueId";
+constexpr char kEqualsSign[] = "=";
+
+template<typename T>
+std::string toBase64StringNoPad(const T* data, size_t size) {
+    // Note that the base 64 conversion only works with arrays of single-byte
+    // values. If the source is empty or is not an array of single-byte values,
+    // return empty string.
+    if (size == 0 || sizeof(data[0]) != 1) {
+      return "";
+    }
+
+    android::AString outputString;
+    encodeBase64(data, size, &outputString);
+    // Remove trailing equals padding if it exists.
+    while (outputString.size() > 0 && outputString.endsWith(kEqualsSign)) {
+        outputString.erase(outputString.size() - 1, 1);
+    }
+
+    return std::string(outputString.c_str(), outputString.size());
+}
+
+}  // anonymous namespace
+
 namespace android {
 
+#define INIT_CHECK() {if (mInitCheck != OK) return mInitCheck;}
+
 static inline int getCallingPid() {
     return IPCThreadState::self()->getCallingPid();
 }
@@ -89,6 +122,42 @@
     return hidl_string(string.string());
 }
 
+static DrmPlugin::SecurityLevel toSecurityLevel(SecurityLevel level) {
+    switch(level) {
+    case SecurityLevel::SW_SECURE_CRYPTO:
+        return DrmPlugin::kSecurityLevelSwSecureCrypto;
+    case SecurityLevel::SW_SECURE_DECODE:
+        return DrmPlugin::kSecurityLevelSwSecureDecode;
+    case SecurityLevel::HW_SECURE_CRYPTO:
+        return DrmPlugin::kSecurityLevelHwSecureCrypto;
+    case SecurityLevel::HW_SECURE_DECODE:
+        return DrmPlugin::kSecurityLevelHwSecureDecode;
+    case SecurityLevel::HW_SECURE_ALL:
+        return DrmPlugin::kSecurityLevelHwSecureAll;
+    default:
+        return DrmPlugin::kSecurityLevelUnknown;
+    }
+}
+
+static DrmPlugin::HdcpLevel toHdcpLevel(HdcpLevel level) {
+    switch(level) {
+    case HdcpLevel::HDCP_NONE:
+        return DrmPlugin::kHdcpNone;
+    case HdcpLevel::HDCP_V1:
+        return DrmPlugin::kHdcpV1;
+    case HdcpLevel::HDCP_V2:
+        return DrmPlugin::kHdcpV2;
+    case HdcpLevel::HDCP_V2_1:
+        return DrmPlugin::kHdcpV2_1;
+    case HdcpLevel::HDCP_V2_2:
+        return DrmPlugin::kHdcpV2_2;
+    case HdcpLevel::HDCP_NO_OUTPUT:
+        return DrmPlugin::kHdcpNoOutput;
+    default:
+        return DrmPlugin::kHdcpLevelUnknown;
+    }
+}
+
 
 static ::KeyedVector toHidlKeyedVector(const KeyedVector<String8, String8>&
         keyedVector) {
@@ -121,6 +190,15 @@
     return secureStops;
 }
 
+static List<Vector<uint8_t>> toSecureStopIds(const hidl_vec<SecureStopId>&
+        hSecureStopIds) {
+    List<Vector<uint8_t>> secureStopIds;
+    for (size_t i = 0; i < hSecureStopIds.size(); i++) {
+        secureStopIds.push_back(toVector(hSecureStopIds[i]));
+    }
+    return secureStopIds;
+}
+
 static status_t toStatusT(Status status) {
     switch (status) {
     case Status::OK:
@@ -196,35 +274,63 @@
 }
 
 void DrmHal::closeOpenSessions() {
-    if (mPlugin != NULL) {
-        for (size_t i = 0; i < mOpenSessions.size(); i++) {
-            mPlugin->closeSession(toHidlVec(mOpenSessions[i]));
-            DrmSessionManager::Instance()->removeSession(mOpenSessions[i]);
-        }
+    Mutex::Autolock autoLock(mLock);
+    auto openSessions = mOpenSessions;
+    for (size_t i = 0; i < openSessions.size(); i++) {
+        mLock.unlock();
+        closeSession(openSessions[i]);
+        mLock.lock();
     }
     mOpenSessions.clear();
 }
 
 DrmHal::~DrmHal() {
-    closeOpenSessions();
     DrmSessionManager::Instance()->removeDrm(mDrmSessionClient);
 }
 
+void DrmHal::cleanup() {
+    closeOpenSessions();
+
+    Mutex::Autolock autoLock(mLock);
+    reportPluginMetrics();
+    reportFrameworkMetrics();
+
+    setListener(NULL);
+    mInitCheck = NO_INIT;
+
+    if (mPlugin != NULL) {
+        if (!mPlugin->setListener(NULL).isOk()) {
+            mInitCheck = DEAD_OBJECT;
+        }
+    }
+    mPlugin.clear();
+    mPluginV1_1.clear();
+}
+
 Vector<sp<IDrmFactory>> DrmHal::makeDrmFactories() {
     Vector<sp<IDrmFactory>> factories;
 
     auto manager = hardware::defaultServiceManager();
 
     if (manager != NULL) {
-        manager->listByInterface(IDrmFactory::descriptor,
+        manager->listByInterface(drm::V1_0::IDrmFactory::descriptor,
                 [&factories](const hidl_vec<hidl_string> &registered) {
                     for (const auto &instance : registered) {
-                        auto factory = IDrmFactory::getService(instance);
+                        auto factory = drm::V1_0::IDrmFactory::getService(instance);
                         if (factory != NULL) {
+                            ALOGD("found drm@1.0 IDrmFactory %s", instance.c_str());
                             factories.push_back(factory);
-                            ALOGI("makeDrmFactories: factory instance %s is %s",
-                                    instance.c_str(),
-                                    factory->isRemote() ? "Remote" : "Not Remote");
+                        }
+                    }
+                }
+            );
+        manager->listByInterface(drm::V1_1::IDrmFactory::descriptor,
+                [&factories](const hidl_vec<hidl_string> &registered) {
+                    for (const auto &instance : registered) {
+                        auto factory = drm::V1_1::IDrmFactory::getService(instance);
+                        if (factory != NULL) {
+                            ALOGD("found drm@1.1 IDrmFactory %s", instance.c_str());
+                            factories.push_back(factory);
                         }
                     }
                 }
@@ -235,7 +341,7 @@
         // must be in passthrough mode, load the default passthrough service
         auto passthrough = IDrmFactory::getService();
         if (passthrough != NULL) {
-            ALOGI("makeDrmFactories: using default drm instance");
+            ALOGI("makeDrmFactories: using default passthrough drm instance");
             factories.push_back(passthrough);
         } else {
             ALOGE("Failed to find any drm factories");
@@ -246,6 +352,8 @@
 
 sp<IDrmPlugin> DrmHal::makeDrmPlugin(const sp<IDrmFactory>& factory,
         const uint8_t uuid[16], const String8& appPackageName) {
+    mAppPackageName = appPackageName;
+    mMetrics.SetAppPackageName(appPackageName);
 
     sp<IDrmPlugin> plugin;
     Return<void> hResult = factory->createPlugin(uuid, appPackageName.string(),
@@ -284,6 +392,7 @@
 
 Return<void> DrmHal::sendEvent(EventType hEventType,
         const hidl_vec<uint8_t>& sessionId, const hidl_vec<uint8_t>& data) {
+    mMetrics.mEventCounter.Increment(hEventType);
 
     mEventLock.lock();
     sp<IDrmClient> listener = mListener;
@@ -374,12 +483,21 @@
                 break;
             }
             obj.writeInt32(type);
+            mMetrics.mKeyStatusChangeCounter.Increment(keyStatus.type);
         }
         obj.writeInt32(hasNewUsableKey);
 
         Mutex::Autolock lock(mNotifyLock);
         listener->notify(DrmPlugin::kDrmPluginEventKeysChange, 0, &obj);
+    } else {
+        // There's no listener. But we still want to count the key change
+        // events.
+        size_t nKeys = keyStatusList.size();
+        for (size_t i = 0; i < nKeys; i++) {
+            mMetrics.mKeyStatusChangeCounter.Increment(keyStatusList[i].type);
+        }
     }
+
     return Void();
 }
 
@@ -407,6 +525,9 @@
     for (size_t i = 0; i < mFactories.size(); i++) {
         if (mFactories[i]->isCryptoSchemeSupported(uuid)) {
             mPlugin = makeDrmPlugin(mFactories[i], uuid, appPackageName);
+            if (mPlugin != NULL) {
+                mPluginV1_1 = drm::V1_1::IDrmPlugin::castFrom(mPlugin);
+            }
         }
     }
 
@@ -424,46 +545,66 @@
 }
 
 status_t DrmHal::destroyPlugin() {
-    Mutex::Autolock autoLock(mLock);
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
-
-    closeOpenSessions();
-    reportMetrics();
-    setListener(NULL);
-    mInitCheck = NO_INIT;
-
-    if (mPlugin != NULL) {
-        if (!mPlugin->setListener(NULL).isOk()) {
-            mInitCheck = DEAD_OBJECT;
-        }
-    }
-    mPlugin.clear();
+    cleanup();
     return OK;
 }
 
-status_t DrmHal::openSession(Vector<uint8_t> &sessionId) {
+status_t DrmHal::openSession(DrmPlugin::SecurityLevel level,
+        Vector<uint8_t> &sessionId) {
     Mutex::Autolock autoLock(mLock);
+    INIT_CHECK();
 
-    if (mInitCheck != OK) {
-        return mInitCheck;
+    SecurityLevel hSecurityLevel;
+    bool setSecurityLevel = true;
+
+    switch(level) {
+    case DrmPlugin::kSecurityLevelSwSecureCrypto:
+        hSecurityLevel = SecurityLevel::SW_SECURE_CRYPTO;
+        break;
+    case DrmPlugin::kSecurityLevelSwSecureDecode:
+        hSecurityLevel = SecurityLevel::SW_SECURE_DECODE;
+        break;
+    case DrmPlugin::kSecurityLevelHwSecureCrypto:
+        hSecurityLevel = SecurityLevel::HW_SECURE_CRYPTO;
+        break;
+    case DrmPlugin::kSecurityLevelHwSecureDecode:
+        hSecurityLevel = SecurityLevel::HW_SECURE_DECODE;
+        break;
+    case DrmPlugin::kSecurityLevelHwSecureAll:
+        hSecurityLevel = SecurityLevel::HW_SECURE_ALL;
+        break;
+    case DrmPlugin::kSecurityLevelMax:
+        setSecurityLevel = false;
+        break;
+    default:
+        return ERROR_DRM_CANNOT_HANDLE;
     }
 
     status_t  err = UNKNOWN_ERROR;
-
     bool retry = true;
     do {
         hidl_vec<uint8_t> hSessionId;
 
-        Return<void> hResult = mPlugin->openSession(
-                [&](Status status, const hidl_vec<uint8_t>& id) {
-                    if (status == Status::OK) {
-                        sessionId = toVector(id);
+        Return<void> hResult;
+        if (mPluginV1_1 == NULL || !setSecurityLevel) {
+            hResult = mPlugin->openSession(
+                    [&](Status status,const hidl_vec<uint8_t>& id) {
+                        if (status == Status::OK) {
+                            sessionId = toVector(id);
+                        }
+                        err = toStatusT(status);
                     }
-                    err = toStatusT(status);
-                }
-            );
+                );
+        } else {
+            hResult = mPluginV1_1->openSession_1_1(hSecurityLevel,
+                    [&](Status status, const hidl_vec<uint8_t>& id) {
+                        if (status == Status::OK) {
+                            sessionId = toVector(id);
+                        }
+                        err = toStatusT(status);
+                    }
+                );
+        }
 
         if (!hResult.isOk()) {
             err = DEAD_OBJECT;
@@ -485,16 +626,16 @@
         DrmSessionManager::Instance()->addSession(getCallingPid(),
                 mDrmSessionClient, sessionId);
         mOpenSessions.push(sessionId);
+        mMetrics.SetSessionStart(sessionId);
     }
+
+    mMetrics.mOpenSessionCounter.Increment(err);
     return err;
 }
 
 status_t DrmHal::closeSession(Vector<uint8_t> const &sessionId) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     Return<Status> status = mPlugin->closeSession(toHidlVec(sessionId));
     if (status.isOk()) {
@@ -507,9 +648,12 @@
                 }
             }
         }
-        reportMetrics();
-        return toStatusT(status);
+        status_t response = toStatusT(status);
+        mMetrics.SetSessionEnd(sessionId);
+        mMetrics.mCloseSessionCounter.Increment(response);
+        return response;
     }
+    mMetrics.mCloseSessionCounter.Increment(DEAD_OBJECT);
     return DEAD_OBJECT;
 }
 
@@ -519,10 +663,8 @@
         String8> const &optionalParameters, Vector<uint8_t> &request,
         String8 &defaultUrl, DrmPlugin::KeyRequestType *keyRequestType) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
+    EventTimer<status_t> keyRequestTimer(&mMetrics.mGetKeyRequestTimeUs);
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
@@ -534,6 +676,7 @@
     } else if (keyType == DrmPlugin::kKeyType_Release) {
         hKeyType = KeyType::RELEASE;
     } else {
+        keyRequestTimer.SetAttribute(BAD_VALUE);
         return BAD_VALUE;
     }
 
@@ -541,23 +684,63 @@
 
     status_t err = UNKNOWN_ERROR;
 
+    if (mPluginV1_1 != NULL) {
+        Return<void> hResult =
+            mPluginV1_1->getKeyRequest_1_1(
+                toHidlVec(sessionId), toHidlVec(initData),
+                toHidlString(mimeType), hKeyType, hOptionalParameters,
+                [&](Status status, const hidl_vec<uint8_t>& hRequest,
+                    drm::V1_1::KeyRequestType hKeyRequestType,
+                    const hidl_string& hDefaultUrl) {
+
+            if (status == Status::OK) {
+                request = toVector(hRequest);
+                defaultUrl = toString8(hDefaultUrl);
+
+                switch (hKeyRequestType) {
+                    case drm::V1_1::KeyRequestType::INITIAL:
+                        *keyRequestType = DrmPlugin::kKeyRequestType_Initial;
+                        break;
+                    case drm::V1_1::KeyRequestType::RENEWAL:
+                        *keyRequestType = DrmPlugin::kKeyRequestType_Renewal;
+                        break;
+                    case drm::V1_1::KeyRequestType::RELEASE:
+                        *keyRequestType = DrmPlugin::kKeyRequestType_Release;
+                        break;
+                    case drm::V1_1::KeyRequestType::NONE:
+                        *keyRequestType = DrmPlugin::kKeyRequestType_None;
+                        break;
+                    case drm::V1_1::KeyRequestType::UPDATE:
+                        *keyRequestType = DrmPlugin::kKeyRequestType_Update;
+                        break;
+                    default:
+                        *keyRequestType = DrmPlugin::kKeyRequestType_Unknown;
+                        break;
+                }
+                err = toStatusT(status);
+            }
+        });
+        return hResult.isOk() ? err : DEAD_OBJECT;
+    }
+
     Return<void> hResult = mPlugin->getKeyRequest(toHidlVec(sessionId),
             toHidlVec(initData), toHidlString(mimeType), hKeyType, hOptionalParameters,
             [&](Status status, const hidl_vec<uint8_t>& hRequest,
-                    KeyRequestType hKeyRequestType, const hidl_string& hDefaultUrl) {
+                    drm::V1_0::KeyRequestType hKeyRequestType,
+                    const hidl_string& hDefaultUrl) {
 
                 if (status == Status::OK) {
                     request = toVector(hRequest);
                     defaultUrl = toString8(hDefaultUrl);
 
                     switch (hKeyRequestType) {
-                    case KeyRequestType::INITIAL:
+                    case drm::V1_0::KeyRequestType::INITIAL:
                         *keyRequestType = DrmPlugin::kKeyRequestType_Initial;
                         break;
-                    case KeyRequestType::RENEWAL:
+                    case drm::V1_0::KeyRequestType::RENEWAL:
                         *keyRequestType = DrmPlugin::kKeyRequestType_Renewal;
                         break;
-                    case KeyRequestType::RELEASE:
+                    case drm::V1_0::KeyRequestType::RELEASE:
                         *keyRequestType = DrmPlugin::kKeyRequestType_Release;
                         break;
                     default:
@@ -568,16 +751,17 @@
                 }
             });
 
-    return hResult.isOk() ? err : DEAD_OBJECT;
+    err = hResult.isOk() ? err : DEAD_OBJECT;
+    keyRequestTimer.SetAttribute(err);
+    return err;
 }
 
 status_t DrmHal::provideKeyResponse(Vector<uint8_t> const &sessionId,
         Vector<uint8_t> const &response, Vector<uint8_t> &keySetId) {
     Mutex::Autolock autoLock(mLock);
+    EventTimer<status_t> keyResponseTimer(&mMetrics.mProvideKeyResponseTimeUs);
 
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
@@ -592,41 +776,35 @@
                 err = toStatusT(status);
             }
         );
-
-    return hResult.isOk() ? err : DEAD_OBJECT;
+    err = hResult.isOk() ? err : DEAD_OBJECT;
+    keyResponseTimer.SetAttribute(err);
+    return err;
 }
 
 status_t DrmHal::removeKeys(Vector<uint8_t> const &keySetId) {
     Mutex::Autolock autoLock(mLock);
+    INIT_CHECK();
 
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
-
-    return toStatusT(mPlugin->removeKeys(toHidlVec(keySetId)));
+    Return<Status> status = mPlugin->removeKeys(toHidlVec(keySetId));
+    return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
 }
 
 status_t DrmHal::restoreKeys(Vector<uint8_t> const &sessionId,
         Vector<uint8_t> const &keySetId) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
-    return toStatusT(mPlugin->restoreKeys(toHidlVec(sessionId),
-                    toHidlVec(keySetId)));
+    Return<Status> status = mPlugin->restoreKeys(toHidlVec(sessionId),
+            toHidlVec(keySetId));
+    return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
 }
 
 status_t DrmHal::queryKeyStatus(Vector<uint8_t> const &sessionId,
         KeyedVector<String8, String8> &infoMap) const {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
@@ -650,10 +828,7 @@
         String8 const &certAuthority, Vector<uint8_t> &request,
         String8 &defaultUrl) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     status_t err = UNKNOWN_ERROR;
 
@@ -669,16 +844,15 @@
             }
         );
 
-    return hResult.isOk() ? err : DEAD_OBJECT;
+    err = hResult.isOk() ? err : DEAD_OBJECT;
+    mMetrics.mGetProvisionRequestCounter.Increment(err);
+    return err;
 }
 
 status_t DrmHal::provideProvisionResponse(Vector<uint8_t> const &response,
         Vector<uint8_t> &certificate, Vector<uint8_t> &wrappedKey) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     status_t err = UNKNOWN_ERROR;
 
@@ -693,15 +867,14 @@
             }
         );
 
-    return hResult.isOk() ? err : DEAD_OBJECT;
+    err = hResult.isOk() ? err : DEAD_OBJECT;
+    mMetrics.mProvideProvisionResponseCounter.Increment(err);
+    return err;
 }
 
 status_t DrmHal::getSecureStops(List<Vector<uint8_t>> &secureStops) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     status_t err = UNKNOWN_ERROR;
 
@@ -718,13 +891,36 @@
 }
 
 
-status_t DrmHal::getSecureStop(Vector<uint8_t> const &ssid, Vector<uint8_t> &secureStop) {
+status_t DrmHal::getSecureStopIds(List<Vector<uint8_t>> &secureStopIds) {
     Mutex::Autolock autoLock(mLock);
 
     if (mInitCheck != OK) {
         return mInitCheck;
     }
 
+    if (mPluginV1_1 == NULL) {
+        return ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    status_t err = UNKNOWN_ERROR;
+
+    Return<void> hResult = mPluginV1_1->getSecureStopIds(
+            [&](Status status, const hidl_vec<SecureStopId>& hSecureStopIds) {
+                if (status == Status::OK) {
+                    secureStopIds = toSecureStopIds(hSecureStopIds);
+                }
+                err = toStatusT(status);
+            }
+    );
+
+    return hResult.isOk() ? err : DEAD_OBJECT;
+}
+
+
+status_t DrmHal::getSecureStop(Vector<uint8_t> const &ssid, Vector<uint8_t> &secureStop) {
+    Mutex::Autolock autoLock(mLock);
+    INIT_CHECK();
+
     status_t err = UNKNOWN_ERROR;
 
     Return<void> hResult = mPlugin->getSecureStop(toHidlVec(ssid),
@@ -741,22 +937,132 @@
 
 status_t DrmHal::releaseSecureStops(Vector<uint8_t> const &ssRelease) {
     Mutex::Autolock autoLock(mLock);
+    INIT_CHECK();
 
-    if (mInitCheck != OK) {
-        return mInitCheck;
+    Return<Status> status(Status::ERROR_DRM_UNKNOWN);
+    if (mPluginV1_1 != NULL) {
+        SecureStopRelease secureStopRelease;
+        secureStopRelease.opaqueData = toHidlVec(ssRelease);
+        status = mPluginV1_1->releaseSecureStops(secureStopRelease);
+    } else {
+        status = mPlugin->releaseSecureStop(toHidlVec(ssRelease));
     }
-
-    return toStatusT(mPlugin->releaseSecureStop(toHidlVec(ssRelease)));
+    return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
 }
 
-status_t DrmHal::releaseAllSecureStops() {
+status_t DrmHal::removeSecureStop(Vector<uint8_t> const &ssid) {
     Mutex::Autolock autoLock(mLock);
 
     if (mInitCheck != OK) {
         return mInitCheck;
     }
 
-    return toStatusT(mPlugin->releaseAllSecureStops());
+    if (mPluginV1_1 == NULL) {
+        return ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    Return<Status> status = mPluginV1_1->removeSecureStop(toHidlVec(ssid));
+    return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
+}
+
+status_t DrmHal::removeAllSecureStops() {
+    Mutex::Autolock autoLock(mLock);
+    INIT_CHECK();
+
+    Return<Status> status(Status::ERROR_DRM_UNKNOWN);
+    if (mPluginV1_1 != NULL) {
+        status = mPluginV1_1->removeAllSecureStops();
+    } else {
+        status = mPlugin->releaseAllSecureStops();
+    }
+    return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
+}
+
+status_t DrmHal::getHdcpLevels(DrmPlugin::HdcpLevel *connected,
+            DrmPlugin::HdcpLevel *max) const {
+    Mutex::Autolock autoLock(mLock);
+    INIT_CHECK();
+
+    if (connected == NULL || max == NULL) {
+        return BAD_VALUE;
+    }
+    status_t err = UNKNOWN_ERROR;
+
+    if (mPluginV1_1 == NULL) {
+        return ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    *connected = DrmPlugin::kHdcpLevelUnknown;
+    *max = DrmPlugin::kHdcpLevelUnknown;
+
+    Return<void> hResult = mPluginV1_1->getHdcpLevels(
+            [&](Status status, const HdcpLevel& hConnected, const HdcpLevel& hMax) {
+                if (status == Status::OK) {
+                    *connected = toHdcpLevel(hConnected);
+                    *max = toHdcpLevel(hMax);
+                }
+                err = toStatusT(status);
+            }
+    );
+
+    return hResult.isOk() ? err : DEAD_OBJECT;
+}
+
+status_t DrmHal::getNumberOfSessions(uint32_t *open, uint32_t *max) const {
+    Mutex::Autolock autoLock(mLock);
+    INIT_CHECK();
+
+    if (open == NULL || max == NULL) {
+        return BAD_VALUE;
+    }
+    status_t err = UNKNOWN_ERROR;
+
+    *open = 0;
+    *max = 0;
+
+    if (mPluginV1_1 == NULL) {
+        return ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    Return<void> hResult = mPluginV1_1->getNumberOfSessions(
+            [&](Status status, uint32_t hOpen, uint32_t hMax) {
+                if (status == Status::OK) {
+                    *open = hOpen;
+                    *max = hMax;
+                }
+                err = toStatusT(status);
+            }
+    );
+
+    return hResult.isOk() ? err : DEAD_OBJECT;
+}
+
+status_t DrmHal::getSecurityLevel(Vector<uint8_t> const &sessionId,
+        DrmPlugin::SecurityLevel *level) const {
+    Mutex::Autolock autoLock(mLock);
+    INIT_CHECK();
+
+    if (level == NULL) {
+        return BAD_VALUE;
+    }
+    status_t err = UNKNOWN_ERROR;
+
+    if (mPluginV1_1 == NULL) {
+        return ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    *level = DrmPlugin::kSecurityLevelUnknown;
+
+    Return<void> hResult = mPluginV1_1->getSecurityLevel(toHidlVec(sessionId),
+            [&](Status status, SecurityLevel hLevel) {
+                if (status == Status::OK) {
+                    *level = toSecurityLevel(hLevel);
+                }
+                err = toStatusT(status);
+            }
+    );
+
+    return hResult.isOk() ? err : DEAD_OBJECT;
 }
 
 status_t DrmHal::getPropertyString(String8 const &name, String8 &value ) const {
@@ -767,10 +1073,7 @@
 status_t DrmHal::getPropertyStringInternal(String8 const &name, String8 &value) const {
     // This function is internal to the class and should only be called while
     // mLock is already held.
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     status_t err = UNKNOWN_ERROR;
 
@@ -794,10 +1097,7 @@
 status_t DrmHal::getPropertyByteArrayInternal(String8 const &name, Vector<uint8_t> &value ) const {
     // This function is internal to the class and should only be called while
     // mLock is already held.
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     status_t err = UNKNOWN_ERROR;
 
@@ -810,73 +1110,106 @@
             }
     );
 
-    return hResult.isOk() ? err : DEAD_OBJECT;
+    err = hResult.isOk() ? err : DEAD_OBJECT;
+    if (name == kPropertyDeviceUniqueId) {
+        mMetrics.mGetDeviceUniqueIdCounter.Increment(err);
+    }
+    return err;
 }
 
 status_t DrmHal::setPropertyString(String8 const &name, String8 const &value ) const {
     Mutex::Autolock autoLock(mLock);
+    INIT_CHECK();
 
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
-
-    Status status =  mPlugin->setPropertyString(toHidlString(name),
+    Return<Status> status = mPlugin->setPropertyString(toHidlString(name),
             toHidlString(value));
-    return toStatusT(status);
+    return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
 }
 
 status_t DrmHal::setPropertyByteArray(String8 const &name,
                                    Vector<uint8_t> const &value ) const {
     Mutex::Autolock autoLock(mLock);
+    INIT_CHECK();
 
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
-
-    Status status = mPlugin->setPropertyByteArray(toHidlString(name),
+    Return<Status> status = mPlugin->setPropertyByteArray(toHidlString(name),
             toHidlVec(value));
-    return toStatusT(status);
+    return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
 }
 
+status_t DrmHal::getMetrics(PersistableBundle* metrics) {
+    if (metrics == nullptr) {
+        return UNEXPECTED_NULL;
+    }
+    mMetrics.Export(metrics);
+
+    // Append vendor metrics if they are supported.
+    if (mPluginV1_1 != NULL) {
+        String8 vendor;
+        String8 description;
+        if (getPropertyStringInternal(String8("vendor"), vendor) != OK
+            || vendor.isEmpty()) {
+          ALOGE("Get vendor failed or is empty");
+          vendor = "NONE";
+        }
+        if (getPropertyStringInternal(String8("description"), description) != OK
+            || description.isEmpty()) {
+          ALOGE("Get description failed or is empty.");
+          description = "NONE";
+        }
+        vendor += ".";
+        vendor += description;
+
+        hidl_vec<DrmMetricGroup> pluginMetrics;
+        status_t err = UNKNOWN_ERROR;
+
+        Return<void> status = mPluginV1_1->getMetrics(
+                [&](Status status, hidl_vec<DrmMetricGroup> pluginMetrics) {
+                    if (status != Status::OK) {
+                      ALOGV("Error getting plugin metrics: %d", status);
+                    } else {
+                        PersistableBundle pluginBundle;
+                        if (MediaDrmMetrics::HidlMetricsToBundle(
+                                pluginMetrics, &pluginBundle) == OK) {
+                            metrics->putPersistableBundle(String16(vendor), pluginBundle);
+                        }
+                    }
+                    err = toStatusT(status);
+                });
+        return status.isOk() ? err : DEAD_OBJECT;
+    }
+
+    return OK;
+}
 
 status_t DrmHal::setCipherAlgorithm(Vector<uint8_t> const &sessionId,
                                  String8 const &algorithm) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
-    Status status = mPlugin->setCipherAlgorithm(toHidlVec(sessionId),
+    Return<Status> status = mPlugin->setCipherAlgorithm(toHidlVec(sessionId),
             toHidlString(algorithm));
-    return toStatusT(status);
+    return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
 }
 
 status_t DrmHal::setMacAlgorithm(Vector<uint8_t> const &sessionId,
                               String8 const &algorithm) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
-    Status status = mPlugin->setMacAlgorithm(toHidlVec(sessionId),
+    Return<Status> status = mPlugin->setMacAlgorithm(toHidlVec(sessionId),
             toHidlString(algorithm));
-    return toStatusT(status);
+    return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
 }
 
 status_t DrmHal::encrypt(Vector<uint8_t> const &sessionId,
         Vector<uint8_t> const &keyId, Vector<uint8_t> const &input,
         Vector<uint8_t> const &iv, Vector<uint8_t> &output) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
@@ -899,10 +1232,7 @@
         Vector<uint8_t> const &keyId, Vector<uint8_t> const &input,
         Vector<uint8_t> const &iv, Vector<uint8_t> &output) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
@@ -925,10 +1255,7 @@
         Vector<uint8_t> const &keyId, Vector<uint8_t> const &message,
         Vector<uint8_t> &signature) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
@@ -951,10 +1278,7 @@
         Vector<uint8_t> const &keyId, Vector<uint8_t> const &message,
         Vector<uint8_t> const &signature, bool &match) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
@@ -979,10 +1303,7 @@
         String8 const &algorithm, Vector<uint8_t> const &message,
         Vector<uint8_t> const &wrappedKey, Vector<uint8_t> &signature) {
     Mutex::Autolock autoLock(mLock);
-
-    if (mInitCheck != OK) {
-        return mInitCheck;
-    }
+    INIT_CHECK();
 
     if (!checkPermission("android.permission.ACCESS_DRM_CERTIFICATES")) {
         return -EPERM;
@@ -1007,17 +1328,7 @@
 
 void DrmHal::binderDied(const wp<IBinder> &the_late_who __unused)
 {
-    Mutex::Autolock autoLock(mLock);
-    closeOpenSessions();
-    setListener(NULL);
-    mInitCheck = NO_INIT;
-
-    if (mPlugin != NULL) {
-        if (!mPlugin->setListener(NULL).isOk()) {
-            mInitCheck = DEAD_OBJECT;
-        }
-    }
-    mPlugin.clear();
+    cleanup();
 }
 
 void DrmHal::writeByteArray(Parcel &obj, hidl_vec<uint8_t> const &vec)
@@ -1030,18 +1341,55 @@
     }
 }
 
-void DrmHal::reportMetrics() const
+void DrmHal::reportFrameworkMetrics() const
 {
-    Vector<uint8_t> metrics;
+    MediaAnalyticsItem item("mediadrm");
+    item.generateSessionID();
+    item.setPkgName(mMetrics.GetAppPackageName().c_str());
+    String8 vendor;
+    String8 description;
+    status_t result = getPropertyStringInternal(String8("vendor"), vendor);
+    if (result != OK) {
+        ALOGE("Failed to get vendor from drm plugin: %d", result);
+    } else {
+        item.setCString("vendor", vendor.c_str());
+    }
+    result = getPropertyStringInternal(String8("description"), description);
+    if (result != OK) {
+        ALOGE("Failed to get description from drm plugin: %d", result);
+    } else {
+        item.setCString("description", description.c_str());
+    }
+
+    std::string serializedMetrics;
+    result = mMetrics.GetSerializedMetrics(&serializedMetrics);
+    if (result != OK) {
+        ALOGE("Failed to serialize framework metrics: %d", result);
+    }
+    std::string b64EncodedMetrics = toBase64StringNoPad(serializedMetrics.data(),
+                                                        serializedMetrics.size());
+    if (!b64EncodedMetrics.empty()) {
+        item.setCString("serialized_metrics", b64EncodedMetrics.c_str());
+    }
+    if (!item.selfrecord()) {
+        ALOGE("Failed to self record framework metrics");
+    }
+}
+
+void DrmHal::reportPluginMetrics() const
+{
+    Vector<uint8_t> metricsVector;
     String8 vendor;
     String8 description;
     if (getPropertyStringInternal(String8("vendor"), vendor) == OK &&
             getPropertyStringInternal(String8("description"), description) == OK &&
-            getPropertyByteArrayInternal(String8("metrics"), metrics) == OK) {
-        status_t res = android::reportDrmPluginMetrics(
-                metrics, vendor, description);
+            getPropertyByteArrayInternal(String8("metrics"), metricsVector) == OK) {
+        std::string metricsString = toBase64StringNoPad(metricsVector.array(),
+                                                        metricsVector.size());
+        status_t res = android::reportDrmPluginMetrics(metricsString, vendor,
+                                                       description, mAppPackageName);
         if (res != OK) {
-            ALOGE("Metrics were retrieved but could not be reported: %i", res);
+            ALOGE("Metrics were retrieved but could not be reported: %d", res);
         }
     }
 }

diff --git a/drm/libmediadrm/DrmMetrics.cpp b/drm/libmediadrm/DrmMetrics.cpp
new file mode 100644
index 0000000..4fed707
--- /dev/null
+++ b/drm/libmediadrm/DrmMetrics.cpp

@@ -0,0 +1,422 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#define LOG_TAG "DrmMetrics"
+#include <iomanip>
+#include <utility>
+
+#include <android-base/macros.h>
+#include <media/stagefright/foundation/base64.h>
+#include <mediadrm/DrmMetrics.h>
+#include <sys/time.h>
+#include <utils/Log.h>
+#include <utils/Timers.h>
+
+#include "protos/metrics.pb.h"
+
+using ::android::String16;
+using ::android::String8;
+using ::android::drm_metrics::DrmFrameworkMetrics;
+using ::android::hardware::hidl_string;
+using ::android::hardware::hidl_vec;
+using ::android::hardware::drm::V1_0::EventType;
+using ::android::hardware::drm::V1_0::KeyStatusType;
+using ::android::hardware::drm::V1_1::DrmMetricGroup;
+using ::android::os::PersistableBundle;
+
+namespace {
+
+template <typename T> std::string GetAttributeName(T type);
+
+template <> std::string GetAttributeName<KeyStatusType>(KeyStatusType type) {
+    static const char *type_names[] = {"USABLE", "EXPIRED",
+                                       "OUTPUT_NOT_ALLOWED", "STATUS_PENDING",
+                                       "INTERNAL_ERROR"};
+    if (((size_t)type) > arraysize(type_names)) {
+        return "UNKNOWN_TYPE";
+    }
+    return type_names[(size_t)type];
+}
+
+template <> std::string GetAttributeName<EventType>(EventType type) {
+    static const char *type_names[] = {"PROVISION_REQUIRED", "KEY_NEEDED",
+                                       "KEY_EXPIRED", "VENDOR_DEFINED",
+                                       "SESSION_RECLAIMED"};
+    if (((size_t)type) > arraysize(type_names)) {
+        return "UNKNOWN_TYPE";
+    }
+    return type_names[(size_t)type];
+}
+
+template <typename T>
+void ExportCounterMetric(const android::CounterMetric<T> &counter,
+                         PersistableBundle *metrics) {
+    if (!metrics) {
+        ALOGE("metrics was unexpectedly null.");
+        return;
+    }
+    std::string success_count_name = counter.metric_name() + ".ok.count";
+    std::string error_count_name = counter.metric_name() + ".error.count";
+    std::vector<int64_t> status_values;
+    counter.ExportValues(
+        [&](const android::status_t status, const int64_t value) {
+            if (status == android::OK) {
+                metrics->putLong(android::String16(success_count_name.c_str()),
+                                 value);
+            } else {
+                int64_t total_errors(0);
+                metrics->getLong(android::String16(error_count_name.c_str()),
+                                 &total_errors);
+                metrics->putLong(android::String16(error_count_name.c_str()),
+                                 total_errors + value);
+                status_values.push_back(status);
+            }
+        });
+    if (!status_values.empty()) {
+        std::string error_list_name = counter.metric_name() + ".error.list";
+        metrics->putLongVector(android::String16(error_list_name.c_str()),
+                               status_values);
+    }
+}
+
+template <typename T>
+void ExportCounterMetricWithAttributeNames(
+    const android::CounterMetric<T> &counter, PersistableBundle *metrics) {
+    if (!metrics) {
+        ALOGE("metrics was unexpectedly null.");
+        return;
+    }
+    counter.ExportValues([&](const T &attribute, const int64_t value) {
+        std::string name = counter.metric_name() + "." +
+                           GetAttributeName(attribute) + ".count";
+        metrics->putLong(android::String16(name.c_str()), value);
+    });
+}
+
+template <typename T>
+void ExportEventMetric(const android::EventMetric<T> &event,
+                       PersistableBundle *metrics) {
+    if (!metrics) {
+        ALOGE("metrics was unexpectedly null.");
+        return;
+    }
+    std::string success_count_name = event.metric_name() + ".ok.count";
+    std::string error_count_name = event.metric_name() + ".error.count";
+    std::string timing_name = event.metric_name() + ".ok.average_time_micros";
+    std::vector<int64_t> status_values;
+    event.ExportValues([&](const android::status_t &status,
+                           const android::EventStatistics &value) {
+        if (status == android::OK) {
+            metrics->putLong(android::String16(success_count_name.c_str()),
+                             value.count);
+            metrics->putLong(android::String16(timing_name.c_str()),
+                             value.mean);
+        } else {
+            int64_t total_errors(0);
+            metrics->getLong(android::String16(error_count_name.c_str()),
+                             &total_errors);
+            metrics->putLong(android::String16(error_count_name.c_str()),
+                             total_errors + value.count);
+            status_values.push_back(status);
+        }
+    });
+    if (!status_values.empty()) {
+        std::string error_list_name = event.metric_name() + ".error.list";
+        metrics->putLongVector(android::String16(error_list_name.c_str()),
+                               status_values);
+    }
+}
+
+void ExportSessionLifespans(
+    const std::map<std::string, std::pair<int64_t, int64_t>> &mSessionLifespans,
+    PersistableBundle *metrics) {
+    if (!metrics) {
+        ALOGE("metrics was unexpectedly null.");
+        return;
+    }
+
+    if (mSessionLifespans.empty()) {
+        return;
+    }
+
+    PersistableBundle startTimesBundle;
+    PersistableBundle endTimesBundle;
+    for (auto it = mSessionLifespans.begin(); it != mSessionLifespans.end();
+         it++) {
+        String16 key(it->first.c_str(), it->first.size());
+        startTimesBundle.putLong(key, it->second.first);
+        endTimesBundle.putLong(key, it->second.second);
+    }
+    metrics->putPersistableBundle(
+        android::String16("drm.mediadrm.session_start_times_ms"),
+        startTimesBundle);
+    metrics->putPersistableBundle(
+        android::String16("drm.mediadrm.session_end_times_ms"), endTimesBundle);
+}
+
+std::string ToHexString(const android::Vector<uint8_t> &sessionId) {
+    std::ostringstream out;
+    out << std::hex << std::setfill('0');
+    for (size_t i = 0; i < sessionId.size(); i++) {
+        out << std::setw(2) << (int)(sessionId[i]);
+    }
+    return out.str();
+}
+
+template <typename CT>
+void SetValue(const String16 &name, DrmMetricGroup::ValueType type,
+              const CT &value, PersistableBundle *bundle) {
+    switch (type) {
+    case DrmMetricGroup::ValueType::INT64_TYPE:
+        bundle->putLong(name, value.int64Value);
+        break;
+    case DrmMetricGroup::ValueType::DOUBLE_TYPE:
+        bundle->putDouble(name, value.doubleValue);
+        break;
+    case DrmMetricGroup::ValueType::STRING_TYPE:
+        bundle->putString(name, String16(value.stringValue.c_str()));
+        break;
+    default:
+        ALOGE("Unexpected value type: %hhu", type);
+    }
+}
+
+inline String16 MakeIndexString(unsigned int index) {
+  std::string str("[");
+  str.append(std::to_string(index));
+  str.append("]");
+  return String16(str.c_str());
+}
+
+} // namespace
+
+namespace android {
+
+MediaDrmMetrics::MediaDrmMetrics()
+    : mOpenSessionCounter("drm.mediadrm.open_session", "status"),
+      mCloseSessionCounter("drm.mediadrm.close_session", "status"),
+      mGetKeyRequestTimeUs("drm.mediadrm.get_key_request", "status"),
+      mProvideKeyResponseTimeUs("drm.mediadrm.provide_key_response", "status"),
+      mGetProvisionRequestCounter("drm.mediadrm.get_provision_request",
+                                  "status"),
+      mProvideProvisionResponseCounter(
+          "drm.mediadrm.provide_provision_response", "status"),
+      mKeyStatusChangeCounter("drm.mediadrm.key_status_change",
+                              "key_status_type"),
+      mEventCounter("drm.mediadrm.event", "event_type"),
+      mGetDeviceUniqueIdCounter("drm.mediadrm.get_device_unique_id", "status") {
+}
+
+void MediaDrmMetrics::SetSessionStart(
+    const android::Vector<uint8_t> &sessionId) {
+    std::string sessionIdHex = ToHexString(sessionId);
+    mSessionLifespans[sessionIdHex] =
+        std::make_pair(GetCurrentTimeMs(), (int64_t)0);
+}
+
+void MediaDrmMetrics::SetSessionEnd(const android::Vector<uint8_t> &sessionId) {
+    std::string sessionIdHex = ToHexString(sessionId);
+    int64_t endTimeMs = GetCurrentTimeMs();
+    if (mSessionLifespans.find(sessionIdHex) != mSessionLifespans.end()) {
+        mSessionLifespans[sessionIdHex] =
+            std::make_pair(mSessionLifespans[sessionIdHex].first, endTimeMs);
+    } else {
+        mSessionLifespans[sessionIdHex] = std::make_pair((int64_t)0, endTimeMs);
+    }
+}
+
+void MediaDrmMetrics::Export(PersistableBundle *metrics) {
+    if (!metrics) {
+        ALOGE("metrics was unexpectedly null.");
+        return;
+    }
+    ExportCounterMetric(mOpenSessionCounter, metrics);
+    ExportCounterMetric(mCloseSessionCounter, metrics);
+    ExportEventMetric(mGetKeyRequestTimeUs, metrics);
+    ExportEventMetric(mProvideKeyResponseTimeUs, metrics);
+    ExportCounterMetric(mGetProvisionRequestCounter, metrics);
+    ExportCounterMetric(mProvideProvisionResponseCounter, metrics);
+    ExportCounterMetricWithAttributeNames(mKeyStatusChangeCounter, metrics);
+    ExportCounterMetricWithAttributeNames(mEventCounter, metrics);
+    ExportCounterMetric(mGetDeviceUniqueIdCounter, metrics);
+    ExportSessionLifespans(mSessionLifespans, metrics);
+}
+
+status_t MediaDrmMetrics::GetSerializedMetrics(std::string *serializedMetrics) {
+
+    if (!serializedMetrics) {
+        ALOGE("serializedMetrics was unexpectedly null.");
+        return UNEXPECTED_NULL;
+    }
+
+    DrmFrameworkMetrics metrics;
+
+    mOpenSessionCounter.ExportValues(
+        [&](const android::status_t status, const int64_t value) {
+            DrmFrameworkMetrics::Counter *counter =
+                metrics.add_open_session_counter();
+            counter->set_count(value);
+            counter->mutable_attributes()->set_error_code(status);
+        });
+
+    mCloseSessionCounter.ExportValues(
+        [&](const android::status_t status, const int64_t value) {
+            DrmFrameworkMetrics::Counter *counter =
+                metrics.add_close_session_counter();
+            counter->set_count(value);
+            counter->mutable_attributes()->set_error_code(status);
+        });
+
+    mGetProvisionRequestCounter.ExportValues(
+        [&](const android::status_t status, const int64_t value) {
+            DrmFrameworkMetrics::Counter *counter =
+                metrics.add_get_provisioning_request_counter();
+            counter->set_count(value);
+            counter->mutable_attributes()->set_error_code(status);
+        });
+
+    mProvideProvisionResponseCounter.ExportValues(
+        [&](const android::status_t status, const int64_t value) {
+            DrmFrameworkMetrics::Counter *counter =
+                metrics.add_provide_provisioning_response_counter();
+            counter->set_count(value);
+            counter->mutable_attributes()->set_error_code(status);
+        });
+
+    mKeyStatusChangeCounter.ExportValues(
+        [&](const KeyStatusType key_status_type, const int64_t value) {
+            DrmFrameworkMetrics::Counter *counter =
+                metrics.add_key_status_change_counter();
+            counter->set_count(value);
+            counter->mutable_attributes()->set_key_status_type(
+                (uint32_t)key_status_type);
+        });
+
+    mEventCounter.ExportValues(
+        [&](const EventType event_type, const int64_t value) {
+            DrmFrameworkMetrics::Counter *counter =
+                metrics.add_event_callback_counter();
+            counter->set_count(value);
+            counter->mutable_attributes()->set_event_type((uint32_t)event_type);
+        });
+
+    mGetDeviceUniqueIdCounter.ExportValues(
+        [&](const status_t status, const int64_t value) {
+            DrmFrameworkMetrics::Counter *counter =
+                metrics.add_get_device_unique_id_counter();
+            counter->set_count(value);
+            counter->mutable_attributes()->set_error_code(status);
+        });
+
+    mGetKeyRequestTimeUs.ExportValues(
+        [&](const status_t status, const EventStatistics &stats) {
+            DrmFrameworkMetrics::DistributionMetric *metric =
+                metrics.add_get_key_request_time_us();
+            metric->set_min(stats.min);
+            metric->set_max(stats.max);
+            metric->set_mean(stats.mean);
+            metric->set_operation_count(stats.count);
+            metric->set_variance(stats.sum_squared_deviation / stats.count);
+            metric->mutable_attributes()->set_error_code(status);
+        });
+
+    mProvideKeyResponseTimeUs.ExportValues(
+        [&](const status_t status, const EventStatistics &stats) {
+            DrmFrameworkMetrics::DistributionMetric *metric =
+                metrics.add_provide_key_response_time_us();
+            metric->set_min(stats.min);
+            metric->set_max(stats.max);
+            metric->set_mean(stats.mean);
+            metric->set_operation_count(stats.count);
+            metric->set_variance(stats.sum_squared_deviation / stats.count);
+            metric->mutable_attributes()->set_error_code(status);
+        });
+
+    for (const auto &sessionLifespan : mSessionLifespans) {
+        auto *map = metrics.mutable_session_lifetimes();
+
+        (*map)[sessionLifespan.first].set_start_time_ms(
+            sessionLifespan.second.first);
+        (*map)[sessionLifespan.first].set_end_time_ms(
+            sessionLifespan.second.second);
+    }
+
+    if (!metrics.SerializeToString(serializedMetrics)) {
+        ALOGE("Failed to serialize metrics.");
+        return UNKNOWN_ERROR;
+    }
+
+    return OK;
+}
+
+int64_t MediaDrmMetrics::GetCurrentTimeMs() {
+    struct timeval tv;
+    gettimeofday(&tv, NULL);
+    return ((int64_t)tv.tv_sec * 1000) + ((int64_t)tv.tv_usec / 1000);
+}
+
+status_t MediaDrmMetrics::HidlMetricsToBundle(
+    const hidl_vec<DrmMetricGroup> &hidlMetricGroups,
+    PersistableBundle *bundleMetricGroups) {
+    if (bundleMetricGroups == nullptr) {
+        return UNEXPECTED_NULL;
+    }
+    if (hidlMetricGroups.size() == 0) {
+        return OK;
+    }
+
+    int groupIndex = 0;
+    std::map<String16, int> indexMap;
+    for (const auto &hidlMetricGroup : hidlMetricGroups) {
+        PersistableBundle bundleMetricGroup;
+        for (const auto &hidlMetric : hidlMetricGroup.metrics) {
+            String16 metricName(hidlMetric.name.c_str());
+            PersistableBundle bundleMetric;
+            // Add metric component values.
+            for (const auto &value : hidlMetric.values) {
+                SetValue(String16(value.componentName.c_str()), value.type,
+                         value, &bundleMetric);
+            }
+            // Set metric attributes.
+            PersistableBundle bundleMetricAttributes;
+            for (const auto &attribute : hidlMetric.attributes) {
+                SetValue(String16(attribute.name.c_str()), attribute.type,
+                         attribute, &bundleMetricAttributes);
+            }
+            // Add attributes to the bundle metric.
+            bundleMetric.putPersistableBundle(String16("attributes"),
+                                              bundleMetricAttributes);
+            // Add one layer of indirection, allowing for repeated metric names.
+            PersistableBundle repeatedMetrics;
+            bundleMetricGroup.getPersistableBundle(metricName,
+                                                   &repeatedMetrics);
+            int index = indexMap[metricName];
+            repeatedMetrics.putPersistableBundle(MakeIndexString(index),
+                                                 bundleMetric);
+            indexMap[metricName] = ++index;
+
+            // Add the bundle metric to the group of metrics.
+            bundleMetricGroup.putPersistableBundle(metricName,
+                                                   repeatedMetrics);
+        }
+        // Add the bundle metric group to the collection of groups.
+        bundleMetricGroups->putPersistableBundle(MakeIndexString(groupIndex++),
+                                                 bundleMetricGroup);
+    }
+
+    return OK;
+}
+
+} // namespace android

diff --git a/drm/libmediadrm/DrmPluginPath.cpp b/drm/libmediadrm/DrmPluginPath.cpp
index c760825..ac8607c 100644
--- a/drm/libmediadrm/DrmPluginPath.cpp
+++ b/drm/libmediadrm/DrmPluginPath.cpp

@@ -19,7 +19,7 @@
 #include <utils/Log.h>
 
 #include <cutils/properties.h>
-#include <media/DrmPluginPath.h>
+#include <mediadrm/DrmPluginPath.h>
 
 namespace android {
 

diff --git a/drm/libmediadrm/DrmSessionManager.cpp b/drm/libmediadrm/DrmSessionManager.cpp
index 02270d0..375644c 100644
--- a/drm/libmediadrm/DrmSessionManager.cpp
+++ b/drm/libmediadrm/DrmSessionManager.cpp

@@ -21,9 +21,9 @@
 #include <binder/IPCThreadState.h>
 #include <binder/IProcessInfoService.h>
 #include <binder/IServiceManager.h>
-#include <media/DrmSessionManager.h>
-#include <media/DrmSessionClientInterface.h>
 #include <media/stagefright/ProcessInfo.h>
+#include <mediadrm/DrmSessionClientInterface.h>
+#include <mediadrm/DrmSessionManager.h>
 #include <unistd.h>
 #include <utils/String8.h>
 

diff --git a/drm/libmediadrm/ICrypto.cpp b/drm/libmediadrm/ICrypto.cpp
index 7931ed7..8d8d088 100644
--- a/drm/libmediadrm/ICrypto.cpp
+++ b/drm/libmediadrm/ICrypto.cpp

@@ -19,10 +19,10 @@
 #include <binder/Parcel.h>
 #include <binder/IMemory.h>
 #include <cutils/log.h>
-#include <media/ICrypto.h>
 #include <media/stagefright/MediaErrors.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AString.h>
+#include <mediadrm/ICrypto.h>
 #include <utils/Log.h>
 
 namespace android {
@@ -354,10 +354,10 @@
                 return OK;
             }
 
-            CryptoPlugin::SubSample *subSamples =
-                    new CryptoPlugin::SubSample[numSubSamples];
+            std::unique_ptr<CryptoPlugin::SubSample[]> subSamples =
+                    std::make_unique<CryptoPlugin::SubSample[]>(numSubSamples);
 
-            data.read(subSamples,
+            data.read(subSamples.get(),
                     sizeof(CryptoPlugin::SubSample) * numSubSamples);
 
             DestinationBuffer destination;
@@ -415,7 +415,7 @@
                 result = -EINVAL;
             } else {
                 result = decrypt(key, iv, mode, pattern, source, offset,
-                        subSamples, numSubSamples, destination, &errorDetailMsg);
+                        subSamples.get(), numSubSamples, destination, &errorDetailMsg);
             }
 
             reply->writeInt32(result);
@@ -434,9 +434,7 @@
                 }
             }
 
-            delete[] subSamples;
-            subSamples = NULL;
-
+            subSamples.reset();
             return OK;
         }
 

diff --git a/drm/libmediadrm/IDrm.cpp b/drm/libmediadrm/IDrm.cpp
index 8ff6e6a..509961f 100644
--- a/drm/libmediadrm/IDrm.cpp
+++ b/drm/libmediadrm/IDrm.cpp

@@ -19,10 +19,10 @@
 #include <utils/Log.h>
 
 #include <binder/Parcel.h>
-#include <media/IDrm.h>
 #include <media/stagefright/MediaErrors.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AString.h>
+#include <mediadrm/IDrm.h>
 
 namespace android {
 
@@ -46,6 +46,7 @@
     GET_PROPERTY_BYTE_ARRAY,
     SET_PROPERTY_STRING,
     SET_PROPERTY_BYTE_ARRAY,
+    GET_METRICS,
     SET_CIPHER_ALGORITHM,
     SET_MAC_ALGORITHM,
     ENCRYPT,
@@ -55,7 +56,12 @@
     VERIFY,
     SET_LISTENER,
     GET_SECURE_STOP,
-    RELEASE_ALL_SECURE_STOPS
+    REMOVE_ALL_SECURE_STOPS,
+    GET_HDCP_LEVELS,
+    GET_NUMBER_OF_SESSIONS,
+    GET_SECURITY_LEVEL,
+    REMOVE_SECURE_STOP,
+    GET_SECURE_STOP_IDS
 };
 
 struct BpDrm : public BpInterface<IDrm> {
@@ -114,9 +120,11 @@
         return reply.readInt32();
     }
 
-    virtual status_t openSession(Vector<uint8_t> &sessionId) {
+    virtual status_t openSession(DrmPlugin::SecurityLevel securityLevel,
+            Vector<uint8_t> &sessionId) {
         Parcel data, reply;
         data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
+        data.writeInt32(securityLevel);
 
         status_t status = remote()->transact(OPEN_SESSION, data, &reply);
         if (status != OK) {
@@ -297,6 +305,25 @@
         return reply.readInt32();
     }
 
+    virtual status_t getSecureStopIds(List<Vector<uint8_t> > &secureStopIds) {
+        Parcel data, reply;
+        data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
+
+        status_t status = remote()->transact(GET_SECURE_STOP_IDS, data, &reply);
+        if (status != OK) {
+            return status;
+        }
+
+        secureStopIds.clear();
+        uint32_t count = reply.readInt32();
+        for (size_t i = 0; i < count; i++) {
+            Vector<uint8_t> secureStopId;
+            readVector(reply, secureStopId);
+            secureStopIds.push_back(secureStopId);
+        }
+        return reply.readInt32();
+    }
+
     virtual status_t getSecureStop(Vector<uint8_t> const &ssid, Vector<uint8_t> &secureStop) {
         Parcel data, reply;
         data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
@@ -324,11 +351,24 @@
         return reply.readInt32();
     }
 
-    virtual status_t releaseAllSecureStops() {
+    virtual status_t removeSecureStop(Vector<uint8_t> const &ssid) {
         Parcel data, reply;
         data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
 
-        status_t status = remote()->transact(RELEASE_ALL_SECURE_STOPS, data, &reply);
+        writeVector(data, ssid);
+        status_t status = remote()->transact(REMOVE_SECURE_STOP, data, &reply);
+        if (status != OK) {
+            return status;
+        }
+
+        return reply.readInt32();
+    }
+
+    virtual status_t removeAllSecureStops() {
+        Parcel data, reply;
+        data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
+
+        status_t status = remote()->transact(REMOVE_ALL_SECURE_STOPS, data, &reply);
         if (status != OK) {
             return status;
         }
@@ -350,6 +390,65 @@
         return reply.readInt32();
     }
 
+    virtual status_t getHdcpLevels(DrmPlugin::HdcpLevel *connected,
+            DrmPlugin::HdcpLevel *max) const {
+        Parcel data, reply;
+
+        if (connected == NULL || max == NULL) {
+            return BAD_VALUE;
+        }
+
+        data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
+
+        status_t status = remote()->transact(GET_HDCP_LEVELS, data, &reply);
+        if (status != OK) {
+            return status;
+        }
+
+        *connected = static_cast<DrmPlugin::HdcpLevel>(reply.readInt32());
+        *max = static_cast<DrmPlugin::HdcpLevel>(reply.readInt32());
+        return reply.readInt32();
+    }
+
+    virtual status_t getNumberOfSessions(uint32_t *open, uint32_t *max) const {
+        Parcel data, reply;
+
+        if (open == NULL || max == NULL) {
+            return BAD_VALUE;
+        }
+
+        data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
+
+        status_t status = remote()->transact(GET_NUMBER_OF_SESSIONS, data, &reply);
+        if (status != OK) {
+            return status;
+        }
+
+        *open = reply.readInt32();
+        *max = reply.readInt32();
+        return reply.readInt32();
+    }
+
+    virtual status_t getSecurityLevel(Vector<uint8_t> const &sessionId,
+            DrmPlugin::SecurityLevel *level) const {
+        Parcel data, reply;
+
+        if (level == NULL) {
+            return BAD_VALUE;
+        }
+
+        data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
+
+        writeVector(data, sessionId);
+        status_t status = remote()->transact(GET_SECURITY_LEVEL, data, &reply);
+        if (status != OK) {
+            return status;
+        }
+
+        *level = static_cast<DrmPlugin::SecurityLevel>(reply.readInt32());
+        return reply.readInt32();
+    }
+
     virtual status_t getPropertyByteArray(String8 const &name, Vector<uint8_t> &value) const {
         Parcel data, reply;
         data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
@@ -393,6 +492,35 @@
         return reply.readInt32();
     }
 
+    virtual status_t getMetrics(os::PersistableBundle *metrics) {
+        if (metrics == NULL) {
+            return BAD_VALUE;
+        }
+        Parcel data, reply;
+        data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
+
+        status_t status = remote()->transact(GET_METRICS, data, &reply);
+        if (status != OK) {
+            return status;
+        }
+        // The reply data is ordered as
+        // 1) 32 bit integer reply followed by
+        // 2) Serialized PersistableBundle containing metrics.
+        status_t reply_status;
+        if (reply.readInt32(&reply_status) != OK
+           || reply_status != OK) {
+          ALOGE("Failed to read getMetrics response code from parcel. %d",
+                reply_status);
+          return reply_status;
+        }
+
+        status = metrics->readFromParcel(&reply);
+        if (status != OK) {
+            ALOGE("Failed to read metrics from parcel. %d", status);
+            return status;
+        }
+        return reply_status;
+    }
 
     virtual status_t setCipherAlgorithm(Vector<uint8_t> const &sessionId,
                                         String8 const &algorithm) {
@@ -615,8 +743,10 @@
         case OPEN_SESSION:
         {
             CHECK_INTERFACE(IDrm, data, reply);
+            DrmPlugin::SecurityLevel level =
+                    static_cast<DrmPlugin::SecurityLevel>(data.readInt32());
             Vector<uint8_t> sessionId;
-            status_t result = openSession(sessionId);
+            status_t result = openSession(level, sessionId);
             writeVector(reply, sessionId);
             reply->writeInt32(result);
             return OK;
@@ -761,6 +891,24 @@
             return OK;
         }
 
+        case GET_SECURE_STOP_IDS:
+        {
+            CHECK_INTERFACE(IDrm, data, reply);
+            List<Vector<uint8_t> > secureStopIds;
+            status_t result = getSecureStopIds(secureStopIds);
+            size_t count = secureStopIds.size();
+            reply->writeInt32(count);
+            List<Vector<uint8_t> >::iterator iter = secureStopIds.begin();
+            while(iter != secureStopIds.end()) {
+                size_t size = iter->size();
+                reply->writeInt32(size);
+                reply->write(iter->array(), iter->size());
+                iter++;
+            }
+            reply->writeInt32(result);
+            return OK;
+        }
+
         case GET_SECURE_STOP:
         {
             CHECK_INTERFACE(IDrm, data, reply);
@@ -781,10 +929,54 @@
             return OK;
         }
 
-        case RELEASE_ALL_SECURE_STOPS:
+        case REMOVE_SECURE_STOP:
         {
             CHECK_INTERFACE(IDrm, data, reply);
-            reply->writeInt32(releaseAllSecureStops());
+            Vector<uint8_t> ssid;
+            readVector(data, ssid);
+            reply->writeInt32(removeSecureStop(ssid));
+            return OK;
+        }
+
+        case REMOVE_ALL_SECURE_STOPS:
+        {
+            CHECK_INTERFACE(IDrm, data, reply);
+            reply->writeInt32(removeAllSecureStops());
+            return OK;
+        }
+
+        case GET_HDCP_LEVELS:
+        {
+            CHECK_INTERFACE(IDrm, data, reply);
+            DrmPlugin::HdcpLevel connected = DrmPlugin::kHdcpLevelUnknown;
+            DrmPlugin::HdcpLevel max = DrmPlugin::kHdcpLevelUnknown;
+            status_t result = getHdcpLevels(&connected, &max);
+            reply->writeInt32(connected);
+            reply->writeInt32(max);
+            reply->writeInt32(result);
+            return OK;
+        }
+
+        case GET_NUMBER_OF_SESSIONS:
+        {
+            CHECK_INTERFACE(IDrm, data, reply);
+            uint32_t open = 0, max = 0;
+            status_t result = getNumberOfSessions(&open, &max);
+            reply->writeInt32(open);
+            reply->writeInt32(max);
+            reply->writeInt32(result);
+            return OK;
+        }
+
+        case GET_SECURITY_LEVEL:
+        {
+            CHECK_INTERFACE(IDrm, data, reply);
+            Vector<uint8_t> sessionId;
+            readVector(data, sessionId);
+            DrmPlugin::SecurityLevel level = DrmPlugin::kSecurityLevelUnknown;
+            status_t result = getSecurityLevel(sessionId, &level);
+            reply->writeInt32(level);
+            reply->writeInt32(result);
             return OK;
         }
 
@@ -829,6 +1021,24 @@
             return OK;
         }
 
+        case GET_METRICS:
+        {
+            CHECK_INTERFACE(IDrm, data, reply);
+
+            os::PersistableBundle metrics;
+            status_t result = getMetrics(&metrics);
+            // The reply data is ordered as
+            // 1) 32 bit integer reply followed by
+            // 2) Serialized PersistableBundle containing metrics.
+            // Only write the metrics if the getMetrics result was
+            // OK and we successfully added the status to reply.
+            status_t parcel_result = reply->writeInt32(result);
+            if (result == OK && parcel_result == OK) {
+                parcel_result = metrics.writeToParcel(reply);
+            }
+            return parcel_result;
+        }
+
         case SET_CIPHER_ALGORITHM:
         {
             CHECK_INTERFACE(IDrm, data, reply);

diff --git a/drm/libmediadrm/IDrmClient.cpp b/drm/libmediadrm/IDrmClient.cpp
index 444201f..357de9d 100644
--- a/drm/libmediadrm/IDrmClient.cpp
+++ b/drm/libmediadrm/IDrmClient.cpp

@@ -24,7 +24,7 @@
 #include <binder/Parcel.h>
 
 #include <media/IMediaPlayerClient.h>
-#include <media/IDrmClient.h>
+#include <mediadrm/IDrmClient.h>
 
 namespace android {
 

diff --git a/drm/libmediadrm/IMediaDrmService.cpp b/drm/libmediadrm/IMediaDrmService.cpp
index 84812dc..f320d0b 100644
--- a/drm/libmediadrm/IMediaDrmService.cpp
+++ b/drm/libmediadrm/IMediaDrmService.cpp

@@ -20,9 +20,9 @@
 
 #include <binder/Parcel.h>
 #include <binder/IMemory.h>
-#include <media/ICrypto.h>
-#include <media/IDrm.h>
-#include <media/IMediaDrmService.h>
+#include <mediadrm/ICrypto.h>
+#include <mediadrm/IDrm.h>
+#include <mediadrm/IMediaDrmService.h>
 
 #include <utils/Errors.h>  // for status_t
 #include <utils/String8.h>

diff --git a/drm/libmediadrm/PluginMetricsReporting.cpp b/drm/libmediadrm/PluginMetricsReporting.cpp
index 57ff5b8..5cb48bf 100644
--- a/drm/libmediadrm/PluginMetricsReporting.cpp
+++ b/drm/libmediadrm/PluginMetricsReporting.cpp

@@ -16,83 +16,35 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "PluginMetricsReporting"
-#include <utils/Log.h>
 
 #include <media/PluginMetricsReporting.h>
 
-#include <media/MediaAnalyticsItem.h>
+#include <inttypes.h>
 
-#include "protos/plugin_metrics.pb.h"
+#include <media/MediaAnalyticsItem.h>
+#include <utils/Log.h>
+
 
 namespace android {
 
 namespace {
 
-using android::drm_metrics::MetricsGroup;
-using android::drm_metrics::MetricsGroup_Metric;
-using android::drm_metrics::MetricsGroup_Metric_MetricValue;
+constexpr char kSerializedMetricsField[] = "serialized_metrics";
 
-const char* const kParentAttribute = "/parent/external";
-
-status_t reportMetricsGroup(const MetricsGroup& metricsGroup,
-                            const String8& batchName,
-                            const int64_t* parentId) {
-    MediaAnalyticsItem analyticsItem(batchName.c_str());
+status_t reportVendorMetrics(const std::string& metrics,
+                             const String8& name,
+                             const String8& appPackageName) {
+    MediaAnalyticsItem analyticsItem(name.c_str());
     analyticsItem.generateSessionID();
-    int64_t sessionId = analyticsItem.getSessionID();
-    if (parentId != NULL) {
-        analyticsItem.setInt64(kParentAttribute, *parentId);
+
+    std::string app_package_name(appPackageName.c_str(), appPackageName.size());
+    analyticsItem.setPkgName(app_package_name);
+    if (metrics.size() > 0) {
+        analyticsItem.setCString(kSerializedMetricsField, metrics.c_str());
     }
 
-    // Report the package name.
-    if (metricsGroup.has_app_package_name()) {
-      AString app_package_name(metricsGroup.app_package_name().c_str(),
-                               metricsGroup.app_package_name().size());
-      analyticsItem.setPkgName(app_package_name);
-    }
-
-    for (int i = 0; i < metricsGroup.metric_size(); ++i) {
-        const MetricsGroup_Metric& metric = metricsGroup.metric(i);
-        if (!metric.has_name()) {
-            ALOGE("Metric with no name.");
-            return BAD_VALUE;
-        }
-
-        if (!metric.has_value()) {
-            ALOGE("Metric with no value.");
-            return BAD_VALUE;
-        }
-
-        const MetricsGroup_Metric_MetricValue& value = metric.value();
-        if (value.has_int_value()) {
-            analyticsItem.setInt64(metric.name().c_str(),
-                                   value.int_value());
-        } else if (value.has_double_value()) {
-            analyticsItem.setDouble(metric.name().c_str(),
-                                    value.double_value());
-        } else if (value.has_string_value()) {
-            analyticsItem.setCString(metric.name().c_str(),
-                                     value.string_value().c_str());
-        } else {
-            ALOGE("Metric Value with no actual value.");
-            return BAD_VALUE;
-        }
-    }
-
-    analyticsItem.setFinalized(true);
     if (!analyticsItem.selfrecord()) {
-      // Note the cast to int is because we build on 32 and 64 bit.
-      // The cast prevents a peculiar printf problem where one format cannot
-      // satisfy both.
-      ALOGE("selfrecord() returned false. sessioId %d", (int) sessionId);
-    }
-
-    for (int i = 0; i < metricsGroup.metric_sub_group_size(); ++i) {
-        const MetricsGroup& subGroup = metricsGroup.metric_sub_group(i);
-        status_t res = reportMetricsGroup(subGroup, batchName, &sessionId);
-        if (res != OK) {
-            return res;
-        }
+      ALOGE("selfrecord() returned false. sessioId %" PRId64, analyticsItem.getSessionID());
     }
 
     return OK;
@@ -114,21 +66,16 @@
 
 }  // namespace
 
-status_t reportDrmPluginMetrics(const Vector<uint8_t>& serializedMetrics,
+status_t reportDrmPluginMetrics(const std::string& b64EncodedMetrics,
                                 const String8& vendor,
-                                const String8& description) {
-    MetricsGroup root_metrics_group;
-    if (!root_metrics_group.ParseFromArray(serializedMetrics.array(),
-                                           serializedMetrics.size())) {
-        ALOGE("Failure to parse.");
-        return BAD_VALUE;
-    }
+                                const String8& description,
+                                const String8& appPackageName) {
 
     String8 name = String8::format("drm.vendor.%s.%s",
                                    sanitize(vendor).c_str(),
                                    sanitize(description).c_str());
 
-    return reportMetricsGroup(root_metrics_group, name, NULL);
+    return reportVendorMetrics(b64EncodedMetrics, name, appPackageName);
 }
 
 }  // namespace android

diff --git a/drm/libmediadrm/SharedLibrary.cpp b/drm/libmediadrm/SharedLibrary.cpp
index bebafa8..b2d635d 100644
--- a/drm/libmediadrm/SharedLibrary.cpp
+++ b/drm/libmediadrm/SharedLibrary.cpp

@@ -19,7 +19,7 @@
 
 #include <dlfcn.h>
 #include <media/stagefright/foundation/ADebug.h>
-#include <media/SharedLibrary.h>
+#include <mediadrm/SharedLibrary.h>
 #include <utils/Log.h>
 
 namespace android {

diff --git a/drm/libmediadrm/protos/metrics.proto b/drm/libmediadrm/protos/metrics.proto
new file mode 100644
index 0000000..6160e6f
--- /dev/null
+++ b/drm/libmediadrm/protos/metrics.proto

@@ -0,0 +1,107 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+syntax = "proto2";
+
+package android.drm_metrics;
+
+
+// This message contains the specific metrics captured by DrmMetrics. It is
+// used for serializing and logging metrics.
+// next id: 11.
+message DrmFrameworkMetrics {
+  // TODO: Consider using extensions.
+
+  // Attributes are associated with a recorded value. E.g. A counter may
+  // represent a count of an operation returning a specific error code. The
+  // error code will be an attribute.
+  message Attributes {
+    // Reserved for compatibility with logging proto.
+    reserved 2 to 13;
+
+    // A general purpose error code where 0 means OK.
+    optional int32 error_code = 1;
+
+    // Defined at ::android::hardware::drm::V1_0::KeyStatusType;
+    optional uint32 key_status_type = 14;
+
+    // Defined at ::android::hardware::drm::V1_0::EventType;
+    optional uint32 event_type = 15;
+  }
+
+  // The Counter message is used to store a count value with an associated
+  // Attribute.
+  message Counter {
+    optional uint64 count = 1;
+    // Represents the attributes associated with this counter instance.
+    optional Attributes attributes = 2;
+  }
+
+  // The DistributionMetric is meant to capture the moments of a normally
+  // distributed (or approximately normal) value.
+  message DistributionMetric {
+    optional float min = 1;
+    optional float max = 2;
+    optional float mean = 3;
+    optional double variance = 4;
+    optional uint64 operation_count = 5;
+
+    // Represents the attributes assocated with this distribution metric
+    // instance.
+    optional Attributes attributes = 6;
+  }
+
+  message SessionLifetime {
+    // Start time of the session in milliseconds since epoch.
+    optional uint64 start_time_ms = 1;
+    // End time of the session in milliseconds since epoch.
+    optional uint64 end_time_ms = 2;
+  }
+
+  // The count of open session operations. Each instance has a specific error
+  // code associated with it.
+  repeated Counter open_session_counter = 1;
+
+  // The count of close session operations. Each instance has a specific error
+  // code associated with it.
+  repeated Counter close_session_counter = 2;
+
+  // Count and execution time of getKeyRequest calls.
+  repeated DistributionMetric get_key_request_time_us = 3;
+
+  // Count and execution time of provideKeyResponse calls.
+  repeated DistributionMetric provide_key_response_time_us = 4;
+
+  // Count of getProvisionRequest calls.
+  repeated Counter get_provisioning_request_counter = 5;
+
+  // Count of provideProvisionResponse calls.
+  repeated Counter provide_provisioning_response_counter = 6;
+
+  // Count of key status events broken out by status type.
+  repeated Counter key_status_change_counter = 7;
+
+  // Count of events broken out by event type
+  repeated Counter event_callback_counter = 8;
+
+  // Count getPropertyByteArray calls to retrieve the device unique id.
+  repeated Counter get_device_unique_id_counter = 9;
+
+  // Session ids to lifetime (start and end time) map.
+  // Session ids are strings of hex-encoded byte arrays.
+  map<string, SessionLifetime> session_lifetimes = 10;
+}
+

diff --git a/drm/libmediadrm/protos/plugin_metrics.proto b/drm/libmediadrm/protos/plugin_metrics.proto
deleted file mode 100644
index 7e3bcf5..0000000
--- a/drm/libmediadrm/protos/plugin_metrics.proto
+++ /dev/null

@@ -1,50 +0,0 @@
-/*
- * Copyright (C) 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-syntax = "proto2";
-
-package android.drm_metrics;
-
-// need this if we are using libprotobuf-cpp-2.3.0-lite
-option optimize_for = LITE_RUNTIME;
-
-// The MetricsGroup is a collection of metric name/value pair instances
-// that can be serialized and provided to a caller.
-message MetricsGroup {
-  message Metric {
-    message MetricValue {
-      // Exactly one of the following values must be set.
-      optional int64 int_value = 1;
-      optional double double_value = 2;
-      optional string string_value = 3;
-    }
-
-    // The name of the metric. Must be valid UTF-8. Required.
-    optional string name = 1;
-
-    // The value of the metric. Required.
-    optional MetricValue value = 2;
-  }
-
-  // The list of name/value pairs of metrics.
-  repeated Metric metric = 1;
-
-  // Allow multiple sub groups of metrics.
-  repeated MetricsGroup metric_sub_group = 2;
-
-  // Name of the application package associated with the metrics.
-  optional string app_package_name = 3;
-}

diff --git a/drm/libmediadrm/tests/Android.bp b/drm/libmediadrm/tests/Android.bp
new file mode 100644
index 0000000..66c906f
--- /dev/null
+++ b/drm/libmediadrm/tests/Android.bp

@@ -0,0 +1,53 @@
+// Build definitions for unit tests.
+
+cc_test {
+    name: "CounterMetric_test",
+    srcs: ["CounterMetric_test.cpp"],
+    shared_libs: ["libmediadrm"],
+    include_dirs: ["frameworks/av/include/media"],
+    cflags: [
+      "-Werror",
+      "-Wall",
+    ],
+}
+
+cc_test {
+    name: "DrmMetrics_test",
+    srcs: ["DrmMetrics_test.cpp"],
+    shared_libs: [
+      "android.hardware.drm@1.0",
+      "android.hardware.drm@1.1",
+      "libbinder",
+      "libhidlbase",
+      "liblog",
+      "libmediadrmmetrics_full",
+      "libmediametrics",
+      "libprotobuf-cpp-full",
+      "libutils",
+    ],
+    static_libs: ["libgmock"],
+    include_dirs: [
+      "frameworks/av/include/media",
+    ],
+    cflags: [
+        // Suppress unused parameter and no error options. These cause problems
+        // when using the map type in a proto definition.
+        "-Wno-unused-parameter",
+        "-Wno-error",
+    ]
+}
+
+cc_test {
+    name: "EventMetric_test",
+    srcs: ["EventMetric_test.cpp"],
+    shared_libs: [
+      "liblog",
+      "libmediadrm",
+      "libutils",
+    ],
+    include_dirs: ["frameworks/av/include/media"],
+    cflags: [
+      "-Werror",
+      "-Wall",
+    ],
+}

diff --git a/drm/libmediadrm/tests/CounterMetric_test.cpp b/drm/libmediadrm/tests/CounterMetric_test.cpp
new file mode 100644
index 0000000..6bca0da
--- /dev/null
+++ b/drm/libmediadrm/tests/CounterMetric_test.cpp

@@ -0,0 +1,80 @@
+/*
+ * Copyright 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <gtest/gtest.h>
+
+#include "CounterMetric.h"
+
+namespace android {
+
+/**
+ * Unit tests for the CounterMetric class.
+ */
+class CounterMetricTest : public ::testing::Test {
+};
+
+TEST_F(CounterMetricTest, IntDataTypeEmpty) {
+  CounterMetric<int> metric("MyMetricName", "MetricAttributeName");
+
+  std::map<int, int64_t> values;
+
+  metric.ExportValues(
+      [&] (int attribute_value, int64_t value) {
+          values[attribute_value] = value;
+      });
+
+  EXPECT_TRUE(values.empty());
+}
+
+TEST_F(CounterMetricTest, IntDataType) {
+  CounterMetric<int> metric("MyMetricName", "MetricAttributeName");
+
+  std::map<int, int64_t> values;
+
+  metric.Increment(7);
+  metric.Increment(8);
+  metric.Increment(8);
+
+  metric.ExportValues(
+      [&] (int attribute_value, int64_t value) {
+          values[attribute_value] = value;
+      });
+
+  ASSERT_EQ(2u, values.size());
+  EXPECT_EQ(1, values[7]);
+  EXPECT_EQ(2, values[8]);
+}
+
+TEST_F(CounterMetricTest, StringDataType) {
+  CounterMetric<std::string> metric("MyMetricName", "MetricAttributeName");
+
+  std::map<std::string, int64_t> values;
+
+  metric.Increment("a");
+  metric.Increment("b");
+  metric.Increment("b");
+
+  metric.ExportValues(
+      [&] (std::string attribute_value, int64_t value) {
+          values[attribute_value] = value;
+      });
+
+  ASSERT_EQ(2u, values.size());
+  EXPECT_EQ(1, values["a"]);
+  EXPECT_EQ(2, values["b"]);
+}
+
+}  // namespace android

diff --git a/drm/libmediadrm/tests/DrmMetrics_test.cpp b/drm/libmediadrm/tests/DrmMetrics_test.cpp
new file mode 100644
index 0000000..64aa9d0
--- /dev/null
+++ b/drm/libmediadrm/tests/DrmMetrics_test.cpp

@@ -0,0 +1,475 @@
+/*
+ * Copyright 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "DrmMetricsTest"
+#include "mediadrm/DrmMetrics.h"
+
+#include <android/hardware/drm/1.0/types.h>
+#include <android/hardware/drm/1.1/types.h>
+#include <binder/PersistableBundle.h>
+#include <google/protobuf/text_format.h>
+#include <google/protobuf/util/message_differencer.h>
+#include <gtest/gtest.h>
+#include <utils/Log.h>
+
+#include "protos/metrics.pb.h"
+
+using ::android::drm_metrics::DrmFrameworkMetrics;
+using ::android::hardware::hidl_vec;
+using ::android::hardware::drm::V1_0::EventType;
+using ::android::hardware::drm::V1_0::KeyStatusType;
+using ::android::hardware::drm::V1_0::Status;
+using ::android::hardware::drm::V1_1::DrmMetricGroup;
+using ::android::os::PersistableBundle;
+using ::google::protobuf::util::MessageDifferencer;
+using ::google::protobuf::TextFormat;
+
+namespace android {
+
+/**
+ * Unit tests for the MediaDrmMetrics class.
+ */
+class MediaDrmMetricsTest : public ::testing::Test {};
+
+/**
+ * This derived class mocks the clock for testing purposes.
+ */
+class FakeMediaDrmMetrics : public MediaDrmMetrics {
+ public:
+  FakeMediaDrmMetrics() : MediaDrmMetrics(), time_(0) {};
+
+  int64_t GetCurrentTimeMs() { return time_++; }
+  int64_t time_;
+};
+
+TEST_F(MediaDrmMetricsTest, EmptySuccess) {
+  MediaDrmMetrics metrics;
+  PersistableBundle bundle;
+
+  metrics.Export(&bundle);
+  EXPECT_TRUE(bundle.empty());
+}
+
+TEST_F(MediaDrmMetricsTest, AllValuesSuccessCounts) {
+  MediaDrmMetrics metrics;
+
+  metrics.mOpenSessionCounter.Increment(OK);
+  metrics.mCloseSessionCounter.Increment(OK);
+
+  {
+    EventTimer<status_t> get_key_request_timer(&metrics.mGetKeyRequestTimeUs);
+    EventTimer<status_t> provide_key_response_timer(
+        &metrics.mProvideKeyResponseTimeUs);
+    get_key_request_timer.SetAttribute(OK);
+    provide_key_response_timer.SetAttribute(OK);
+  }
+
+  metrics.mGetProvisionRequestCounter.Increment(OK);
+  metrics.mProvideProvisionResponseCounter.Increment(OK);
+  metrics.mGetDeviceUniqueIdCounter.Increment(OK);
+
+  metrics.mKeyStatusChangeCounter.Increment(KeyStatusType::USABLE);
+  metrics.mEventCounter.Increment(EventType::PROVISION_REQUIRED);
+
+  PersistableBundle bundle;
+
+  metrics.Export(&bundle);
+  EXPECT_EQ(11U, bundle.size());
+
+  // Verify the list of pairs of int64 metrics.
+  std::vector<std::pair<std::string, int64_t>> expected_values = {
+      { "drm.mediadrm.open_session.ok.count", 1 },
+      { "drm.mediadrm.close_session.ok.count", 1 },
+      { "drm.mediadrm.get_key_request.ok.count", 1 },
+      { "drm.mediadrm.provide_key_response.ok.count", 1 },
+      { "drm.mediadrm.get_provision_request.ok.count", 1 },
+      { "drm.mediadrm.provide_provision_response.ok.count", 1 },
+      { "drm.mediadrm.key_status_change.USABLE.count", 1 },
+      { "drm.mediadrm.event.PROVISION_REQUIRED.count", 1 },
+      { "drm.mediadrm.get_device_unique_id.ok.count", 1 }};
+  for (const auto& expected_pair : expected_values) {
+    String16 key(expected_pair.first.c_str());
+    int64_t value = -1;
+    EXPECT_TRUE(bundle.getLong(key, &value))
+        << "Unexpected error retrieviing key: " << key;
+    EXPECT_EQ(expected_pair.second, value)
+        << "Unexpected value for " << expected_pair.first << ". " << value;
+  }
+
+  // Validate timing values exist.
+  String16 get_key_request_key(
+      "drm.mediadrm.get_key_request.ok.average_time_micros");
+  String16 provide_key_response_key(
+      "drm.mediadrm.provide_key_response.ok.average_time_micros");
+  int64_t value = -1;
+  EXPECT_TRUE(bundle.getLong(get_key_request_key, &value));
+  EXPECT_GE(value, 0);
+  value = -1;
+  EXPECT_TRUE(bundle.getLong(provide_key_response_key, &value));
+  EXPECT_GE(value, 0);
+}
+
+TEST_F(MediaDrmMetricsTest, AllValuesFull) {
+  MediaDrmMetrics metrics;
+
+  metrics.mOpenSessionCounter.Increment(OK);
+  metrics.mOpenSessionCounter.Increment(UNEXPECTED_NULL);
+
+  metrics.mCloseSessionCounter.Increment(OK);
+  metrics.mCloseSessionCounter.Increment(UNEXPECTED_NULL);
+
+  for (status_t s : {OK, UNEXPECTED_NULL}) {
+    {
+      EventTimer<status_t> get_key_request_timer(&metrics.mGetKeyRequestTimeUs);
+      EventTimer<status_t> provide_key_response_timer(
+          &metrics.mProvideKeyResponseTimeUs);
+      get_key_request_timer.SetAttribute(s);
+      provide_key_response_timer.SetAttribute(s);
+    }
+  }
+
+  metrics.mGetProvisionRequestCounter.Increment(OK);
+  metrics.mGetProvisionRequestCounter.Increment(UNEXPECTED_NULL);
+  metrics.mProvideProvisionResponseCounter.Increment(OK);
+  metrics.mProvideProvisionResponseCounter.Increment(UNEXPECTED_NULL);
+  metrics.mGetDeviceUniqueIdCounter.Increment(OK);
+  metrics.mGetDeviceUniqueIdCounter.Increment(UNEXPECTED_NULL);
+
+  metrics.mKeyStatusChangeCounter.Increment(KeyStatusType::USABLE);
+  metrics.mKeyStatusChangeCounter.Increment(KeyStatusType::EXPIRED);
+  metrics.mKeyStatusChangeCounter.Increment(KeyStatusType::OUTPUTNOTALLOWED);
+  metrics.mKeyStatusChangeCounter.Increment(KeyStatusType::STATUSPENDING);
+  metrics.mKeyStatusChangeCounter.Increment(KeyStatusType::INTERNALERROR);
+  metrics.mEventCounter.Increment(EventType::PROVISION_REQUIRED);
+  metrics.mEventCounter.Increment(EventType::KEY_NEEDED);
+  metrics.mEventCounter.Increment(EventType::KEY_EXPIRED);
+  metrics.mEventCounter.Increment(EventType::VENDOR_DEFINED);
+  metrics.mEventCounter.Increment(EventType::SESSION_RECLAIMED);
+
+  android::Vector<uint8_t> sessionId1;
+  sessionId1.push_back(1);
+  sessionId1.push_back(2);
+  android::Vector<uint8_t> sessionId2;
+  sessionId2.push_back(3);
+  sessionId2.push_back(4);
+  String16 hexSessionId1("0102");
+  String16 hexSessionId2("0304");
+
+  metrics.SetSessionStart(sessionId1);
+  metrics.SetSessionStart(sessionId2);
+  metrics.SetSessionEnd(sessionId2);
+  metrics.SetSessionEnd(sessionId1);
+
+  PersistableBundle bundle;
+  metrics.Export(&bundle);
+  EXPECT_EQ(35U, bundle.size());
+
+  // Verify the list of pairs of int64 metrics.
+  std::vector<std::pair<std::string, int64_t>> expected_values = {
+      { "drm.mediadrm.open_session.ok.count", 1 },
+      { "drm.mediadrm.close_session.ok.count", 1 },
+      { "drm.mediadrm.get_key_request.ok.count", 1 },
+      { "drm.mediadrm.provide_key_response.ok.count", 1 },
+      { "drm.mediadrm.get_provision_request.ok.count", 1 },
+      { "drm.mediadrm.provide_provision_response.ok.count", 1 },
+      { "drm.mediadrm.get_device_unique_id.ok.count", 1 },
+      { "drm.mediadrm.open_session.error.count", 1 },
+      { "drm.mediadrm.close_session.error.count", 1 },
+      { "drm.mediadrm.get_key_request.error.count", 1 },
+      { "drm.mediadrm.provide_key_response.error.count", 1 },
+      { "drm.mediadrm.get_provision_request.error.count", 1 },
+      { "drm.mediadrm.provide_provision_response.error.count", 1 },
+      { "drm.mediadrm.get_device_unique_id.error.count", 1 },
+      { "drm.mediadrm.key_status_change.USABLE.count", 1 },
+      { "drm.mediadrm.key_status_change.EXPIRED.count", 1 },
+      { "drm.mediadrm.key_status_change.OUTPUT_NOT_ALLOWED.count", 1 },
+      { "drm.mediadrm.key_status_change.STATUS_PENDING.count", 1 },
+      { "drm.mediadrm.key_status_change.INTERNAL_ERROR.count", 1 },
+      { "drm.mediadrm.event.PROVISION_REQUIRED.count", 1 },
+      { "drm.mediadrm.event.KEY_NEEDED.count", 1 },
+      { "drm.mediadrm.event.KEY_EXPIRED.count", 1 },
+      { "drm.mediadrm.event.VENDOR_DEFINED.count", 1 },
+      { "drm.mediadrm.event.SESSION_RECLAIMED.count", 1 }};
+  for (const auto& expected_pair : expected_values) {
+    String16 key(expected_pair.first.c_str());
+    int64_t value = -1;
+    EXPECT_TRUE(bundle.getLong(key, &value))
+        << "Unexpected error retrieviing key: " << key;
+    EXPECT_EQ(expected_pair.second, value)
+        << "Unexpected value for " << expected_pair.first << ". " << value;
+  }
+
+  // Verify the error lists
+  std::vector<std::pair<std::string, std::vector<int64_t>>> expected_vector_values = {
+      { "drm.mediadrm.close_session.error.list", { UNEXPECTED_NULL } },
+      { "drm.mediadrm.get_device_unique_id.error.list", { UNEXPECTED_NULL } },
+      { "drm.mediadrm.get_key_request.error.list", { UNEXPECTED_NULL } },
+      { "drm.mediadrm.get_provision_request.error.list", { UNEXPECTED_NULL } },
+      { "drm.mediadrm.open_session.error.list", { UNEXPECTED_NULL } },
+      { "drm.mediadrm.provide_key_response.error.list", { UNEXPECTED_NULL } },
+      { "drm.mediadrm.provide_provision_response.error.list", { UNEXPECTED_NULL } }};
+  for (const auto& expected_pair : expected_vector_values) {
+    String16 key(expected_pair.first.c_str());
+    std::vector<int64_t> values;
+    EXPECT_TRUE(bundle.getLongVector(key, &values))
+        << "Unexpected error retrieviing key: " << key;
+    for (auto expected : expected_pair.second) {
+      EXPECT_TRUE(std::find(values.begin(), values.end(), expected) != values.end())
+          << "Could not find " << expected << " for key " << expected_pair.first;
+    }
+  }
+
+  // Verify the lifespans
+  PersistableBundle start_times;
+  PersistableBundle end_times;
+  String16 start_time_key("drm.mediadrm.session_start_times_ms");
+  String16 end_time_key("drm.mediadrm.session_end_times_ms");
+  ASSERT_TRUE(bundle.getPersistableBundle(start_time_key, &start_times));
+  ASSERT_TRUE(bundle.getPersistableBundle(end_time_key, &end_times));
+  EXPECT_EQ(2U, start_times.size());
+  EXPECT_EQ(2U, end_times.size());
+  int64_t start_time, end_time;
+  for (const auto& sid : { hexSessionId1, hexSessionId2 }) {
+    start_time = -1;
+    end_time = -1;
+    EXPECT_TRUE(start_times.getLong(sid, &start_time));
+    EXPECT_TRUE(end_times.getLong(sid, &end_time));
+    EXPECT_GT(start_time, 0);
+    EXPECT_GE(end_time, start_time);
+  }
+
+  // Validate timing values exist.
+  String16 get_key_request_key(
+      "drm.mediadrm.get_key_request.ok.average_time_micros");
+  String16 provide_key_response_key(
+      "drm.mediadrm.provide_key_response.ok.average_time_micros");
+  int64_t value = -1;
+  EXPECT_TRUE(bundle.getLong(get_key_request_key, &value));
+  EXPECT_GE(value, 0);
+  value = -1;
+  EXPECT_TRUE(bundle.getLong(provide_key_response_key, &value));
+  EXPECT_GE(value, 0);
+}
+
+
+TEST_F(MediaDrmMetricsTest, CounterValuesProtoSerialization) {
+  MediaDrmMetrics metrics;
+
+  metrics.mOpenSessionCounter.Increment(OK);
+  metrics.mOpenSessionCounter.Increment(UNEXPECTED_NULL);
+  metrics.mCloseSessionCounter.Increment(OK);
+  metrics.mCloseSessionCounter.Increment(UNEXPECTED_NULL);
+
+  metrics.mGetProvisionRequestCounter.Increment(OK);
+  metrics.mGetProvisionRequestCounter.Increment(UNEXPECTED_NULL);
+  metrics.mProvideProvisionResponseCounter.Increment(OK);
+  metrics.mProvideProvisionResponseCounter.Increment(UNEXPECTED_NULL);
+  metrics.mGetDeviceUniqueIdCounter.Increment(OK);
+  metrics.mGetDeviceUniqueIdCounter.Increment(UNEXPECTED_NULL);
+
+  metrics.mKeyStatusChangeCounter.Increment(KeyStatusType::USABLE);
+  metrics.mKeyStatusChangeCounter.Increment(KeyStatusType::EXPIRED);
+  metrics.mKeyStatusChangeCounter.Increment(KeyStatusType::OUTPUTNOTALLOWED);
+  metrics.mKeyStatusChangeCounter.Increment(KeyStatusType::STATUSPENDING);
+  metrics.mKeyStatusChangeCounter.Increment(KeyStatusType::INTERNALERROR);
+  metrics.mEventCounter.Increment(EventType::PROVISION_REQUIRED);
+  metrics.mEventCounter.Increment(EventType::KEY_NEEDED);
+  metrics.mEventCounter.Increment(EventType::KEY_EXPIRED);
+  metrics.mEventCounter.Increment(EventType::VENDOR_DEFINED);
+  metrics.mEventCounter.Increment(EventType::SESSION_RECLAIMED);
+
+  std::string serializedMetrics;
+  ASSERT_EQ(OK, metrics.GetSerializedMetrics(&serializedMetrics));
+
+  DrmFrameworkMetrics metricsProto;
+  ASSERT_TRUE(metricsProto.ParseFromString(serializedMetrics));
+
+  std::string expectedMetrics =
+      "open_session_counter { count: 1 attributes { error_code: -0x7FFFFFF8 } } "
+      "open_session_counter { count: 1 attributes { error_code: 0 } } "
+      "close_session_counter { count: 1 attributes { error_code: -0x7FFFFFF8 } } "
+      "close_session_counter { count: 1 attributes { error_code: 0 } } "
+      "get_provisioning_request_counter { count: 1 attributes { error_code: -0x7FFFFFF8 } } "
+      "get_provisioning_request_counter { count: 1 attributes { error_code: 0 } } "
+      "provide_provisioning_response_counter { count: 1 attributes { error_code: -0x7ffffff8 } } "
+      "provide_provisioning_response_counter { count: 1 attributes { error_code: 0 } } "
+      "get_device_unique_id_counter { count: 1 attributes { error_code: -0x7ffffff8 } } "
+      "get_device_unique_id_counter { count: 1 attributes { error_code: 0 } } "
+      "key_status_change_counter { count: 1 attributes { key_status_type: 0 } } "
+      "key_status_change_counter { count: 1 attributes { key_status_type: 1 } } "
+      "key_status_change_counter { count: 1 attributes { key_status_type: 2 } } "
+      "key_status_change_counter { count: 1 attributes { key_status_type: 3 } } "
+      "key_status_change_counter { count: 1 attributes { key_status_type: 4 } } "
+      "event_callback_counter { count: 1 attributes { event_type: 0 } } "
+      "event_callback_counter { count: 1 attributes { event_type: 1 } } "
+      "event_callback_counter { count: 1 attributes { event_type: 2 } } "
+      "event_callback_counter { count: 1 attributes { event_type: 3 } } "
+      "event_callback_counter { count: 1 attributes { event_type: 4 } } ";
+
+  DrmFrameworkMetrics expectedMetricsProto;
+  ASSERT_TRUE(TextFormat::MergeFromString(expectedMetrics, &expectedMetricsProto));
+
+  std::string diffString;
+  MessageDifferencer differ;
+  differ.ReportDifferencesToString(&diffString);
+  ASSERT_TRUE(differ.Compare(expectedMetricsProto, metricsProto))
+      << diffString;
+}
+
+TEST_F(MediaDrmMetricsTest, TimeMetricsProtoSerialization) {
+  MediaDrmMetrics metrics;
+
+  for (status_t s : {OK, UNEXPECTED_NULL}) {
+    double time = 0;
+    for (int i = 0; i < 5; i++) {
+      time += 1.0;
+      metrics.mGetKeyRequestTimeUs.Record(time, s);
+      metrics.mProvideKeyResponseTimeUs.Record(time, s);
+    }
+  }
+
+  std::string serializedMetrics;
+  ASSERT_EQ(OK, metrics.GetSerializedMetrics(&serializedMetrics));
+
+  DrmFrameworkMetrics metricsProto;
+  ASSERT_TRUE(metricsProto.ParseFromString(serializedMetrics));
+
+  std::string expectedMetrics =
+      "get_key_request_time_us { "
+      "  min: 1 max: 5 mean: 3.5 variance: 1 operation_count: 5 "
+      "  attributes { error_code: -0x7FFFFFF8 } "
+      "} "
+      "get_key_request_time_us { "
+      "  min: 1 max: 5 mean: 3.5 variance: 1 operation_count: 5 "
+      "  attributes { error_code: 0 } "
+      "} "
+      "provide_key_response_time_us { "
+      "  min: 1 max: 5 mean: 3.5 variance: 1 operation_count: 5 "
+      "  attributes { error_code: -0x7FFFFFF8 } "
+      "} "
+      "provide_key_response_time_us { "
+      "  min: 1 max: 5 mean: 3.5 variance: 1 operation_count: 5 "
+      "  attributes { error_code: 0 } "
+      "} ";
+
+  DrmFrameworkMetrics expectedMetricsProto;
+  ASSERT_TRUE(TextFormat::MergeFromString(expectedMetrics, &expectedMetricsProto));
+
+  std::string diffString;
+  MessageDifferencer differ;
+  differ.ReportDifferencesToString(&diffString);
+  ASSERT_TRUE(differ.Compare(expectedMetricsProto, metricsProto))
+      << diffString;
+}
+
+TEST_F(MediaDrmMetricsTest, SessionLifetimeProtoSerialization) {
+  // Use the fake so the clock is predictable;
+  FakeMediaDrmMetrics metrics;
+
+  android::Vector<uint8_t> sessionId1;
+  sessionId1.push_back(1);
+  sessionId1.push_back(2);
+  android::Vector<uint8_t> sessionId2;
+  sessionId2.push_back(3);
+  sessionId2.push_back(4);
+
+  metrics.SetSessionStart(sessionId1);
+  metrics.SetSessionStart(sessionId2);
+  metrics.SetSessionEnd(sessionId2);
+  metrics.SetSessionEnd(sessionId1);
+
+  std::string serializedMetrics;
+  ASSERT_EQ(OK, metrics.GetSerializedMetrics(&serializedMetrics));
+
+  DrmFrameworkMetrics metricsProto;
+  ASSERT_TRUE(metricsProto.ParseFromString(serializedMetrics));
+
+  std::string expectedMetrics =
+      "session_lifetimes: { "
+      "  key: '0102' "
+      "  value { start_time_ms: 0 end_time_ms: 3 } "
+      "} "
+      "session_lifetimes: { "
+      "  key: '0304' "
+      "  value { start_time_ms: 1 end_time_ms: 2 } "
+      "} ";
+
+  DrmFrameworkMetrics expectedMetricsProto;
+  ASSERT_TRUE(TextFormat::MergeFromString(expectedMetrics, &expectedMetricsProto));
+
+  std::string diffString;
+  MessageDifferencer differ;
+  differ.ReportDifferencesToString(&diffString);
+  ASSERT_TRUE(differ.Compare(expectedMetricsProto, metricsProto))
+      << diffString;
+}
+
+TEST_F(MediaDrmMetricsTest, HidlToBundleMetricsEmpty) {
+  hidl_vec<DrmMetricGroup> hidlMetricGroups;
+  PersistableBundle bundleMetricGroups;
+
+  ASSERT_EQ(OK, MediaDrmMetrics::HidlMetricsToBundle(hidlMetricGroups, &bundleMetricGroups));
+  ASSERT_EQ(0U, bundleMetricGroups.size());
+}
+
+TEST_F(MediaDrmMetricsTest, HidlToBundleMetricsMultiple) {
+  DrmMetricGroup hidlMetricGroup =
+      { { {
+              "open_session_ok",
+              { { "status", DrmMetricGroup::ValueType::INT64_TYPE,
+                  (int64_t) Status::OK, 0.0, "" } },
+              { { "count", DrmMetricGroup::ValueType::INT64_TYPE, 3, 0.0, "" } }
+          },
+          {
+              "close_session_not_opened",
+              { { "status", DrmMetricGroup::ValueType::INT64_TYPE,
+                  (int64_t) Status::ERROR_DRM_SESSION_NOT_OPENED, 0.0, "" } },
+              { { "count", DrmMetricGroup::ValueType::INT64_TYPE, 7, 0.0, "" } }
+          } } };
+
+  PersistableBundle bundleMetricGroups;
+  ASSERT_EQ(OK, MediaDrmMetrics::HidlMetricsToBundle(hidl_vec<DrmMetricGroup>({hidlMetricGroup}),
+                                                     &bundleMetricGroups));
+  ASSERT_EQ(1U, bundleMetricGroups.size());
+  PersistableBundle bundleMetricGroup;
+  ASSERT_TRUE(bundleMetricGroups.getPersistableBundle(String16("[0]"), &bundleMetricGroup));
+  ASSERT_EQ(2U, bundleMetricGroup.size());
+
+  // Verify each metric.
+  PersistableBundle metric;
+  ASSERT_TRUE(bundleMetricGroup.getPersistableBundle(String16("open_session_ok"), &metric));
+  PersistableBundle metricInstance;
+  ASSERT_TRUE(metric.getPersistableBundle(String16("[0]"), &metricInstance));
+  int64_t value = 0;
+  ASSERT_TRUE(metricInstance.getLong(String16("count"), &value));
+  ASSERT_EQ(3, value);
+  PersistableBundle attributeBundle;
+  ASSERT_TRUE(metricInstance.getPersistableBundle(String16("attributes"), &attributeBundle));
+  ASSERT_TRUE(attributeBundle.getLong(String16("status"), &value));
+  ASSERT_EQ((int64_t) Status::OK, value);
+
+  ASSERT_TRUE(bundleMetricGroup.getPersistableBundle(String16("close_session_not_opened"),
+                                                     &metric));
+  ASSERT_TRUE(metric.getPersistableBundle(String16("[0]"), &metricInstance));
+  ASSERT_TRUE(metricInstance.getLong(String16("count"), &value));
+  ASSERT_EQ(7, value);
+  ASSERT_TRUE(metricInstance.getPersistableBundle(String16("attributes"), &attributeBundle));
+  value = 0;
+  ASSERT_TRUE(attributeBundle.getLong(String16("status"), &value));
+  ASSERT_EQ((int64_t) Status::ERROR_DRM_SESSION_NOT_OPENED, value);
+}
+
+}  // namespace android

diff --git a/drm/libmediadrm/tests/EventMetric_test.cpp b/drm/libmediadrm/tests/EventMetric_test.cpp
new file mode 100644
index 0000000..eb6c4f6
--- /dev/null
+++ b/drm/libmediadrm/tests/EventMetric_test.cpp

@@ -0,0 +1,142 @@
+/*
+ * Copyright 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <gtest/gtest.h>
+
+#include "EventMetric.h"
+
+namespace android {
+
+/**
+ * Unit tests for the EventMetric class.
+ */
+
+TEST(EventMetricTest, IntDataTypeEmpty) {
+  EventMetric<int> metric("MyMetricName", "MetricAttributeName");
+
+  std::map<int, EventStatistics> values;
+
+  metric.ExportValues(
+      [&] (int attribute_value, const EventStatistics& value) {
+          values[attribute_value] = value;
+      });
+
+  EXPECT_TRUE(values.empty());
+}
+
+TEST(EventMetricTest, IntDataType) {
+  EventMetric<int> metric("MyMetricName", "MetricAttributeName");
+
+  std::map<int, EventStatistics> values;
+
+  metric.Record(4, 7);
+  metric.Record(5, 8);
+  metric.Record(5, 8);
+  metric.Record(5, 8);
+  metric.Record(6, 8);
+  metric.Record(6, 8);
+  metric.Record(6, 8);
+
+  metric.ExportValues(
+      [&] (int attribute_value, const EventStatistics& value) {
+          values[attribute_value] = value;
+      });
+
+  ASSERT_EQ(2u, values.size());
+  EXPECT_EQ(4, values[7].min);
+  EXPECT_EQ(4, values[7].max);
+  EXPECT_EQ(4, values[7].mean);
+  EXPECT_EQ(1, values[7].count);
+
+  EXPECT_EQ(5, values[8].min);
+  EXPECT_EQ(6, values[8].max);
+  // This is an approximate value because of the technique we're using.
+  EXPECT_NEAR(5.5, values[8].mean, 0.2);
+  EXPECT_EQ(6, values[8].count);
+}
+
+TEST(EventMetricTest, StringDataType) {
+  EventMetric<std::string> metric("MyMetricName", "MetricAttributeName");
+
+  std::map<std::string, EventStatistics> values;
+
+  metric.Record(1, "a");
+  metric.Record(2, "b");
+  metric.Record(2, "b");
+  metric.Record(3, "b");
+  metric.Record(3, "b");
+
+  metric.ExportValues(
+      [&] (std::string attribute_value, const EventStatistics& value) {
+          values[attribute_value] = value;
+      });
+
+  ASSERT_EQ(2u, values.size());
+  EXPECT_EQ(1, values["a"].min);
+  EXPECT_EQ(1, values["a"].max);
+  EXPECT_EQ(1, values["a"].mean);
+  EXPECT_EQ(1, values["a"].count);
+
+  EXPECT_EQ(2, values["b"].min);
+  EXPECT_EQ(3, values["b"].max);
+  EXPECT_NEAR(2.5, values["b"].mean, 0.2);
+  EXPECT_EQ(4, values["b"].count);
+}
+
+// Helper class that allows us to mock the clock.
+template<typename AttributeType>
+class MockEventTimer : public EventTimer<AttributeType> {
+ public:
+  explicit MockEventTimer(nsecs_t time_delta_ns,
+                          EventMetric<AttributeType>* metric)
+      : EventTimer<AttributeType>(metric) {
+    // Pretend the event started earlier.
+    this->start_time_ = systemTime() - time_delta_ns;
+  }
+};
+
+TEST(EventTimerTest, IntDataType) {
+  EventMetric<int> metric("MyMetricName", "MetricAttributeName");
+
+  for (int i = 0; i < 5; i++) {
+    {
+      // Add a mock time delta.
+      MockEventTimer<int> metric_timer(i * 1000000, &metric);
+      metric_timer.SetAttribute(i % 2);
+    }
+  }
+
+  std::map<int, EventStatistics> values;
+  metric.ExportValues(
+      [&] (int attribute_value, const EventStatistics& value) {
+          values[attribute_value] = value;
+      });
+
+  ASSERT_EQ(2u, values.size());
+  EXPECT_LT(values[0].min, values[0].max);
+  EXPECT_GE(4000, values[0].max);
+  EXPECT_GT(values[0].mean, values[0].min);
+  EXPECT_LE(values[0].mean, values[0].max);
+  EXPECT_EQ(3, values[0].count);
+
+  EXPECT_LT(values[1].min, values[1].max);
+  EXPECT_GE(3000, values[1].max);
+  EXPECT_GT(values[1].mean, values[1].min);
+  EXPECT_LE(values[1].mean, values[1].max);
+  EXPECT_EQ(2, values[1].count);
+}
+
+}  // namespace android

diff --git a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp
index 50acc1d..1558e8b 100644
--- a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp
+++ b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp

@@ -55,7 +55,7 @@
 
 status_t ClearKeyCasFactory::createPlugin(
         int32_t CA_system_id,
-        uint64_t appData,
+        void *appData,
         CasPluginCallback callback,
         CasPlugin **plugin) {
     if (!isSystemIdSupported(CA_system_id)) {
@@ -83,7 +83,7 @@
 
 ///////////////////////////////////////////////////////////////////////////////
 ClearKeyCasPlugin::ClearKeyCasPlugin(
-        uint64_t appData, CasPluginCallback callback)
+        void *appData, CasPluginCallback callback)
     : mCallback(callback), mAppData(appData) {
     ALOGV("CTOR");
 }
@@ -347,6 +347,9 @@
         return ERROR_CAS_CANNOT_HANDLE;
     }
 
+    scramblingControl = (DescramblerPlugin::ScramblingControl)
+        (scramblingControl & DescramblerPlugin::kScrambling_Mask_Key);
+
     AES_KEY contentKey;
 
     if (scramblingControl != DescramblerPlugin::kScrambling_Unscrambled) {

diff --git a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h
index 8a9ea83..389e172 100644
--- a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h
+++ b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h

@@ -44,7 +44,7 @@
             std::vector<CasPluginDescriptor> *descriptors) const override;
     virtual status_t createPlugin(
             int32_t CA_system_id,
-            uint64_t appData,
+            void *appData,
             CasPluginCallback callback,
             CasPlugin **plugin) override;
 };
@@ -62,7 +62,7 @@
 
 class ClearKeyCasPlugin : public CasPlugin {
 public:
-    ClearKeyCasPlugin(uint64_t appData, CasPluginCallback callback);
+    ClearKeyCasPlugin(void *appData, CasPluginCallback callback);
     virtual ~ClearKeyCasPlugin();
 
     virtual status_t setPrivateData(
@@ -94,7 +94,7 @@
     Mutex mKeyFetcherLock;
     std::unique_ptr<KeyFetcher> mKeyFetcher;
     CasPluginCallback mCallback;
-    uint64_t mAppData;
+    void* mAppData;
 };
 
 class ClearKeyDescramblerPlugin : public DescramblerPlugin {

diff --git a/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp b/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp
index cb69f91..eaa3390 100644
--- a/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp
+++ b/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp

@@ -89,7 +89,7 @@
     // asset_id change. If it sends an EcmContainer with 2 Ecms with different
     // asset_ids (old and new) then it might be best to prefetch the Emm.
     if ((asset_.id() != 0) && (*asset_id != asset_.id())) {
-        ALOGW("Asset_id change from %" PRIu64 " to %" PRIu64, asset_.id(), *asset_id);
+        ALOGW("Asset_id change from %llu to %" PRIu64, asset_.id(), *asset_id);
         asset_.Clear();
     }
 

diff --git a/drm/mediacas/plugins/clearkey/ClearKeySessionLibrary.cpp b/drm/mediacas/plugins/clearkey/ClearKeySessionLibrary.cpp
index 9fd2d4d..3bb1176 100644
--- a/drm/mediacas/plugins/clearkey/ClearKeySessionLibrary.cpp
+++ b/drm/mediacas/plugins/clearkey/ClearKeySessionLibrary.cpp

@@ -95,7 +95,7 @@
 void ClearKeySessionLibrary::destroyPlugin(CasPlugin *plugin) {
     Mutex::Autolock lock(mSessionsLock);
 
-    for (ssize_t index = mIDToSessionMap.size() - 1; index >= 0; index--) {
+    for (ssize_t index = (ssize_t)mIDToSessionMap.size() - 1; index >= 0; index--) {
         std::shared_ptr<ClearKeyCasSession> session = mIDToSessionMap.valueAt(index);
         if (session->getPlugin() == plugin) {
             mIDToSessionMap.removeItemsAt(index);

diff --git a/drm/mediacas/plugins/clearkey/JsonAssetLoader.cpp b/drm/mediacas/plugins/clearkey/JsonAssetLoader.cpp
index 6e1004c..ee8dba3 100644
--- a/drm/mediacas/plugins/clearkey/JsonAssetLoader.cpp
+++ b/drm/mediacas/plugins/clearkey/JsonAssetLoader.cpp

@@ -36,8 +36,6 @@
 const String8 kCasTypeTag("cas_type");
 const String8 kBase64Padding("=");
 
-const uint32_t kKeyLength = 16;
-
 JsonAssetLoader::JsonAssetLoader() {
 }
 

diff --git a/drm/mediacas/plugins/clearkey/ecm_generator.cpp b/drm/mediacas/plugins/clearkey/ecm_generator.cpp
index 7d29659..218ce35 100644
--- a/drm/mediacas/plugins/clearkey/ecm_generator.cpp
+++ b/drm/mediacas/plugins/clearkey/ecm_generator.cpp

@@ -38,8 +38,6 @@
 const uint16_t kTotalEcmSize =
         kEcmClearFieldsSize + kContentKeyByteSize; // clear fields + clear key
 
-const uint32_t kKeyLength = 16;
-
 #define UNALIGNED_LOAD32(_p) (*reinterpret_cast<const uint32_t *>(_p))
 
 static uint32_t Load32(const void *p) {

diff --git a/drm/mediacas/plugins/clearkey/tests/ClearKeyFetcherTest.cpp b/drm/mediacas/plugins/clearkey/tests/ClearKeyFetcherTest.cpp
index ace086a..d12cfeb 100644
--- a/drm/mediacas/plugins/clearkey/tests/ClearKeyFetcherTest.cpp
+++ b/drm/mediacas/plugins/clearkey/tests/ClearKeyFetcherTest.cpp

@@ -93,7 +93,7 @@
     uint64_t asset_id;
     std::vector<KeyFetcher::KeyInfo> keys;
     EXPECT_EQ(OK, fetcher.ObtainKey(ecm_, &asset_id, &keys));
-    EXPECT_EQ(2, keys.size());
+    EXPECT_EQ(2U, keys.size());
     EXPECT_EQ(0, keys[0].key_id);
     EXPECT_EQ(content_key_[0]->size(), keys[0].key_bytes->size());
     EXPECT_EQ(0, memcmp(content_key_[0]->data(),

diff --git a/drm/mediacas/plugins/mock/MockCasPlugin.cpp b/drm/mediacas/plugins/mock/MockCasPlugin.cpp
index 18cd9a4..8404a83 100644
--- a/drm/mediacas/plugins/mock/MockCasPlugin.cpp
+++ b/drm/mediacas/plugins/mock/MockCasPlugin.cpp

@@ -49,8 +49,8 @@
 
 status_t MockCasFactory::createPlugin(
         int32_t CA_system_id,
-        uint64_t appData,
-        CasPluginCallback callback,
+        void* /*appData*/,
+        CasPluginCallback /*callback*/,
         CasPlugin **plugin) {
     if (!isSystemIdSupported(CA_system_id)) {
         return BAD_VALUE;
@@ -98,7 +98,7 @@
     MockSessionLibrary::get()->destroyPlugin(this);
 }
 
-status_t MockCasPlugin::setPrivateData(const CasData &data) {
+status_t MockCasPlugin::setPrivateData(const CasData& /*data*/) {
     ALOGV("setPrivateData");
     return OK;
 }
@@ -123,7 +123,7 @@
 }
 
 status_t MockCasPlugin::setSessionPrivateData(
-        const CasSessionId &sessionId, const CasData &data) {
+        const CasSessionId &sessionId, const CasData& /*data*/) {
     ALOGV("setSessionPrivateData: sessionId=%s",
             arrayToString(sessionId).string());
     Mutex::Autolock lock(mLock);
@@ -146,7 +146,7 @@
     if (session == NULL) {
         return BAD_VALUE;
     }
-    ALOGV("ECM: size=%d", ecm.size());
+    ALOGV("ECM: size=%zu", ecm.size());
     ALOGV("ECM: data=%s", arrayToString(ecm).string());
 
     return OK;
@@ -156,14 +156,14 @@
     ALOGV("processEmm");
     Mutex::Autolock lock(mLock);
 
-    ALOGV("EMM: size=%d", emm.size());
+    ALOGV("EMM: size=%zu", emm.size());
     ALOGV("EMM: data=%s", arrayToString(emm).string());
 
     return OK;
 }
 
 status_t MockCasPlugin::sendEvent(
-        int32_t event, int arg, const CasData &eventData) {
+        int32_t event, int /*arg*/, const CasData& /*eventData*/) {
     ALOGV("sendEvent: event=%d", event);
     Mutex::Autolock lock(mLock);
 
@@ -178,7 +178,7 @@
 }
 
 status_t MockCasPlugin::refreshEntitlements(
-        int32_t refreshType, const CasData &refreshData) {
+        int32_t /*refreshType*/, const CasData &refreshData) {
     ALOGV("refreshEntitlements: refreshData=%s", arrayToString(refreshData).string());
     Mutex::Autolock lock(mLock);
 
@@ -216,7 +216,7 @@
         int32_t srcOffset,
         void *dstPtr,
         int32_t dstOffset,
-        AString *errorDetailMsg) {
+        AString* /*errorDetailMsg*/) {
     ALOGV("MockDescramblerPlugin::descramble(secure=%d, sctrl=%d,"
           "subSamples=%s, srcPtr=%p, dstPtr=%p, srcOffset=%d, dstOffset=%d)",
           (int)secure, (int)scramblingControl,

diff --git a/drm/mediacas/plugins/mock/MockCasPlugin.h b/drm/mediacas/plugins/mock/MockCasPlugin.h
index 9632492..8106990 100644
--- a/drm/mediacas/plugins/mock/MockCasPlugin.h
+++ b/drm/mediacas/plugins/mock/MockCasPlugin.h

@@ -39,7 +39,7 @@
             std::vector<CasPluginDescriptor> *descriptors) const override;
     virtual status_t createPlugin(
             int32_t CA_system_id,
-            uint64_t appData,
+            void *appData,
             CasPluginCallback callback,
             CasPlugin **plugin) override;
 };

diff --git a/drm/mediadrm/plugins/clearkey/Android.bp b/drm/mediadrm/plugins/clearkey/Android.bp
deleted file mode 100644
index 2973fcf..0000000
--- a/drm/mediadrm/plugins/clearkey/Android.bp
+++ /dev/null

@@ -1,59 +0,0 @@
-//
-// Copyright (C) 2014 The Android Open Source Project
-//
-// Licensed under the Apache License, Version 2.0 (the "License");
-// you may not use this file except in compliance with the License.
-// You may obtain a copy of the License at
-//
-//      http://www.apache.org/licenses/LICENSE-2.0
-//
-// Unless required by applicable law or agreed to in writing, software
-// distributed under the License is distributed on an "AS IS" BASIS,
-// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-// See the License for the specific language governing permissions and
-// limitations under the License.
-//
-
-cc_library_shared {
-    name: "libdrmclearkeyplugin",
-
-    srcs: [
-        "AesCtrDecryptor.cpp",
-        "ClearKeyUUID.cpp",
-        "CreatePluginFactories.cpp",
-        "CryptoFactory.cpp",
-        "CryptoPlugin.cpp",
-        "DrmFactory.cpp",
-        "DrmPlugin.cpp",
-        "InitDataParser.cpp",
-        "JsonWebKey.cpp",
-        "Session.cpp",
-        "SessionLibrary.cpp",
-        "Utils.cpp",
-    ],
-
-    vendor: true,
-    relative_install_path: "mediadrm",
-
-    shared_libs: [
-        "libcrypto",
-        "liblog",
-        "libstagefright_foundation",
-        "libutils",
-    ],
-
-    static_libs: ["libjsmn"],
-
-    include_dirs: [
-        "frameworks/native/include",
-        "frameworks/av/include",
-    ],
-
-    export_include_dirs: ["."],
-    export_static_lib_headers: ["libjsmn"],
-}
-
-//########################################################################
-// Build unit tests
-
-subdirs = ["tests"]

diff --git a/drm/mediadrm/plugins/clearkey/CryptoFactory.cpp b/drm/mediadrm/plugins/clearkey/CryptoFactory.cpp
deleted file mode 100644
index eeb64c3..0000000
--- a/drm/mediadrm/plugins/clearkey/CryptoFactory.cpp
+++ /dev/null

@@ -1,60 +0,0 @@
-/*
- * Copyright (C) 2014 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "ClearKeyCryptoPlugin"
-#include <utils/Log.h>
-
-#include <utils/Errors.h>
-#include <utils/StrongPointer.h>
-
-#include "CryptoFactory.h"
-
-#include "ClearKeyUUID.h"
-#include "CryptoPlugin.h"
-#include "Session.h"
-#include "SessionLibrary.h"
-
-namespace clearkeydrm {
-
-bool CryptoFactory::isCryptoSchemeSupported(const uint8_t uuid[16]) const {
-    return isClearKeyUUID(uuid);
-}
-
-android::status_t CryptoFactory::createPlugin(
-        const uint8_t uuid[16],
-        const void* data, size_t size,
-        android::CryptoPlugin** plugin) {
-    if (!isCryptoSchemeSupported(uuid)) {
-        *plugin = NULL;
-        return android::BAD_VALUE;
-    }
-
-    android::Vector<uint8_t> sessionId;
-    sessionId.appendArray(reinterpret_cast<const uint8_t*>(data), size);
-
-    CryptoPlugin *clearKeyPlugin = new CryptoPlugin(sessionId);
-    android::status_t result = clearKeyPlugin->getInitStatus();
-    if (result == android::OK) {
-        *plugin = clearKeyPlugin;
-    } else {
-        delete clearKeyPlugin;
-        *plugin = NULL;
-    }
-    return result;
-}
-
-} // namespace clearkeydrm

diff --git a/drm/mediadrm/plugins/clearkey/DrmFactory.cpp b/drm/mediadrm/plugins/clearkey/DrmFactory.cpp
deleted file mode 100644
index c83321b..0000000
--- a/drm/mediadrm/plugins/clearkey/DrmFactory.cpp
+++ /dev/null

@@ -1,58 +0,0 @@
-/*
- * Copyright (C) 2014 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "ClearKeyCryptoPlugin"
-#include <utils/Log.h>
-
-#include <utils/Errors.h>
-
-#include "DrmFactory.h"
-
-#include "DrmPlugin.h"
-#include "ClearKeyUUID.h"
-#include "MimeType.h"
-#include "SessionLibrary.h"
-
-namespace clearkeydrm {
-
-bool DrmFactory::isCryptoSchemeSupported(const uint8_t uuid[16]) {
-    return isClearKeyUUID(uuid);
-}
-
-bool DrmFactory::isContentTypeSupported(const android::String8 &type) {
-    // This should match the types handed by InitDataParser.
-    return type == kIsoBmffVideoMimeType ||
-        type == kIsoBmffAudioMimeType ||
-        type == kCencInitDataFormat ||
-        type == kWebmVideoMimeType ||
-        type == kWebmAudioMimeType ||
-        type == kWebmInitDataFormat;
-}
-
-android::status_t DrmFactory::createDrmPlugin(
-        const uint8_t uuid[16],
-        android::DrmPlugin** plugin) {
-    if (!isCryptoSchemeSupported(uuid)) {
-        *plugin = NULL;
-        return android::BAD_VALUE;
-    }
-
-    *plugin = new DrmPlugin(SessionLibrary::get());
-    return android::OK;
-}
-
-} // namespace clearkeydrm

diff --git a/drm/mediadrm/plugins/clearkey/DrmPlugin.cpp b/drm/mediadrm/plugins/clearkey/DrmPlugin.cpp
deleted file mode 100644
index ec07d87..0000000
--- a/drm/mediadrm/plugins/clearkey/DrmPlugin.cpp
+++ /dev/null

@@ -1,158 +0,0 @@
-/*
- * Copyright (C) 2014 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "ClearKeyCryptoPlugin"
-#include <utils/Log.h>
-
-#include <media/stagefright/MediaErrors.h>
-#include <utils/StrongPointer.h>
-
-#include "DrmPlugin.h"
-
-#include "Session.h"
-
-namespace {
-const android::String8 kStreaming("Streaming");
-const android::String8 kOffline("Offline");
-const android::String8 kTrue("True");
-
-const android::String8 kQueryKeyLicenseType("LicenseType");
-    // Value: "Streaming" or "Offline"
-const android::String8 kQueryKeyPlayAllowed("PlayAllowed");
-    // Value: "True" or "False"
-const android::String8 kQueryKeyRenewAllowed("RenewAllowed");
-    // Value: "True" or "False"
-};
-
-namespace clearkeydrm {
-
-using android::sp;
-
-DrmPlugin::DrmPlugin(SessionLibrary* sessionLibrary)
-        : mSessionLibrary(sessionLibrary) {
-    mPlayPolicy.clear();
-}
-
-status_t DrmPlugin::openSession(Vector<uint8_t>& sessionId) {
-    sp<Session> session = mSessionLibrary->createSession();
-    sessionId = session->sessionId();
-    return android::OK;
-}
-
-status_t DrmPlugin::closeSession(const Vector<uint8_t>& sessionId) {
-    sp<Session> session = mSessionLibrary->findSession(sessionId);
-    if (sessionId.size() == 0) {
-        return android::BAD_VALUE;
-    }
-    if (session.get()) {
-        mSessionLibrary->destroySession(session);
-        return android::OK;
-    }
-    return android::ERROR_DRM_SESSION_NOT_OPENED;
-}
-
-status_t DrmPlugin::getKeyRequest(
-        const Vector<uint8_t>& scope,
-        const Vector<uint8_t>& initData,
-        const String8& mimeType,
-        KeyType keyType,
-        const KeyedVector<String8, String8>& optionalParameters,
-        Vector<uint8_t>& request,
-        String8& defaultUrl,
-        DrmPlugin::KeyRequestType *keyRequestType) {
-    UNUSED(optionalParameters);
-    if (scope.size() == 0) {
-        return android::BAD_VALUE;
-    }
-
-    if (keyType != kKeyType_Streaming) {
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    *keyRequestType = DrmPlugin::kKeyRequestType_Initial;
-    defaultUrl.clear();
-    sp<Session> session = mSessionLibrary->findSession(scope);
-    if (!session.get()) {
-        return android::ERROR_DRM_SESSION_NOT_OPENED;
-    }
-
-    return session->getKeyRequest(initData, mimeType, &request);
-}
-
-void DrmPlugin::setPlayPolicy() {
-    mPlayPolicy.clear();
-    mPlayPolicy.add(kQueryKeyLicenseType, kStreaming);
-    mPlayPolicy.add(kQueryKeyPlayAllowed, kTrue);
-    mPlayPolicy.add(kQueryKeyRenewAllowed, kTrue);
-}
-
-status_t DrmPlugin::provideKeyResponse(
-        const Vector<uint8_t>& scope,
-        const Vector<uint8_t>& response,
-        Vector<uint8_t>& keySetId) {
-    if (scope.size() == 0 || response.size() == 0) {
-        return android::BAD_VALUE;
-    }
-    sp<Session> session = mSessionLibrary->findSession(scope);
-    if (!session.get()) {
-        return android::ERROR_DRM_SESSION_NOT_OPENED;
-    }
-
-    setPlayPolicy();
-    status_t res = session->provideKeyResponse(response);
-    if (res == android::OK) {
-        // This is for testing AMediaDrm_setOnEventListener only.
-        sendEvent(kDrmPluginEventVendorDefined, 0, &scope, NULL);
-        keySetId.clear();
-    }
-    return res;
-}
-
-status_t DrmPlugin::getPropertyString(
-        const String8& name, String8& value) const {
-    if (name == "vendor") {
-        value = "Google";
-    } else if (name == "version") {
-        value = "1.0";
-    } else if (name == "description") {
-        value = "ClearKey CDM";
-    } else if (name == "algorithms") {
-        value = "";
-    } else if (name == "listenerTestSupport") {
-        value = "true";
-    } else {
-        ALOGE("App requested unknown string property %s", name.string());
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-    return android::OK;
-}
-
-status_t DrmPlugin::queryKeyStatus(
-        const Vector<uint8_t>& sessionId,
-        KeyedVector<String8, String8>& infoMap) const {
-
-    if (sessionId.size() == 0) {
-        return android::BAD_VALUE;
-    }
-
-    infoMap.clear();
-    for (size_t i = 0; i < mPlayPolicy.size(); ++i) {
-        infoMap.add(mPlayPolicy.keyAt(i), mPlayPolicy.valueAt(i));
-    }
-    return android::OK;
-}
-}  // namespace clearkeydrm

diff --git a/drm/mediadrm/plugins/clearkey/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/DrmPlugin.h
deleted file mode 100644
index f37a706..0000000
--- a/drm/mediadrm/plugins/clearkey/DrmPlugin.h
+++ /dev/null

@@ -1,255 +0,0 @@
-/*
- * Copyright (C) 2014 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef CLEARKEY_DRM_PLUGIN_H_
-#define CLEARKEY_DRM_PLUGIN_H_
-
-#include <media/drm/DrmAPI.h>
-#include <media/stagefright/foundation/ABase.h>
-#include <media/stagefright/MediaErrors.h>
-#include <utils/Errors.h>
-#include <utils/KeyedVector.h>
-#include <utils/List.h>
-#include <utils/String8.h>
-#include <utils/Vector.h>
-
-#include "SessionLibrary.h"
-#include "Utils.h"
-
-namespace clearkeydrm {
-
-using android::KeyedVector;
-using android::List;
-using android::status_t;
-using android::String8;
-using android::Vector;
-
-class DrmPlugin : public android::DrmPlugin {
-public:
-    explicit DrmPlugin(SessionLibrary* sessionLibrary);
-
-    virtual ~DrmPlugin() {}
-
-    virtual status_t openSession(Vector<uint8_t>& sessionId);
-
-    virtual status_t closeSession(const Vector<uint8_t>& sessionId);
-
-    virtual status_t getKeyRequest(
-            const Vector<uint8_t>& scope,
-            const Vector<uint8_t>& mimeType,
-            const String8& initDataType,
-            KeyType keyType,
-            const KeyedVector<String8, String8>& optionalParameters,
-            Vector<uint8_t>& request,
-            String8& defaultUrl,
-            DrmPlugin::KeyRequestType *keyRequestType);
-
-    virtual status_t provideKeyResponse(
-            const Vector<uint8_t>& scope,
-            const Vector<uint8_t>& response,
-            Vector<uint8_t>& keySetId);
-
-    virtual status_t removeKeys(const Vector<uint8_t>& sessionId) {
-        if (sessionId.size() == 0) {
-            return android::BAD_VALUE;
-        }
-
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t restoreKeys(
-            const Vector<uint8_t>& sessionId,
-            const Vector<uint8_t>& keySetId) {
-        if (sessionId.size() == 0 || keySetId.size() == 0) {
-            return android::BAD_VALUE;
-        }
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t queryKeyStatus(
-            const Vector<uint8_t>& sessionId,
-            KeyedVector<String8, String8>& infoMap) const;
-
-    virtual status_t getProvisionRequest(
-            const String8& cert_type,
-            const String8& cert_authority,
-            Vector<uint8_t>& request,
-            String8& defaultUrl) {
-        UNUSED(cert_type);
-        UNUSED(cert_authority);
-        UNUSED(request);
-        UNUSED(defaultUrl);
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t provideProvisionResponse(
-            const Vector<uint8_t>& response,
-            Vector<uint8_t>& certificate,
-            Vector<uint8_t>& wrappedKey) {
-        UNUSED(certificate);
-        UNUSED(wrappedKey);
-        if (response.size() == 0) {
-            // empty response
-            return android::BAD_VALUE;
-        }
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t getSecureStops(List<Vector<uint8_t> >& secureStops) {
-        UNUSED(secureStops);
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t getSecureStop(Vector<uint8_t> const &ssid, Vector<uint8_t> &secureStop) {
-        if (ssid.size() == 0) {
-            return android::BAD_VALUE;
-        }
-
-        UNUSED(secureStop);
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t releaseSecureStops(const Vector<uint8_t>& ssRelease) {
-        if (ssRelease.size() == 0) {
-            return android::BAD_VALUE;
-        }
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t releaseAllSecureStops() {
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t getPropertyString(
-            const String8& name, String8& value) const;
-
-    virtual status_t getPropertyByteArray(
-            const String8& name, Vector<uint8_t>& value) const {
-        UNUSED(name);
-        UNUSED(value);
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t setPropertyString(
-            const String8& name, const String8& value) {
-        UNUSED(name);
-        UNUSED(value);
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t setPropertyByteArray(
-            const String8& name, const Vector<uint8_t>& value) {
-        UNUSED(name);
-        UNUSED(value);
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t setCipherAlgorithm(
-            const Vector<uint8_t>& sessionId, const String8& algorithm) {
-        if (sessionId.size() == 0 || algorithm.size() == 0) {
-            return android::BAD_VALUE;
-        }
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t setMacAlgorithm(
-            const Vector<uint8_t>& sessionId, const String8& algorithm) {
-        if (sessionId.size() == 0 || algorithm.size() == 0) {
-            return android::BAD_VALUE;
-        }
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t encrypt(
-            const Vector<uint8_t>& sessionId,
-            const Vector<uint8_t>& keyId,
-            const Vector<uint8_t>& input,
-            const Vector<uint8_t>& iv,
-            Vector<uint8_t>& output) {
-        if (sessionId.size() == 0 || keyId.size() == 0 ||
-                input.size() == 0 || iv.size() == 0) {
-            return android::BAD_VALUE;
-        }
-        UNUSED(output);
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t decrypt(
-            const Vector<uint8_t>& sessionId,
-            const Vector<uint8_t>& keyId,
-            const Vector<uint8_t>& input,
-            const Vector<uint8_t>& iv,
-            Vector<uint8_t>& output) {
-        if (sessionId.size() == 0 || keyId.size() == 0 ||
-                input.size() == 0 || iv.size() == 0) {
-            return android::BAD_VALUE;
-        }
-        UNUSED(output);
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t sign(
-            const Vector<uint8_t>& sessionId,
-            const Vector<uint8_t>& keyId,
-            const Vector<uint8_t>& message,
-            Vector<uint8_t>& signature) {
-        if (sessionId.size() == 0 || keyId.size() == 0 ||
-                message.size() == 0) {
-            return android::BAD_VALUE;
-        }
-        UNUSED(signature);
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t verify(
-            const Vector<uint8_t>& sessionId,
-            const Vector<uint8_t>& keyId,
-            const Vector<uint8_t>& message,
-            const Vector<uint8_t>& signature, bool& match) {
-        if (sessionId.size() == 0 || keyId.size() == 0 ||
-                message.size() == 0 || signature.size() == 0) {
-            return android::BAD_VALUE;
-        }
-        UNUSED(match);
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-    virtual status_t signRSA(
-            const Vector<uint8_t>& sessionId,
-            const String8& algorithm,
-            const Vector<uint8_t>& message,
-            const Vector<uint8_t>& wrappedKey,
-            Vector<uint8_t>& signature) {
-        if (sessionId.size() == 0 || algorithm.size() == 0 ||
-                message.size() == 0 || wrappedKey.size() == 0) {
-            return android::BAD_VALUE;
-        }
-        UNUSED(signature);
-        return android::ERROR_DRM_CANNOT_HANDLE;
-    }
-
-private:
-    void setPlayPolicy();
-
-    android::KeyedVector<android::String8, android::String8> mPlayPolicy;
-    SessionLibrary* mSessionLibrary;
-
-    DISALLOW_EVIL_CONSTRUCTORS(DrmPlugin);
-};
-
-} // namespace clearkeydrm
-
-#endif // CLEARKEY_DRM_PLUGIN_H_

diff --git a/drm/mediadrm/plugins/clearkey/Session.cpp b/drm/mediadrm/plugins/clearkey/Session.cpp
deleted file mode 100644
index d210f5e..0000000
--- a/drm/mediadrm/plugins/clearkey/Session.cpp
+++ /dev/null

@@ -1,85 +0,0 @@
-/*
- * Copyright (C) 2014 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "ClearKeyCryptoPlugin"
-#include <utils/Log.h>
-
-#include <media/stagefright/MediaErrors.h>
-#include <utils/String8.h>
-
-#include "Session.h"
-
-#include "AesCtrDecryptor.h"
-#include "InitDataParser.h"
-#include "JsonWebKey.h"
-
-namespace clearkeydrm {
-
-using android::Mutex;
-using android::String8;
-using android::Vector;
-using android::status_t;
-
-status_t Session::getKeyRequest(
-        const Vector<uint8_t>& initData,
-        const String8& mimeType,
-        Vector<uint8_t>* keyRequest) const {
-    InitDataParser parser;
-    return parser.parse(initData, mimeType, keyRequest);
-}
-
-status_t Session::provideKeyResponse(const Vector<uint8_t>& response) {
-    String8 responseString(
-            reinterpret_cast<const char*>(response.array()), response.size());
-    KeyMap keys;
-
-    Mutex::Autolock lock(mMapLock);
-    JsonWebKey parser;
-    if (parser.extractKeysFromJsonWebKeySet(responseString, &keys)) {
-        for (size_t i = 0; i < keys.size(); ++i) {
-            const KeyMap::key_type& keyId = keys.keyAt(i);
-            const KeyMap::value_type& key = keys.valueAt(i);
-            mKeyMap.add(keyId, key);
-        }
-        return android::OK;
-    } else {
-        return android::ERROR_DRM_UNKNOWN;
-    }
-}
-
-status_t Session::decrypt(
-        const KeyId keyId, const Iv iv, const void* source,
-        void* destination, const SubSample* subSamples,
-        size_t numSubSamples, size_t* bytesDecryptedOut) {
-    Mutex::Autolock lock(mMapLock);
-
-    Vector<uint8_t> keyIdVector;
-    keyIdVector.appendArray(keyId, kBlockSize);
-    if (mKeyMap.indexOfKey(keyIdVector) < 0) {
-        return android::ERROR_DRM_NO_LICENSE;
-    }
-
-    const Vector<uint8_t>& key = mKeyMap.valueFor(keyIdVector);
-    AesCtrDecryptor decryptor;
-    return decryptor.decrypt(
-            key, iv,
-            reinterpret_cast<const uint8_t*>(source),
-            reinterpret_cast<uint8_t*>(destination), subSamples,
-            numSubSamples, bytesDecryptedOut);
-}
-
-} // namespace clearkeydrm

diff --git a/drm/mediadrm/plugins/clearkey/SessionLibrary.cpp b/drm/mediadrm/plugins/clearkey/SessionLibrary.cpp
deleted file mode 100644
index 0419f97..0000000
--- a/drm/mediadrm/plugins/clearkey/SessionLibrary.cpp
+++ /dev/null

@@ -1,74 +0,0 @@
-/*
- * Copyright (C) 2014 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "ClearKeyCryptoPlugin"
-#include <utils/Log.h>
-
-#include <utils/String8.h>
-
-#include "SessionLibrary.h"
-
-namespace clearkeydrm {
-
-using android::Mutex;
-using android::sp;
-using android::String8;
-using android::Vector;
-
-Mutex SessionLibrary::sSingletonLock;
-SessionLibrary* SessionLibrary::sSingleton = NULL;
-
-SessionLibrary* SessionLibrary::get() {
-    Mutex::Autolock lock(sSingletonLock);
-
-    if (sSingleton == NULL) {
-        ALOGD("Instantiating Session Library Singleton.");
-        sSingleton = new SessionLibrary();
-    }
-
-    return sSingleton;
-}
-
-sp<Session> SessionLibrary::createSession() {
-    Mutex::Autolock lock(mSessionsLock);
-
-    String8 sessionIdString = String8::format("%u", mNextSessionId);
-    mNextSessionId += 1;
-    Vector<uint8_t> sessionId;
-    sessionId.appendArray(
-            reinterpret_cast<const uint8_t*>(sessionIdString.string()),
-            sessionIdString.size());
-
-    mSessions.add(sessionId, new Session(sessionId));
-    return mSessions.valueFor(sessionId);
-}
-
-sp<Session> SessionLibrary::findSession(
-        const Vector<uint8_t>& sessionId) {
-    Mutex::Autolock lock(mSessionsLock);
-    if (mSessions.indexOfKey(sessionId) < 0) {
-        return sp<Session>(NULL);
-    }
-    return mSessions.valueFor(sessionId);
-}
-
-void SessionLibrary::destroySession(const sp<Session>& session) {
-    Mutex::Autolock lock(mSessionsLock);\
-    mSessions.removeItem(session->sessionId());
-}
-
-} // namespace clearkeydrm

diff --git a/drm/mediadrm/plugins/clearkey/common/Android.bp b/drm/mediadrm/plugins/clearkey/common/Android.bp
new file mode 100644
index 0000000..2c674e1
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/common/Android.bp

@@ -0,0 +1,38 @@
+//
+// Copyright (C) 2018 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//      http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+
+cc_library_static {
+    name: "libclearkeycommon",
+    vendor: true,
+
+    srcs: [
+        "ClearKeyUUID.cpp",
+        "Utils.cpp",
+    ],
+
+    cflags: ["-Wall", "-Werror"],
+
+    include_dirs: ["frameworks/av/include"],
+
+    shared_libs: ["libutils"],
+
+    export_include_dirs: ["include"],
+
+    sanitize: {
+        integer_overflow: true,
+    },
+}
+

diff --git a/drm/mediadrm/plugins/clearkey/ClearKeyUUID.cpp b/drm/mediadrm/plugins/clearkey/common/ClearKeyUUID.cpp
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/ClearKeyUUID.cpp
rename to drm/mediadrm/plugins/clearkey/common/ClearKeyUUID.cpp


diff --git a/drm/mediadrm/plugins/clearkey/Utils.cpp b/drm/mediadrm/plugins/clearkey/common/Utils.cpp
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/Utils.cpp
rename to drm/mediadrm/plugins/clearkey/common/Utils.cpp


diff --git a/drm/mediadrm/plugins/clearkey/ClearKeyUUID.h b/drm/mediadrm/plugins/clearkey/common/include/ClearKeyUUID.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/ClearKeyUUID.h
rename to drm/mediadrm/plugins/clearkey/common/include/ClearKeyUUID.h


diff --git a/drm/mediadrm/plugins/clearkey/MimeType.h b/drm/mediadrm/plugins/clearkey/common/include/MimeType.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/MimeType.h
rename to drm/mediadrm/plugins/clearkey/common/include/MimeType.h


diff --git a/drm/mediadrm/plugins/clearkey/Utils.h b/drm/mediadrm/plugins/clearkey/common/include/Utils.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/Utils.h
rename to drm/mediadrm/plugins/clearkey/common/include/Utils.h


diff --git a/drm/mediadrm/plugins/clearkey/AesCtrDecryptor.cpp b/drm/mediadrm/plugins/clearkey/default/AesCtrDecryptor.cpp
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/AesCtrDecryptor.cpp
rename to drm/mediadrm/plugins/clearkey/default/AesCtrDecryptor.cpp


diff --git a/drm/mediadrm/plugins/clearkey/default/Android.bp b/drm/mediadrm/plugins/clearkey/default/Android.bp
new file mode 100644
index 0000000..7ba5708
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/default/Android.bp

@@ -0,0 +1,67 @@
+//
+// Copyright (C) 2014 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//      http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+
+cc_library_shared {
+    name: "libdrmclearkeyplugin",
+    vendor: true,
+
+    srcs: [
+        "AesCtrDecryptor.cpp",
+        "CreatePluginFactories.cpp",
+        "CryptoFactory.cpp",
+        "CryptoPlugin.cpp",
+        "DrmFactory.cpp",
+        "DrmPlugin.cpp",
+        "InitDataParser.cpp",
+        "JsonWebKey.cpp",
+        "Session.cpp",
+        "SessionLibrary.cpp",
+    ],
+
+    relative_install_path: "mediadrm",
+
+    cflags: ["-Wall", "-Werror"],
+
+    shared_libs: [
+        "libcrypto",
+        "liblog",
+        "libstagefright_foundation",
+        "libutils",
+    ],
+
+    static_libs: [
+        "libclearkeycommon",
+        "libjsmn"
+    ],
+
+    local_include_dirs: ["include"],
+    export_include_dirs: ["include"],
+    export_static_lib_headers: ["libjsmn"],
+
+    include_dirs: [
+        "frameworks/native/include",
+        "frameworks/av/include",
+    ],
+
+    sanitize: {
+        integer_overflow: true,
+    },
+}
+
+//########################################################################
+// Build unit tests
+
+subdirs = ["tests"]

diff --git a/drm/mediadrm/plugins/clearkey/CreatePluginFactories.cpp b/drm/mediadrm/plugins/clearkey/default/CreatePluginFactories.cpp
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/CreatePluginFactories.cpp
rename to drm/mediadrm/plugins/clearkey/default/CreatePluginFactories.cpp


diff --git a/drm/mediadrm/plugins/clearkey/default/CryptoFactory.cpp b/drm/mediadrm/plugins/clearkey/default/CryptoFactory.cpp
new file mode 100644
index 0000000..f15f92b
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/default/CryptoFactory.cpp

@@ -0,0 +1,60 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "ClearKeyCryptoFactory"
+#include <utils/Log.h>
+
+#include <utils/Errors.h>
+#include <utils/StrongPointer.h>
+
+#include "CryptoFactory.h"
+
+#include "ClearKeyUUID.h"
+#include "CryptoPlugin.h"
+#include "Session.h"
+#include "SessionLibrary.h"
+
+namespace clearkeydrm {
+
+bool CryptoFactory::isCryptoSchemeSupported(const uint8_t uuid[16]) const {
+    return isClearKeyUUID(uuid);
+}
+
+android::status_t CryptoFactory::createPlugin(
+        const uint8_t uuid[16],
+        const void* data, size_t size,
+        android::CryptoPlugin** plugin) {
+    if (!isCryptoSchemeSupported(uuid)) {
+        *plugin = NULL;
+        return android::BAD_VALUE;
+    }
+
+    android::Vector<uint8_t> sessionId;
+    sessionId.appendArray(reinterpret_cast<const uint8_t*>(data), size);
+
+    CryptoPlugin *clearKeyPlugin = new CryptoPlugin(sessionId);
+    android::status_t result = clearKeyPlugin->getInitStatus();
+    if (result == android::OK) {
+        *plugin = clearKeyPlugin;
+    } else {
+        delete clearKeyPlugin;
+        *plugin = NULL;
+    }
+    return result;
+}
+
+} // namespace clearkeydrm

diff --git a/drm/mediadrm/plugins/clearkey/CryptoPlugin.cpp b/drm/mediadrm/plugins/clearkey/default/CryptoPlugin.cpp
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/CryptoPlugin.cpp
rename to drm/mediadrm/plugins/clearkey/default/CryptoPlugin.cpp


diff --git a/drm/mediadrm/plugins/clearkey/default/DrmFactory.cpp b/drm/mediadrm/plugins/clearkey/default/DrmFactory.cpp
new file mode 100644
index 0000000..8301e40
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/default/DrmFactory.cpp

@@ -0,0 +1,58 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "ClearKeyDrmFactory"
+#include <utils/Log.h>
+
+#include <utils/Errors.h>
+
+#include "DrmFactory.h"
+
+#include "DrmPlugin.h"
+#include "ClearKeyUUID.h"
+#include "MimeType.h"
+#include "SessionLibrary.h"
+
+namespace clearkeydrm {
+
+bool DrmFactory::isCryptoSchemeSupported(const uint8_t uuid[16]) {
+    return isClearKeyUUID(uuid);
+}
+
+bool DrmFactory::isContentTypeSupported(const android::String8 &type) {
+    // This should match the types handed by InitDataParser.
+    return type == kIsoBmffVideoMimeType ||
+        type == kIsoBmffAudioMimeType ||
+        type == kCencInitDataFormat ||
+        type == kWebmVideoMimeType ||
+        type == kWebmAudioMimeType ||
+        type == kWebmInitDataFormat;
+}
+
+android::status_t DrmFactory::createDrmPlugin(
+        const uint8_t uuid[16],
+        android::DrmPlugin** plugin) {
+    if (!isCryptoSchemeSupported(uuid)) {
+        *plugin = NULL;
+        return android::BAD_VALUE;
+    }
+
+    *plugin = new DrmPlugin(SessionLibrary::get());
+    return android::OK;
+}
+
+} // namespace clearkeydrm

diff --git a/drm/mediadrm/plugins/clearkey/default/DrmPlugin.cpp b/drm/mediadrm/plugins/clearkey/default/DrmPlugin.cpp
new file mode 100644
index 0000000..1b8b8c1
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/default/DrmPlugin.cpp

@@ -0,0 +1,214 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "ClearKeyDrmPlugin"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaErrors.h>
+#include <utils/StrongPointer.h>
+
+#include "DrmPlugin.h"
+#include "ClearKeyDrmProperties.h"
+#include "Session.h"
+
+namespace {
+const android::String8 kStreaming("Streaming");
+const android::String8 kOffline("Offline");
+const android::String8 kTrue("True");
+
+const android::String8 kQueryKeyLicenseType("LicenseType");
+    // Value: "Streaming" or "Offline"
+const android::String8 kQueryKeyPlayAllowed("PlayAllowed");
+    // Value: "True" or "False"
+const android::String8 kQueryKeyRenewAllowed("RenewAllowed");
+    // Value: "True" or "False"
+};
+
+namespace clearkeydrm {
+
+using android::sp;
+
+DrmPlugin::DrmPlugin(SessionLibrary* sessionLibrary)
+        : mSessionLibrary(sessionLibrary) {
+
+    mPlayPolicy.clear();
+    initProperties();
+}
+
+void DrmPlugin::initProperties() {
+    mStringProperties.clear();
+    mStringProperties.add(kVendorKey, kVendorValue);
+    mStringProperties.add(kVersionKey, kVersionValue);
+    mStringProperties.add(kPluginDescriptionKey, kPluginDescriptionValue);
+    mStringProperties.add(kAlgorithmsKey, kAlgorithmsValue);
+    mStringProperties.add(kListenerTestSupportKey, kListenerTestSupportValue);
+
+    Vector<uint8_t> testDeviceId;
+    testDeviceId.appendArray(kTestDeviceIdData, sizeof(kTestDeviceIdData) / sizeof(uint8_t));
+    mByteArrayProperties.add(kDeviceIdKey, testDeviceId);
+}
+
+status_t DrmPlugin::openSession(Vector<uint8_t>& sessionId) {
+    sp<Session> session = mSessionLibrary->createSession();
+    sessionId = session->sessionId();
+    return android::OK;
+}
+
+status_t DrmPlugin::closeSession(const Vector<uint8_t>& sessionId) {
+    sp<Session> session = mSessionLibrary->findSession(sessionId);
+    if (sessionId.size() == 0) {
+        return android::BAD_VALUE;
+    }
+    if (session.get()) {
+        mSessionLibrary->destroySession(session);
+        return android::OK;
+    }
+    return android::ERROR_DRM_SESSION_NOT_OPENED;
+}
+
+status_t DrmPlugin::getKeyRequest(
+        const Vector<uint8_t>& scope,
+        const Vector<uint8_t>& initData,
+        const String8& mimeType,
+        KeyType keyType,
+        const KeyedVector<String8, String8>& optionalParameters,
+        Vector<uint8_t>& request,
+        String8& defaultUrl,
+        DrmPlugin::KeyRequestType *keyRequestType) {
+    UNUSED(optionalParameters);
+    if (scope.size() == 0) {
+        return android::BAD_VALUE;
+    }
+
+    if (keyType != kKeyType_Streaming) {
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    *keyRequestType = DrmPlugin::kKeyRequestType_Initial;
+    defaultUrl.clear();
+    sp<Session> session = mSessionLibrary->findSession(scope);
+    if (!session.get()) {
+        return android::ERROR_DRM_SESSION_NOT_OPENED;
+    }
+
+    return session->getKeyRequest(initData, mimeType, &request);
+}
+
+void DrmPlugin::setPlayPolicy() {
+    mPlayPolicy.clear();
+    mPlayPolicy.add(kQueryKeyLicenseType, kStreaming);
+    mPlayPolicy.add(kQueryKeyPlayAllowed, kTrue);
+    mPlayPolicy.add(kQueryKeyRenewAllowed, kTrue);
+}
+
+status_t DrmPlugin::provideKeyResponse(
+        const Vector<uint8_t>& scope,
+        const Vector<uint8_t>& response,
+        Vector<uint8_t>& keySetId) {
+    if (scope.size() == 0 || response.size() == 0) {
+        return android::BAD_VALUE;
+    }
+    sp<Session> session = mSessionLibrary->findSession(scope);
+    if (!session.get()) {
+        return android::ERROR_DRM_SESSION_NOT_OPENED;
+    }
+
+    setPlayPolicy();
+    status_t res = session->provideKeyResponse(response);
+    if (res == android::OK) {
+        // This is for testing AMediaDrm_setOnEventListener only.
+        sendEvent(kDrmPluginEventVendorDefined, 0, &scope, NULL);
+        keySetId.clear();
+    }
+    return res;
+}
+
+status_t DrmPlugin::getPropertyByteArray(
+        const String8& name, Vector<uint8_t>& value) const {
+    ssize_t index = mByteArrayProperties.indexOfKey(name);
+    if (index < 0) {
+        ALOGE("App requested unknown property: %s", name.string());
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+    value = mByteArrayProperties.valueAt(index);
+    return android::OK;
+}
+
+status_t DrmPlugin::setPropertyByteArray(
+        const String8& name, const Vector<uint8_t>& value)
+{
+    UNUSED(value);
+    if (0 == name.compare(kDeviceIdKey)) {
+        ALOGD("Cannot set immutable property: %s", name.string());
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    // Setting of undefined properties is not supported
+    ALOGE("Failed to set property byte array, key=%s", name.string());
+    return android::ERROR_DRM_CANNOT_HANDLE;
+}
+
+status_t DrmPlugin::getPropertyString(
+        const String8& name, String8& value) const {
+    ssize_t index = mStringProperties.indexOfKey(name);
+    if (index < 0) {
+        ALOGE("App requested unknown property: %s", name.string());
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+    value = mStringProperties.valueAt(index);
+    return android::OK;
+}
+
+status_t DrmPlugin::setPropertyString(
+        const String8& name, const String8& value) {
+    String8 immutableKeys;
+    immutableKeys.appendFormat("%s,%s,%s,%s",
+            kAlgorithmsKey.string(), kPluginDescriptionKey.string(),
+            kVendorKey.string(), kVersionKey.string());
+    if (immutableKeys.contains(name.string())) {
+        ALOGD("Cannot set immutable property: %s", name.string());
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    ssize_t index = mStringProperties.indexOfKey(name);
+    if (index < 0) {
+        ALOGE("Cannot set undefined property string, key=%s", name.string());
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    if (mStringProperties.add(name, value) < 0) {
+        ALOGE("Failed to set property string, key=%s", name.string());
+        return android::ERROR_DRM_UNKNOWN;
+    }
+    return android::OK;
+}
+
+status_t DrmPlugin::queryKeyStatus(
+        const Vector<uint8_t>& sessionId,
+        KeyedVector<String8, String8>& infoMap) const {
+
+    if (sessionId.size() == 0) {
+        return android::BAD_VALUE;
+    }
+
+    infoMap.clear();
+    for (size_t i = 0; i < mPlayPolicy.size(); ++i) {
+        infoMap.add(mPlayPolicy.keyAt(i), mPlayPolicy.valueAt(i));
+    }
+    return android::OK;
+}
+}  // namespace clearkeydrm

diff --git a/drm/mediadrm/plugins/clearkey/InitDataParser.cpp b/drm/mediadrm/plugins/clearkey/default/InitDataParser.cpp
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/InitDataParser.cpp
rename to drm/mediadrm/plugins/clearkey/default/InitDataParser.cpp


diff --git a/drm/mediadrm/plugins/clearkey/JsonWebKey.cpp b/drm/mediadrm/plugins/clearkey/default/JsonWebKey.cpp
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/JsonWebKey.cpp
rename to drm/mediadrm/plugins/clearkey/default/JsonWebKey.cpp


diff --git a/drm/mediadrm/plugins/clearkey/default/Session.cpp b/drm/mediadrm/plugins/clearkey/default/Session.cpp
new file mode 100644
index 0000000..b3ceaec
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/default/Session.cpp

@@ -0,0 +1,85 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "ClearKeySession"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaErrors.h>
+#include <utils/String8.h>
+
+#include "Session.h"
+
+#include "AesCtrDecryptor.h"
+#include "InitDataParser.h"
+#include "JsonWebKey.h"
+
+namespace clearkeydrm {
+
+using android::Mutex;
+using android::String8;
+using android::Vector;
+using android::status_t;
+
+status_t Session::getKeyRequest(
+        const Vector<uint8_t>& initData,
+        const String8& mimeType,
+        Vector<uint8_t>* keyRequest) const {
+    InitDataParser parser;
+    return parser.parse(initData, mimeType, keyRequest);
+}
+
+status_t Session::provideKeyResponse(const Vector<uint8_t>& response) {
+    String8 responseString(
+            reinterpret_cast<const char*>(response.array()), response.size());
+    KeyMap keys;
+
+    Mutex::Autolock lock(mMapLock);
+    JsonWebKey parser;
+    if (parser.extractKeysFromJsonWebKeySet(responseString, &keys)) {
+        for (size_t i = 0; i < keys.size(); ++i) {
+            const KeyMap::key_type& keyId = keys.keyAt(i);
+            const KeyMap::value_type& key = keys.valueAt(i);
+            mKeyMap.add(keyId, key);
+        }
+        return android::OK;
+    } else {
+        return android::ERROR_DRM_UNKNOWN;
+    }
+}
+
+status_t Session::decrypt(
+        const KeyId keyId, const Iv iv, const void* source,
+        void* destination, const SubSample* subSamples,
+        size_t numSubSamples, size_t* bytesDecryptedOut) {
+    Mutex::Autolock lock(mMapLock);
+
+    Vector<uint8_t> keyIdVector;
+    keyIdVector.appendArray(keyId, kBlockSize);
+    if (mKeyMap.indexOfKey(keyIdVector) < 0) {
+        return android::ERROR_DRM_NO_LICENSE;
+    }
+
+    const Vector<uint8_t>& key = mKeyMap.valueFor(keyIdVector);
+    AesCtrDecryptor decryptor;
+    return decryptor.decrypt(
+            key, iv,
+            reinterpret_cast<const uint8_t*>(source),
+            reinterpret_cast<uint8_t*>(destination), subSamples,
+            numSubSamples, bytesDecryptedOut);
+}
+
+} // namespace clearkeydrm

diff --git a/drm/mediadrm/plugins/clearkey/default/SessionLibrary.cpp b/drm/mediadrm/plugins/clearkey/default/SessionLibrary.cpp
new file mode 100644
index 0000000..529230e
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/default/SessionLibrary.cpp

@@ -0,0 +1,74 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "ClearKeySessionLibrary"
+#include <utils/Log.h>
+
+#include <utils/String8.h>
+
+#include "SessionLibrary.h"
+
+namespace clearkeydrm {
+
+using android::Mutex;
+using android::sp;
+using android::String8;
+using android::Vector;
+
+Mutex SessionLibrary::sSingletonLock;
+SessionLibrary* SessionLibrary::sSingleton = NULL;
+
+SessionLibrary* SessionLibrary::get() {
+    Mutex::Autolock lock(sSingletonLock);
+
+    if (sSingleton == NULL) {
+        ALOGD("Instantiating Session Library Singleton.");
+        sSingleton = new SessionLibrary();
+    }
+
+    return sSingleton;
+}
+
+sp<Session> SessionLibrary::createSession() {
+    Mutex::Autolock lock(mSessionsLock);
+
+    String8 sessionIdString = String8::format("%u", mNextSessionId);
+    mNextSessionId += 1;
+    Vector<uint8_t> sessionId;
+    sessionId.appendArray(
+            reinterpret_cast<const uint8_t*>(sessionIdString.string()),
+            sessionIdString.size());
+
+    mSessions.add(sessionId, new Session(sessionId));
+    return mSessions.valueFor(sessionId);
+}
+
+sp<Session> SessionLibrary::findSession(
+        const Vector<uint8_t>& sessionId) {
+    Mutex::Autolock lock(mSessionsLock);
+    if (mSessions.indexOfKey(sessionId) < 0) {
+        return sp<Session>(NULL);
+    }
+    return mSessions.valueFor(sessionId);
+}
+
+void SessionLibrary::destroySession(const sp<Session>& session) {
+    Mutex::Autolock lock(mSessionsLock);\
+    mSessions.removeItem(session->sessionId());
+}
+
+} // namespace clearkeydrm

diff --git a/drm/mediadrm/plugins/clearkey/AesCtrDecryptor.h b/drm/mediadrm/plugins/clearkey/default/include/AesCtrDecryptor.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/AesCtrDecryptor.h
rename to drm/mediadrm/plugins/clearkey/default/include/AesCtrDecryptor.h


diff --git a/drm/mediadrm/plugins/clearkey/default/include/ClearKeyDrmProperties.h b/drm/mediadrm/plugins/clearkey/default/include/ClearKeyDrmProperties.h
new file mode 100644
index 0000000..a99e174
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/default/include/ClearKeyDrmProperties.h

@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_DRM_PROPERTIES_H_
+#define CLEARKEY_DRM_PROPERTIES_H_
+
+#include <utils/String8.h>
+
+namespace clearkeydrm {
+
+static const android::String8 kVendorKey("vendor");
+static const android::String8 kVendorValue("Google");
+static const android::String8 kVersionKey("version");
+static const android::String8 kVersionValue("1.0");
+static const android::String8 kPluginDescriptionKey("description");
+static const android::String8 kPluginDescriptionValue("ClearKey CDM");
+static const android::String8 kAlgorithmsKey("algorithms");
+static const android::String8 kAlgorithmsValue("");
+static const android::String8 kListenerTestSupportKey("listenerTestSupport");
+static const android::String8 kListenerTestSupportValue("true");
+
+static const android::String8 kDeviceIdKey("deviceId");
+static const uint8_t kTestDeviceIdData[] =
+        {0x0, 0x1, 0x2, 0x3, 0x4, 0x5, 0x6, 0x7,
+         0x8, 0x9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf};
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_DRM_PROPERTIES_H_

diff --git a/drm/mediadrm/plugins/clearkey/ClearKeyTypes.h b/drm/mediadrm/plugins/clearkey/default/include/ClearKeyTypes.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/ClearKeyTypes.h
rename to drm/mediadrm/plugins/clearkey/default/include/ClearKeyTypes.h


diff --git a/drm/mediadrm/plugins/clearkey/CreatePluginFactories.h b/drm/mediadrm/plugins/clearkey/default/include/CreatePluginFactories.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/CreatePluginFactories.h
rename to drm/mediadrm/plugins/clearkey/default/include/CreatePluginFactories.h


diff --git a/drm/mediadrm/plugins/clearkey/CryptoFactory.h b/drm/mediadrm/plugins/clearkey/default/include/CryptoFactory.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/CryptoFactory.h
rename to drm/mediadrm/plugins/clearkey/default/include/CryptoFactory.h


diff --git a/drm/mediadrm/plugins/clearkey/CryptoPlugin.h b/drm/mediadrm/plugins/clearkey/default/include/CryptoPlugin.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/CryptoPlugin.h
rename to drm/mediadrm/plugins/clearkey/default/include/CryptoPlugin.h


diff --git a/drm/mediadrm/plugins/clearkey/DrmFactory.h b/drm/mediadrm/plugins/clearkey/default/include/DrmFactory.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/DrmFactory.h
rename to drm/mediadrm/plugins/clearkey/default/include/DrmFactory.h


diff --git a/drm/mediadrm/plugins/clearkey/default/include/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/default/include/DrmPlugin.h
new file mode 100644
index 0000000..4fa42e5
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/default/include/DrmPlugin.h

@@ -0,0 +1,276 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_DRM_PLUGIN_H_
+#define CLEARKEY_DRM_PLUGIN_H_
+
+#include <media/drm/DrmAPI.h>
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/MediaErrors.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/List.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+
+#include "SessionLibrary.h"
+#include "Utils.h"
+
+namespace clearkeydrm {
+
+using android::KeyedVector;
+using android::List;
+using android::status_t;
+using android::String8;
+using android::Vector;
+
+class DrmPlugin : public android::DrmPlugin {
+public:
+    explicit DrmPlugin(SessionLibrary* sessionLibrary);
+
+    virtual ~DrmPlugin() {}
+
+    virtual status_t openSession(Vector<uint8_t>& sessionId);
+
+    virtual status_t closeSession(const Vector<uint8_t>& sessionId);
+
+    virtual status_t getKeyRequest(
+            const Vector<uint8_t>& scope,
+            const Vector<uint8_t>& mimeType,
+            const String8& initDataType,
+            KeyType keyType,
+            const KeyedVector<String8, String8>& optionalParameters,
+            Vector<uint8_t>& request,
+            String8& defaultUrl,
+            DrmPlugin::KeyRequestType *keyRequestType);
+
+    virtual status_t provideKeyResponse(
+            const Vector<uint8_t>& scope,
+            const Vector<uint8_t>& response,
+            Vector<uint8_t>& keySetId);
+
+    virtual status_t removeKeys(const Vector<uint8_t>& sessionId) {
+        if (sessionId.size() == 0) {
+            return android::BAD_VALUE;
+        }
+
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t restoreKeys(
+            const Vector<uint8_t>& sessionId,
+            const Vector<uint8_t>& keySetId) {
+        if (sessionId.size() == 0 || keySetId.size() == 0) {
+            return android::BAD_VALUE;
+        }
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t queryKeyStatus(
+            const Vector<uint8_t>& sessionId,
+            KeyedVector<String8, String8>& infoMap) const;
+
+    virtual status_t getProvisionRequest(
+            const String8& cert_type,
+            const String8& cert_authority,
+            Vector<uint8_t>& request,
+            String8& defaultUrl) {
+        UNUSED(cert_type);
+        UNUSED(cert_authority);
+        UNUSED(request);
+        UNUSED(defaultUrl);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t provideProvisionResponse(
+            const Vector<uint8_t>& response,
+            Vector<uint8_t>& certificate,
+            Vector<uint8_t>& wrappedKey) {
+        UNUSED(certificate);
+        UNUSED(wrappedKey);
+        if (response.size() == 0) {
+            // empty response
+            return android::BAD_VALUE;
+        }
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t getSecureStops(List<Vector<uint8_t> >& secureStops) {
+        UNUSED(secureStops);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t getSecureStop(Vector<uint8_t> const &ssid, Vector<uint8_t> &secureStop) {
+        if (ssid.size() == 0) {
+            return android::BAD_VALUE;
+        }
+
+        UNUSED(secureStop);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t releaseSecureStops(const Vector<uint8_t>& ssRelease) {
+        if (ssRelease.size() == 0) {
+            return android::BAD_VALUE;
+        }
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t releaseAllSecureStops() {
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t getHdcpLevels(HdcpLevel *connectedLevel,
+            HdcpLevel *maxLevel) const {
+        UNUSED(connectedLevel);
+        UNUSED(maxLevel);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+
+    virtual status_t getNumberOfSessions(uint32_t *currentSessions,
+            uint32_t *maxSessions) const {
+        UNUSED(currentSessions);
+        UNUSED(maxSessions);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t getSecurityLevel(Vector<uint8_t> const &sessionId,
+            SecurityLevel *level) const {
+        UNUSED(sessionId);
+        UNUSED(level);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t setSecurityLevel(Vector<uint8_t> const &sessionId,
+            const SecurityLevel& level) {
+        UNUSED(sessionId);
+        UNUSED(level);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t getPropertyString(
+            const String8& name, String8& value) const;
+
+    virtual status_t getPropertyByteArray(
+            const String8& name, Vector<uint8_t>& value) const;
+
+    virtual status_t setPropertyString(
+            const String8& name, const String8& value);
+
+    virtual status_t setPropertyByteArray(
+            const String8& name, const Vector<uint8_t>& value);
+
+    virtual status_t setCipherAlgorithm(
+            const Vector<uint8_t>& sessionId, const String8& algorithm) {
+        if (sessionId.size() == 0 || algorithm.size() == 0) {
+            return android::BAD_VALUE;
+        }
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t setMacAlgorithm(
+            const Vector<uint8_t>& sessionId, const String8& algorithm) {
+        if (sessionId.size() == 0 || algorithm.size() == 0) {
+            return android::BAD_VALUE;
+        }
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t encrypt(
+            const Vector<uint8_t>& sessionId,
+            const Vector<uint8_t>& keyId,
+            const Vector<uint8_t>& input,
+            const Vector<uint8_t>& iv,
+            Vector<uint8_t>& output) {
+        if (sessionId.size() == 0 || keyId.size() == 0 ||
+                input.size() == 0 || iv.size() == 0) {
+            return android::BAD_VALUE;
+        }
+        UNUSED(output);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t decrypt(
+            const Vector<uint8_t>& sessionId,
+            const Vector<uint8_t>& keyId,
+            const Vector<uint8_t>& input,
+            const Vector<uint8_t>& iv,
+            Vector<uint8_t>& output) {
+        if (sessionId.size() == 0 || keyId.size() == 0 ||
+                input.size() == 0 || iv.size() == 0) {
+            return android::BAD_VALUE;
+        }
+        UNUSED(output);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t sign(
+            const Vector<uint8_t>& sessionId,
+            const Vector<uint8_t>& keyId,
+            const Vector<uint8_t>& message,
+            Vector<uint8_t>& signature) {
+        if (sessionId.size() == 0 || keyId.size() == 0 ||
+                message.size() == 0) {
+            return android::BAD_VALUE;
+        }
+        UNUSED(signature);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t verify(
+            const Vector<uint8_t>& sessionId,
+            const Vector<uint8_t>& keyId,
+            const Vector<uint8_t>& message,
+            const Vector<uint8_t>& signature, bool& match) {
+        if (sessionId.size() == 0 || keyId.size() == 0 ||
+                message.size() == 0 || signature.size() == 0) {
+            return android::BAD_VALUE;
+        }
+        UNUSED(match);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    virtual status_t signRSA(
+            const Vector<uint8_t>& sessionId,
+            const String8& algorithm,
+            const Vector<uint8_t>& message,
+            const Vector<uint8_t>& wrappedKey,
+            Vector<uint8_t>& signature) {
+        if (sessionId.size() == 0 || algorithm.size() == 0 ||
+                message.size() == 0 || wrappedKey.size() == 0) {
+            return android::BAD_VALUE;
+        }
+        UNUSED(signature);
+        return android::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+private:
+    void initProperties();
+    void setPlayPolicy();
+
+    android::KeyedVector<String8, String8> mPlayPolicy;
+    android::KeyedVector<String8, String8> mStringProperties;
+    android::KeyedVector<String8, Vector<uint8_t>> mByteArrayProperties;
+
+    SessionLibrary* mSessionLibrary;
+
+    DISALLOW_EVIL_CONSTRUCTORS(DrmPlugin);
+};
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_DRM_PLUGIN_H_

diff --git a/drm/mediadrm/plugins/clearkey/InitDataParser.h b/drm/mediadrm/plugins/clearkey/default/include/InitDataParser.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/InitDataParser.h
rename to drm/mediadrm/plugins/clearkey/default/include/InitDataParser.h


diff --git a/drm/mediadrm/plugins/clearkey/JsonWebKey.h b/drm/mediadrm/plugins/clearkey/default/include/JsonWebKey.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/JsonWebKey.h
rename to drm/mediadrm/plugins/clearkey/default/include/JsonWebKey.h


diff --git a/drm/mediadrm/plugins/clearkey/Session.h b/drm/mediadrm/plugins/clearkey/default/include/Session.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/Session.h
rename to drm/mediadrm/plugins/clearkey/default/include/Session.h


diff --git a/drm/mediadrm/plugins/clearkey/SessionLibrary.h b/drm/mediadrm/plugins/clearkey/default/include/SessionLibrary.h
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/SessionLibrary.h
rename to drm/mediadrm/plugins/clearkey/default/include/SessionLibrary.h


diff --git a/drm/mediadrm/plugins/clearkey/tests/AesCtrDecryptorUnittest.cpp b/drm/mediadrm/plugins/clearkey/default/tests/AesCtrDecryptorUnittest.cpp
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/tests/AesCtrDecryptorUnittest.cpp
rename to drm/mediadrm/plugins/clearkey/default/tests/AesCtrDecryptorUnittest.cpp


diff --git a/drm/mediadrm/plugins/clearkey/default/tests/Android.bp b/drm/mediadrm/plugins/clearkey/default/tests/Android.bp
new file mode 100644
index 0000000..4419865
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/default/tests/Android.bp

@@ -0,0 +1,42 @@
+//
+// Copyright (C) 2014 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//      http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+// ----------------------------------------------------------------
+// Builds ClearKey Drm Tests
+//
+
+cc_test {
+    name: "ClearKeyDrmUnitTest",
+    vendor: true,
+
+    cflags: ["-Wall", "-Werror"],
+
+    srcs: [
+        "AesCtrDecryptorUnittest.cpp",
+        "InitDataParserUnittest.cpp",
+        "JsonWebKeyUnittest.cpp",
+    ],
+
+    static_libs: ["libclearkeycommon"],
+
+    shared_libs: [
+        "libcrypto",
+        "libdrmclearkeyplugin",
+        "liblog",
+        "libstagefright_foundation",
+        "libutils",
+    ],
+    header_libs: ["media_plugin_headers"],
+}

diff --git a/drm/mediadrm/plugins/clearkey/tests/InitDataParserUnittest.cpp b/drm/mediadrm/plugins/clearkey/default/tests/InitDataParserUnittest.cpp
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/tests/InitDataParserUnittest.cpp
rename to drm/mediadrm/plugins/clearkey/default/tests/InitDataParserUnittest.cpp


diff --git a/drm/mediadrm/plugins/clearkey/tests/JsonWebKeyUnittest.cpp b/drm/mediadrm/plugins/clearkey/default/tests/JsonWebKeyUnittest.cpp
similarity index 100%
rename from drm/mediadrm/plugins/clearkey/tests/JsonWebKeyUnittest.cpp
rename to drm/mediadrm/plugins/clearkey/default/tests/JsonWebKeyUnittest.cpp


diff --git a/drm/mediadrm/plugins/clearkey/hidl/AesCtrDecryptor.cpp b/drm/mediadrm/plugins/clearkey/hidl/AesCtrDecryptor.cpp
new file mode 100644
index 0000000..2fce0790
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/AesCtrDecryptor.cpp

@@ -0,0 +1,86 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "hidl_ClearkeyDecryptor"
+#include <utils/Log.h>
+
+#include <openssl/aes.h>
+
+#include "AesCtrDecryptor.h"
+#include "ClearKeyTypes.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_0::SubSample;
+using ::android::hardware::drm::V1_0::Status;
+
+static const size_t kBlockBitCount = kBlockSize * 8;
+
+Status AesCtrDecryptor::decrypt(
+        const std::vector<uint8_t>& key,
+        const Iv iv, const uint8_t* source,
+        uint8_t* destination,
+        const std::vector<SubSample> subSamples,
+        size_t numSubSamples,
+        size_t* bytesDecryptedOut) {
+    uint32_t blockOffset = 0;
+    uint8_t previousEncryptedCounter[kBlockSize];
+    memset(previousEncryptedCounter, 0, kBlockSize);
+
+    if (key.size() != kBlockSize || (sizeof(Iv) / sizeof(uint8_t)) != kBlockSize) {
+        android_errorWriteLog(0x534e4554, "63982768");
+        return Status::ERROR_DRM_DECRYPT;
+    }
+
+    size_t offset = 0;
+    AES_KEY opensslKey;
+    AES_set_encrypt_key(key.data(), kBlockBitCount, &opensslKey);
+    Iv opensslIv;
+    memcpy(opensslIv, iv, sizeof(opensslIv));
+
+    for (size_t i = 0; i < numSubSamples; ++i) {
+        const SubSample& subSample = subSamples[i];
+
+        if (subSample.numBytesOfClearData > 0) {
+            memcpy(destination + offset, source + offset,
+                    subSample.numBytesOfClearData);
+            offset += subSample.numBytesOfClearData;
+        }
+
+        if (subSample.numBytesOfEncryptedData > 0) {
+            AES_ctr128_encrypt(source + offset, destination + offset,
+                    subSample.numBytesOfEncryptedData, &opensslKey,
+                    opensslIv, previousEncryptedCounter,
+                    &blockOffset);
+            offset += subSample.numBytesOfEncryptedData;
+        }
+    }
+
+    *bytesDecryptedOut = offset;
+    return Status::OK;
+}
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+

diff --git a/drm/mediadrm/plugins/clearkey/hidl/Android.bp b/drm/mediadrm/plugins/clearkey/hidl/Android.bp
new file mode 100644
index 0000000..341d4f6
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/Android.bp

@@ -0,0 +1,68 @@
+//
+// Copyright (C) 2018 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//      http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+
+cc_binary {
+    name: "android.hardware.drm@1.1-service.clearkey",
+    vendor: true,
+
+    srcs: [
+        "AesCtrDecryptor.cpp",
+        "Base64.cpp",
+        "Buffer.cpp",
+        "CreatePluginFactories.cpp",
+        "CryptoFactory.cpp",
+        "CryptoPlugin.cpp",
+        "DrmFactory.cpp",
+        "DrmPlugin.cpp",
+        "InitDataParser.cpp",
+        "JsonWebKey.cpp",
+        "Session.cpp",
+        "SessionLibrary.cpp",
+        "service.cpp",
+    ],
+
+    relative_install_path: "hw",
+
+    cflags: ["-Wall", "-Werror"],
+    init_rc: ["android.hardware.drm@1.1-service.clearkey.rc"],
+
+    shared_libs: [
+        "android.hardware.drm@1.0",
+        "android.hardware.drm@1.1",
+        "libbase",
+        "libbinder",
+        "libcrypto",
+        "libhidlbase",
+        "libhidlmemory",
+        "libhidltransport",
+        "liblog",
+        "libutils",
+    ],
+
+    static_libs: [
+        "libclearkeycommon",
+        "libjsmn",
+    ],
+
+    local_include_dirs: ["include"],
+
+    export_static_lib_headers: ["libjsmn"],
+
+    sanitize: {
+        integer_overflow: true,
+    },
+}
+

diff --git a/drm/mediadrm/plugins/clearkey/hidl/Base64.cpp b/drm/mediadrm/plugins/clearkey/hidl/Base64.cpp
new file mode 100644
index 0000000..c2ed751
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/Base64.cpp

@@ -0,0 +1,175 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "Base64.h"
+
+#include <string>
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+sp<Buffer> decodeBase64(const std::string &s) {
+    size_t n = s.size();
+
+    if ((n % 4) != 0) {
+        return nullptr;
+    }
+
+    size_t padding = 0;
+    if (n >= 1 && s.c_str()[n - 1] == '=') {
+        padding = 1;
+
+        if (n >= 2 && s.c_str()[n - 2] == '=') {
+            padding = 2;
+
+            if (n >= 3 && s.c_str()[n - 3] == '=') {
+                padding = 3;
+            }
+        }
+    }
+
+    // We divide first to avoid overflow. It's OK to do this because we
+    // already made sure that n % 4 == 0.
+    size_t outLen = (n / 4) * 3 - padding;
+
+    sp<Buffer> buffer = new Buffer(outLen);
+    uint8_t *out = buffer->data();
+    if (out == nullptr || buffer->size() < outLen) {
+        return nullptr;
+    }
+
+    size_t j = 0;
+    uint32_t accum = 0;
+    for (size_t i = 0; i < n; ++i) {
+        char c = s.c_str()[i];
+        unsigned value;
+        if (c >= 'A' && c <= 'Z') {
+            value = c - 'A';
+        } else if (c >= 'a' && c <= 'z') {
+            value = 26 + c - 'a';
+        } else if (c >= '0' && c <= '9') {
+            value = 52 + c - '0';
+        } else if (c == '+' || c == '-') {
+            value = 62;
+        } else if (c == '/' || c == '_') {
+            value = 63;
+        } else if (c != '=') {
+            return nullptr;
+        } else {
+            if (i < n - padding) {
+                return nullptr;
+            }
+
+            value = 0;
+        }
+
+        accum = (accum << 6) | value;
+
+        if (((i + 1) % 4) == 0) {
+            if (j < outLen) { out[j++] = (accum >> 16); }
+            if (j < outLen) { out[j++] = (accum >> 8) & 0xff; }
+            if (j < outLen) { out[j++] = accum & 0xff; }
+
+            accum = 0;
+        }
+    }
+
+    return buffer;
+}
+
+static char encode6Bit(unsigned x) {
+    if (x <= 25) {
+        return 'A' + x;
+    } else if (x <= 51) {
+        return 'a' + x - 26;
+    } else if (x <= 61) {
+        return '0' + x - 52;
+    } else if (x == 62) {
+        return '+';
+    } else {
+        return '/';
+    }
+}
+
+void encodeBase64(const void *_data, size_t size, std::string *out) {
+    out->clear();
+
+    const uint8_t *data = (const uint8_t *)_data;
+
+    size_t i;
+    for (i = 0; i < (size / 3) * 3; i += 3) {
+        uint8_t x1 = data[i];
+        uint8_t x2 = data[i + 1];
+        uint8_t x3 = data[i + 2];
+
+        out->push_back(encode6Bit(x1 >> 2));
+        out->push_back(encode6Bit((x1 << 4 | x2 >> 4) & 0x3f));
+        out->push_back(encode6Bit((x2 << 2 | x3 >> 6) & 0x3f));
+        out->push_back(encode6Bit(x3 & 0x3f));
+    }
+    switch (size % 3) {
+        case 0:
+            break;
+        case 2:
+        {
+            uint8_t x1 = data[i];
+            uint8_t x2 = data[i + 1];
+            out->push_back(encode6Bit(x1 >> 2));
+            out->push_back(encode6Bit((x1 << 4 | x2 >> 4) & 0x3f));
+            out->push_back(encode6Bit((x2 << 2) & 0x3f));
+            out->push_back('=');
+            break;
+        }
+        default:
+        {
+            uint8_t x1 = data[i];
+            out->push_back(encode6Bit(x1 >> 2));
+            out->push_back(encode6Bit((x1 << 4) & 0x3f));
+            out->append("==");
+            break;
+        }
+    }
+}
+
+void encodeBase64Url(const void *_data, size_t size, std::string *out) {
+    encodeBase64(_data, size, out);
+
+    if ((std::string::npos != out->find("+")) ||
+            (std::string::npos != out->find("/"))) {
+        size_t outLen = out->size();
+        char *base64url = new char[outLen];
+        for (size_t i = 0; i < outLen; ++i) {
+            if (out->c_str()[i] == '+')
+                base64url[i] = '-';
+            else if (out->c_str()[i] == '/')
+                base64url[i] = '_';
+            else
+                base64url[i] = out->c_str()[i];
+        }
+
+        out->assign(base64url, outLen);
+        delete[] base64url;
+    }
+}
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android

diff --git a/drm/mediadrm/plugins/clearkey/hidl/Buffer.cpp b/drm/mediadrm/plugins/clearkey/hidl/Buffer.cpp
new file mode 100644
index 0000000..e58f58a
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/Buffer.cpp

@@ -0,0 +1,53 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "Buffer.h"
+
+#include <android/hardware/drm/1.0/types.h>
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+Buffer::Buffer(size_t capacity)
+      : mRangeOffset(0),
+      mOwnsData(true) {
+    mData = malloc(capacity);
+    if (mData == nullptr) {
+        mCapacity = 0;
+        mRangeLength = 0;
+    } else {
+        mCapacity = capacity;
+        mRangeLength = capacity;
+    }
+}
+
+Buffer::~Buffer() {
+    if (mOwnsData) {
+        if (mData != nullptr) {
+            free(mData);
+            mData = nullptr;
+        }
+    }
+}
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android

diff --git a/drm/mediadrm/plugins/clearkey/hidl/CreatePluginFactories.cpp b/drm/mediadrm/plugins/clearkey/hidl/CreatePluginFactories.cpp
new file mode 100644
index 0000000..1ba5c6a
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/CreatePluginFactories.cpp

@@ -0,0 +1,44 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "CreatePluginFactories.h"
+
+#include "CryptoFactory.h"
+#include "DrmFactory.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+extern "C" {
+
+IDrmFactory* createDrmFactory() {
+    return new DrmFactory();
+}
+
+ICryptoFactory* createCryptoFactory() {
+    return new CryptoFactory();
+}
+
+} // extern "C"
+
+}  // namespace clearkey
+}  // namespace V1_1
+}  // namespace drm
+}  // namespace hardware
+}  // namespace android

diff --git a/drm/mediadrm/plugins/clearkey/hidl/CryptoFactory.cpp b/drm/mediadrm/plugins/clearkey/hidl/CryptoFactory.cpp
new file mode 100644
index 0000000..0848cef
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/CryptoFactory.cpp

@@ -0,0 +1,67 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "hidl_ClearKeyCryptoFactory"
+#include <utils/Log.h>
+
+#include "CryptoFactory.h"
+
+#include "ClearKeyUUID.h"
+#include "CryptoPlugin.h"
+#include "TypeConvert.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+Return<bool> CryptoFactory::isCryptoSchemeSupported(
+    const hidl_array<uint8_t, 16> &uuid)
+{
+    return clearkeydrm::isClearKeyUUID(uuid.data());
+}
+
+Return<void> CryptoFactory::createPlugin(
+    const hidl_array<uint8_t, 16> &uuid,
+    const hidl_vec<uint8_t> &initData,
+    createPlugin_cb _hidl_cb) {
+
+    if (!isCryptoSchemeSupported(uuid.data())) {
+        ALOGE("Clearkey Drm HAL: failed to create clearkey plugin, " \
+                "invalid crypto scheme");
+        _hidl_cb(Status::BAD_VALUE, nullptr);
+        return Void();
+    }
+
+    CryptoPlugin *cryptoPlugin = new CryptoPlugin(initData);
+    Status status = cryptoPlugin->getInitStatus();
+    if (status == Status::OK) {
+        _hidl_cb(Status::OK, cryptoPlugin);
+    } else {
+        delete cryptoPlugin;
+        _hidl_cb(status, nullptr);
+    }
+    return Void();
+}
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+

diff --git a/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp b/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp
new file mode 100644
index 0000000..cd2224d
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp

@@ -0,0 +1,197 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "hidl_ClearKeyCryptoPlugin"
+#include <utils/Log.h>
+
+#include "CryptoPlugin.h"
+#include "SessionLibrary.h"
+#include "TypeConvert.h"
+
+#include <hidlmemory/mapping.h>
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_0::BufferType;
+
+Return<void> CryptoPlugin::setSharedBufferBase(
+        const hidl_memory& base, uint32_t bufferId) {
+    sp<IMemory> hidlMemory = mapMemory(base);
+    ALOGE_IF(hidlMemory == nullptr, "mapMemory returns nullptr");
+
+    // allow mapMemory to return nullptr
+    mSharedBufferMap[bufferId] = hidlMemory;
+    return Void();
+}
+
+// Returns negative values for error code and positive values for the size of
+// decrypted data.  In theory, the output size can be larger than the input
+// size, but in practice this will never happen for AES-CTR.
+Return<void> CryptoPlugin::decrypt(
+        bool secure,
+        const hidl_array<uint8_t, KEY_ID_SIZE>& keyId,
+        const hidl_array<uint8_t, KEY_IV_SIZE>& iv,
+        Mode mode,
+        const Pattern& pattern,
+        const hidl_vec<SubSample>& subSamples,
+        const SharedBuffer& source,
+        uint64_t offset,
+        const DestinationBuffer& destination,
+        decrypt_cb _hidl_cb) {
+    UNUSED(pattern);
+
+    if (secure) {
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, 0,
+            "Secure decryption is not supported with ClearKey.");
+        return Void();
+    }
+
+    if (mSharedBufferMap.find(source.bufferId) == mSharedBufferMap.end()) {
+      _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, 0,
+               "source decrypt buffer base not set");
+      return Void();
+    }
+
+    if (destination.type == BufferType::SHARED_MEMORY) {
+      const SharedBuffer& dest = destination.nonsecureMemory;
+      if (mSharedBufferMap.find(dest.bufferId) == mSharedBufferMap.end()) {
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, 0,
+                 "destination decrypt buffer base not set");
+        return Void();
+      }
+    } else {
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, 0,
+                 "destination type not supported");
+        return Void();
+    }
+
+    sp<IMemory> sourceBase = mSharedBufferMap[source.bufferId];
+    if (sourceBase == nullptr) {
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, 0, "source is a nullptr");
+        return Void();
+    }
+
+    if (source.offset + offset + source.size > sourceBase->getSize()) {
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, 0, "invalid buffer size");
+        return Void();
+    }
+
+    uint8_t *base = static_cast<uint8_t *>
+            (static_cast<void *>(sourceBase->getPointer()));
+    uint8_t* srcPtr = static_cast<uint8_t *>(base + source.offset + offset);
+    void* destPtr = NULL;
+    // destination.type == BufferType::SHARED_MEMORY
+    const SharedBuffer& destBuffer = destination.nonsecureMemory;
+    sp<IMemory> destBase = mSharedBufferMap[destBuffer.bufferId];
+    if (destBase == nullptr) {
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, 0, "destination is a nullptr");
+        return Void();
+    }
+
+    base = static_cast<uint8_t *>(static_cast<void *>(destBase->getPointer()));
+
+    if (destBuffer.offset + destBuffer.size > destBase->getSize()) {
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, 0, "invalid buffer size");
+        return Void();
+    }
+    destPtr = static_cast<void *>(base + destination.nonsecureMemory.offset);
+
+    // Calculate the output buffer size and determine if any subsamples are
+    // encrypted.
+    size_t destSize = 0;
+    bool haveEncryptedSubsamples = false;
+    for (size_t i = 0; i < subSamples.size(); i++) {
+        const SubSample &subSample = subSamples[i];
+        if (__builtin_add_overflow(destSize, subSample.numBytesOfClearData, &destSize)) {
+            _hidl_cb(Status::BAD_VALUE, 0, "subsample clear size overflow");
+            return Void();
+        }
+        if (__builtin_add_overflow(destSize, subSample.numBytesOfEncryptedData, &destSize)) {
+            _hidl_cb(Status::BAD_VALUE, 0, "subsample encrypted size overflow");
+            return Void();
+        }
+        if (subSample.numBytesOfEncryptedData > 0) {
+        haveEncryptedSubsamples = true;
+        }
+    }
+
+    if (destSize > destBuffer.size) {
+        _hidl_cb(Status::BAD_VALUE, 0, "subsample sum too large");
+        return Void();
+    }
+
+    if (mode == Mode::UNENCRYPTED) {
+        if (haveEncryptedSubsamples) {
+            _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, 0,
+                    "Encrypted subsamples found in allegedly unencrypted data.");
+            return Void();
+        }
+
+        size_t offset = 0;
+        for (size_t i = 0; i < subSamples.size(); ++i) {
+            const SubSample& subSample = subSamples[i];
+            if (subSample.numBytesOfClearData != 0) {
+                memcpy(reinterpret_cast<uint8_t*>(destPtr) + offset,
+                       reinterpret_cast<const uint8_t*>(srcPtr) + offset,
+                       subSample.numBytesOfClearData);
+                offset += subSample.numBytesOfClearData;
+            }
+        }
+
+        _hidl_cb(Status::OK, static_cast<ssize_t>(offset), "");
+        return Void();
+    } else if (mode == Mode::AES_CTR) {
+        size_t bytesDecrypted;
+        Status res = mSession->decrypt(keyId.data(), iv.data(), srcPtr,
+                static_cast<uint8_t*>(destPtr), toVector(subSamples), &bytesDecrypted);
+        if (res == Status::OK) {
+            _hidl_cb(Status::OK, static_cast<ssize_t>(bytesDecrypted), "");
+            return Void();
+        } else {
+            _hidl_cb(Status::ERROR_DRM_DECRYPT, static_cast<ssize_t>(res),
+                    "Decryption Error");
+            return Void();
+        }
+    } else {
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, 0,
+                "Selected encryption mode is not supported by the ClearKey DRM Plugin.");
+        return Void();
+    }
+}
+
+Return<Status> CryptoPlugin::setMediaDrmSession(
+        const hidl_vec<uint8_t>& sessionId) {
+    if (!sessionId.size()) {
+        mSession = nullptr;
+    } else {
+        mSession = SessionLibrary::get()->findSession(sessionId);
+        if (!mSession.get()) {
+            return Status::ERROR_DRM_SESSION_NOT_OPENED;
+        }
+    }
+    return Status::OK;
+}
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android

diff --git a/drm/mediadrm/plugins/clearkey/hidl/DrmFactory.cpp b/drm/mediadrm/plugins/clearkey/hidl/DrmFactory.cpp
new file mode 100644
index 0000000..77557f9
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/DrmFactory.cpp

@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "hidl_ClearKeyDrmFactory"
+#include <utils/Log.h>
+
+#include <utils/Errors.h>
+
+#include "DrmFactory.h"
+
+#include "DrmPlugin.h"
+#include "ClearKeyUUID.h"
+#include "MimeType.h"
+#include "SessionLibrary.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_0::Status;
+using ::android::hardware::Void;
+
+Return<bool> DrmFactory::isCryptoSchemeSupported(
+        const hidl_array<uint8_t, 16>& uuid) {
+    return clearkeydrm::isClearKeyUUID(uuid.data());
+}
+
+Return<bool> DrmFactory::isContentTypeSupported(const hidl_string &mimeType) {
+    // This should match the mimeTypes handed by InitDataParser.
+    return mimeType == kIsoBmffVideoMimeType ||
+            mimeType == kIsoBmffAudioMimeType ||
+            mimeType == kCencInitDataFormat ||
+            mimeType == kWebmVideoMimeType ||
+            mimeType == kWebmAudioMimeType ||
+            mimeType == kWebmInitDataFormat;
+}
+
+Return<void> DrmFactory::createPlugin(
+    const hidl_array<uint8_t, 16>& uuid,
+    const hidl_string& appPackageName,
+    createPlugin_cb _hidl_cb) {
+    UNUSED(appPackageName);
+
+    DrmPlugin *plugin = NULL;
+    if (!isCryptoSchemeSupported(uuid.data())) {
+        ALOGE("Clear key Drm HAL: failed to create drm plugin, " \
+                "invalid crypto scheme");
+        _hidl_cb(Status::BAD_VALUE, plugin);
+        return Void();
+    }
+
+    plugin = new DrmPlugin(SessionLibrary::get());
+    _hidl_cb(Status::OK, plugin);
+    return Void();
+}
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android

diff --git a/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp b/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp
new file mode 100644
index 0000000..30f7459
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp

@@ -0,0 +1,578 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "hidl_ClearKeyPlugin"
+#include <utils/Log.h>
+
+#include <stdio.h>
+#include <inttypes.h>
+
+#include "DrmPlugin.h"
+#include "ClearKeyDrmProperties.h"
+#include "Session.h"
+#include "TypeConvert.h"
+
+namespace {
+const int kSecureStopIdStart = 100;
+const std::string kStreaming("Streaming");
+const std::string kOffline("Offline");
+const std::string kTrue("True");
+
+const std::string kQueryKeyLicenseType("LicenseType");
+    // Value: "Streaming" or "Offline"
+const std::string kQueryKeyPlayAllowed("PlayAllowed");
+    // Value: "True" or "False"
+const std::string kQueryKeyRenewAllowed("RenewAllowed");
+    // Value: "True" or "False"
+
+const int kSecureStopIdSize = 10;
+
+std::vector<uint8_t> uint32ToVector(uint32_t value) {
+    // 10 bytes to display max value 4294967295 + one byte null terminator
+    char buffer[kSecureStopIdSize];
+    memset(buffer, 0, kSecureStopIdSize);
+    snprintf(buffer, kSecureStopIdSize, "%" PRIu32, value);
+    return std::vector<uint8_t>(buffer, buffer + sizeof(buffer));
+}
+
+}; // unnamed namespace
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+DrmPlugin::DrmPlugin(SessionLibrary* sessionLibrary)
+        : mSessionLibrary(sessionLibrary),
+          mOpenSessionOkCount(0),
+          mCloseSessionOkCount(0),
+          mCloseSessionNotOpenedCount(0),
+          mNextSecureStopId(kSecureStopIdStart) {
+    mPlayPolicy.clear();
+    initProperties();
+    mSecureStops.clear();
+}
+
+void DrmPlugin::initProperties() {
+    mStringProperties.clear();
+    mStringProperties[kVendorKey] = kVendorValue;
+    mStringProperties[kVersionKey] = kVersionValue;
+    mStringProperties[kPluginDescriptionKey] = kPluginDescriptionValue;
+    mStringProperties[kAlgorithmsKey] = kAlgorithmsValue;
+    mStringProperties[kListenerTestSupportKey] = kListenerTestSupportValue;
+
+    std::vector<uint8_t> valueVector;
+    valueVector.clear();
+    valueVector.insert(valueVector.end(),
+            kTestDeviceIdData, kTestDeviceIdData + sizeof(kTestDeviceIdData) / sizeof(uint8_t));
+    mByteArrayProperties[kDeviceIdKey] = valueVector;
+
+    valueVector.clear();
+    valueVector.insert(valueVector.end(),
+            kMetricsData, kMetricsData + sizeof(kMetricsData) / sizeof(uint8_t));
+    mByteArrayProperties[kMetricsKey] = valueVector;
+}
+
+// The secure stop in ClearKey implementation is not installed securely.
+// This function merely creates a test environment for testing secure stops APIs.
+// The content in this secure stop is implementation dependent, the clearkey
+// secureStop does not serve as a reference implementation.
+void DrmPlugin::installSecureStop(const hidl_vec<uint8_t>& sessionId) {
+    ClearkeySecureStop clearkeySecureStop;
+    clearkeySecureStop.id = uint32ToVector(++mNextSecureStopId);
+    clearkeySecureStop.data.assign(sessionId.begin(), sessionId.end());
+
+    mSecureStops.insert(std::pair<std::vector<uint8_t>, ClearkeySecureStop>(
+            clearkeySecureStop.id, clearkeySecureStop));
+}
+
+Return<void> DrmPlugin::openSession(openSession_cb _hidl_cb) {
+    sp<Session> session = mSessionLibrary->createSession();
+    std::vector<uint8_t> sessionId = session->sessionId();
+
+    Status status = setSecurityLevel(sessionId, SecurityLevel::SW_SECURE_CRYPTO);
+    _hidl_cb(status, toHidlVec(sessionId));
+    mOpenSessionOkCount++;
+    return Void();
+}
+
+Return<void> DrmPlugin::openSession_1_1(SecurityLevel securityLevel,
+        openSession_1_1_cb _hidl_cb) {
+    sp<Session> session = mSessionLibrary->createSession();
+    std::vector<uint8_t> sessionId = session->sessionId();
+
+    Status status = setSecurityLevel(sessionId, securityLevel);
+    _hidl_cb(status, toHidlVec(sessionId));
+    mOpenSessionOkCount++;
+    return Void();
+}
+
+Return<Status> DrmPlugin::closeSession(const hidl_vec<uint8_t>& sessionId) {
+    if (sessionId.size() == 0) {
+        return Status::BAD_VALUE;
+    }
+
+    sp<Session> session = mSessionLibrary->findSession(toVector(sessionId));
+    if (session.get()) {
+        mCloseSessionOkCount++;
+        mSessionLibrary->destroySession(session);
+        return Status::OK;
+    }
+    mCloseSessionNotOpenedCount++;
+    return Status::ERROR_DRM_SESSION_NOT_OPENED;
+}
+
+Status DrmPlugin::getKeyRequestCommon(const hidl_vec<uint8_t>& scope,
+        const hidl_vec<uint8_t>& initData,
+        const hidl_string& mimeType,
+        KeyType keyType,
+        const hidl_vec<KeyValue>& optionalParameters,
+        std::vector<uint8_t> *request,
+        KeyRequestType *keyRequestType,
+        std::string *defaultUrl) {
+        UNUSED(optionalParameters);
+
+    *defaultUrl = "";
+    *keyRequestType = KeyRequestType::UNKNOWN;
+    *request = std::vector<uint8_t>();
+
+    if (scope.size() == 0) {
+        return Status::BAD_VALUE;
+    }
+
+    if (keyType != KeyType::STREAMING) {
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    sp<Session> session = mSessionLibrary->findSession(toVector(scope));
+    if (!session.get()) {
+        return Status::ERROR_DRM_SESSION_NOT_OPENED;
+    }
+
+    Status status = session->getKeyRequest(initData, mimeType, request);
+    *keyRequestType = KeyRequestType::INITIAL;
+    return status;
+}
+
+Return<void> DrmPlugin::getKeyRequest(
+        const hidl_vec<uint8_t>& scope,
+        const hidl_vec<uint8_t>& initData,
+        const hidl_string& mimeType,
+        KeyType keyType,
+        const hidl_vec<KeyValue>& optionalParameters,
+        getKeyRequest_cb _hidl_cb) {
+    UNUSED(optionalParameters);
+
+    KeyRequestType keyRequestType = KeyRequestType::UNKNOWN;
+    std::string defaultUrl("");
+    std::vector<uint8_t> request;
+    Status status = getKeyRequestCommon(
+            scope, initData, mimeType, keyType, optionalParameters,
+            &request, &keyRequestType, &defaultUrl);
+
+    _hidl_cb(status, toHidlVec(request),
+            static_cast<drm::V1_0::KeyRequestType>(keyRequestType),
+            hidl_string(defaultUrl));
+    return Void();
+}
+
+Return<void> DrmPlugin::getKeyRequest_1_1(
+        const hidl_vec<uint8_t>& scope,
+        const hidl_vec<uint8_t>& initData,
+        const hidl_string& mimeType,
+        KeyType keyType,
+        const hidl_vec<KeyValue>& optionalParameters,
+        getKeyRequest_1_1_cb _hidl_cb) {
+    UNUSED(optionalParameters);
+
+    KeyRequestType keyRequestType = KeyRequestType::UNKNOWN;
+    std::string defaultUrl("");
+    std::vector<uint8_t> request;
+    Status status = getKeyRequestCommon(
+            scope, initData, mimeType, keyType, optionalParameters,
+            &request, &keyRequestType, &defaultUrl);
+
+    _hidl_cb(status, toHidlVec(request), keyRequestType, hidl_string(defaultUrl));
+    return Void();
+}
+
+void DrmPlugin::setPlayPolicy() {
+    mPlayPolicy.clear();
+
+    KeyValue policy;
+    policy.key = kQueryKeyLicenseType;
+    policy.value = kStreaming;
+    mPlayPolicy.push_back(policy);
+
+    policy.key = kQueryKeyPlayAllowed;
+    policy.value = kTrue;
+    mPlayPolicy.push_back(policy);
+
+    policy.key = kQueryKeyRenewAllowed;
+    mPlayPolicy.push_back(policy);
+}
+
+Return<void> DrmPlugin::provideKeyResponse(
+        const hidl_vec<uint8_t>& scope,
+        const hidl_vec<uint8_t>& response,
+        provideKeyResponse_cb _hidl_cb) {
+    if (scope.size() == 0 || response.size() == 0) {
+        // Returns empty keySetId
+        _hidl_cb(Status::BAD_VALUE, hidl_vec<uint8_t>());
+        return Void();
+    }
+
+    sp<Session> session = mSessionLibrary->findSession(toVector(scope));
+    if (!session.get()) {
+        _hidl_cb(Status::ERROR_DRM_SESSION_NOT_OPENED, hidl_vec<uint8_t>());
+        return Void();
+    }
+
+    setPlayPolicy();
+    std::vector<uint8_t> keySetId;
+    Status status = session->provideKeyResponse(response);
+    if (status == Status::OK) {
+        // This is for testing AMediaDrm_setOnEventListener only.
+        sendEvent(EventType::VENDOR_DEFINED, 0, scope);
+        keySetId.clear();
+    }
+
+    installSecureStop(scope);
+
+    // Returns status and empty keySetId
+    _hidl_cb(status, toHidlVec(keySetId));
+    return Void();
+}
+
+Return<void> DrmPlugin::getPropertyString(
+        const hidl_string& propertyName, getPropertyString_cb _hidl_cb) {
+    std::string name(propertyName.c_str());
+    std::string value;
+
+    if (name == kVendorKey) {
+        value = mStringProperties[kVendorKey];
+    } else if (name == kVersionKey) {
+        value = mStringProperties[kVersionKey];
+    } else if (name == kPluginDescriptionKey) {
+        value = mStringProperties[kPluginDescriptionKey];
+    } else if (name == kAlgorithmsKey) {
+        value = mStringProperties[kAlgorithmsKey];
+    } else if (name == kListenerTestSupportKey) {
+        value = mStringProperties[kListenerTestSupportKey];
+    } else {
+        ALOGE("App requested unknown string property %s", name.c_str());
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, "");
+        return Void();
+    }
+    _hidl_cb(Status::OK, value.c_str());
+    return Void();
+}
+
+Return<void> DrmPlugin::getPropertyByteArray(
+        const hidl_string& propertyName, getPropertyByteArray_cb _hidl_cb) {
+    std::map<std::string, std::vector<uint8_t> >::iterator itr =
+            mByteArrayProperties.find(std::string(propertyName.c_str()));
+    if (itr == mByteArrayProperties.end()) {
+        ALOGE("App requested unknown property: %s", propertyName.c_str());
+        _hidl_cb(Status::BAD_VALUE, std::vector<uint8_t>());
+        return Void();
+    }
+    _hidl_cb(Status::OK, itr->second);
+    return Void();
+
+}
+
+Return<Status> DrmPlugin::setPropertyString(
+    const hidl_string& name, const hidl_string& value) {
+    std::string immutableKeys;
+    immutableKeys.append(kAlgorithmsKey + ",");
+    immutableKeys.append(kPluginDescriptionKey + ",");
+    immutableKeys.append(kVendorKey + ",");
+    immutableKeys.append(kVersionKey + ",");
+
+    std::string key = std::string(name.c_str());
+    if (immutableKeys.find(key) != std::string::npos) {
+        ALOGD("Cannot set immutable property: %s", key.c_str());
+        return Status::BAD_VALUE;
+    }
+
+    std::map<std::string, std::string>::iterator itr =
+            mStringProperties.find(key);
+    if (itr == mStringProperties.end()) {
+        ALOGE("Cannot set undefined property string, key=%s", key.c_str());
+        return Status::BAD_VALUE;
+    }
+
+    mStringProperties[key] = std::string(value.c_str());
+    return Status::OK;
+}
+
+Return<Status> DrmPlugin::setPropertyByteArray(
+    const hidl_string& name, const hidl_vec<uint8_t>& value) {
+   UNUSED(value);
+   if (name == kDeviceIdKey) {
+      ALOGD("Cannot set immutable property: %s", name.c_str());
+      return Status::BAD_VALUE;
+   }
+
+   // Setting of undefined properties is not supported
+   ALOGE("Failed to set property byte array, key=%s", name.c_str());
+   return Status::ERROR_DRM_CANNOT_HANDLE;
+}
+
+Return<void> DrmPlugin::queryKeyStatus(
+        const hidl_vec<uint8_t>& sessionId,
+        queryKeyStatus_cb _hidl_cb) {
+
+    if (sessionId.size() == 0) {
+        // Returns empty key status KeyValue pair
+        _hidl_cb(Status::BAD_VALUE, hidl_vec<KeyValue>());
+        return Void();
+    }
+
+    std::vector<KeyValue> infoMapVec;
+    infoMapVec.clear();
+
+    KeyValue keyValuePair;
+    for (size_t i = 0; i < mPlayPolicy.size(); ++i) {
+        keyValuePair.key = mPlayPolicy[i].key;
+        keyValuePair.value = mPlayPolicy[i].value;
+        infoMapVec.push_back(keyValuePair);
+    }
+    _hidl_cb(Status::OK, toHidlVec(infoMapVec));
+    return Void();
+}
+
+Return<void> DrmPlugin::getNumberOfSessions(getNumberOfSessions_cb _hidl_cb) {
+        uint32_t currentSessions = mSessionLibrary->numOpenSessions();
+        uint32_t maxSessions = 10;
+        _hidl_cb(Status::OK, currentSessions, maxSessions);
+        return Void();
+}
+
+Return<void> DrmPlugin::getSecurityLevel(const hidl_vec<uint8_t>& sessionId,
+            getSecurityLevel_cb _hidl_cb) {
+    if (sessionId.size() == 0) {
+        _hidl_cb(Status::BAD_VALUE, SecurityLevel::UNKNOWN);
+        return Void();
+    }
+
+    std::vector<uint8_t> sid = toVector(sessionId);
+    sp<Session> session = mSessionLibrary->findSession(sid);
+    if (!session.get()) {
+        _hidl_cb(Status::ERROR_DRM_SESSION_NOT_OPENED, SecurityLevel::UNKNOWN);
+        return Void();
+    }
+
+    std::map<std::vector<uint8_t>, SecurityLevel>::iterator itr =
+            mSecurityLevel.find(sid);
+    if (itr == mSecurityLevel.end()) {
+        ALOGE("Session id not found");
+        _hidl_cb(Status::ERROR_DRM_INVALID_STATE, SecurityLevel::UNKNOWN);
+        return Void();
+    }
+
+    _hidl_cb(Status::OK, itr->second);
+    return Void();
+}
+
+Return<Status> DrmPlugin::setSecurityLevel(const hidl_vec<uint8_t>& sessionId,
+            SecurityLevel level) {
+    if (sessionId.size() == 0) {
+        ALOGE("Invalid empty session id");
+        return Status::BAD_VALUE;
+    }
+
+    if (level > SecurityLevel::SW_SECURE_CRYPTO) {
+        ALOGE("Cannot set security level > max");
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    std::vector<uint8_t> sid = toVector(sessionId);
+    sp<Session> session = mSessionLibrary->findSession(sid);
+    if (!session.get()) {
+        return Status::ERROR_DRM_SESSION_NOT_OPENED;
+    }
+
+    std::map<std::vector<uint8_t>, SecurityLevel>::iterator itr =
+            mSecurityLevel.find(sid);
+    if (itr != mSecurityLevel.end()) {
+        mSecurityLevel[sid] = level;
+    } else {
+        if (!mSecurityLevel.insert(
+                std::pair<std::vector<uint8_t>, SecurityLevel>(sid, level)).second) {
+            ALOGE("Failed to set security level");
+            return Status::ERROR_DRM_INVALID_STATE;
+        }
+    }
+    return Status::OK;
+}
+
+Return<void> DrmPlugin::getMetrics(getMetrics_cb _hidl_cb) {
+    // Set the open session count metric.
+    DrmMetricGroup::Attribute openSessionOkAttribute = {
+      "status", DrmMetricGroup::ValueType::INT64_TYPE, (int64_t) Status::OK, 0.0, ""
+    };
+    DrmMetricGroup::Value openSessionMetricValue = {
+      "count", DrmMetricGroup::ValueType::INT64_TYPE, mOpenSessionOkCount, 0.0, ""
+    };
+    DrmMetricGroup::Metric openSessionMetric = {
+      "open_session", { openSessionOkAttribute }, { openSessionMetricValue }
+    };
+
+    // Set the close session count metric.
+    DrmMetricGroup::Attribute closeSessionOkAttribute = {
+      "status", DrmMetricGroup::ValueType::INT64_TYPE, (int64_t) Status::OK, 0.0, ""
+    };
+    DrmMetricGroup::Value closeSessionMetricValue = {
+      "count", DrmMetricGroup::ValueType::INT64_TYPE, mCloseSessionOkCount, 0.0, ""
+    };
+    DrmMetricGroup::Metric closeSessionMetric = {
+      "close_session", { closeSessionOkAttribute }, { closeSessionMetricValue }
+    };
+
+    // Set the close session, not opened metric.
+    DrmMetricGroup::Attribute closeSessionNotOpenedAttribute = {
+      "status", DrmMetricGroup::ValueType::INT64_TYPE,
+      (int64_t) Status::ERROR_DRM_SESSION_NOT_OPENED, 0.0, ""
+    };
+    DrmMetricGroup::Value closeSessionNotOpenedMetricValue = {
+      "count", DrmMetricGroup::ValueType::INT64_TYPE, mCloseSessionNotOpenedCount, 0.0, ""
+    };
+    DrmMetricGroup::Metric closeSessionNotOpenedMetric = {
+      "close_session", { closeSessionNotOpenedAttribute }, { closeSessionNotOpenedMetricValue }
+    };
+
+    DrmMetricGroup metrics = { { openSessionMetric, closeSessionMetric,
+                                closeSessionNotOpenedMetric } };
+
+    _hidl_cb(Status::OK, hidl_vec<DrmMetricGroup>({metrics}));
+    return Void();
+}
+
+Return<void> DrmPlugin::getSecureStops(getSecureStops_cb _hidl_cb) {
+    std::vector<SecureStop> stops;
+    for (auto itr = mSecureStops.begin(); itr != mSecureStops.end(); ++itr) {
+        ClearkeySecureStop clearkeyStop = itr->second;
+        std::vector<uint8_t> stopVec;
+        stopVec.insert(stopVec.end(), clearkeyStop.id.begin(), clearkeyStop.id.end());
+        stopVec.insert(stopVec.end(), clearkeyStop.data.begin(), clearkeyStop.data.end());
+
+        SecureStop stop;
+        stop.opaqueData = toHidlVec(stopVec);
+        stops.push_back(stop);
+    }
+    _hidl_cb(Status::OK, stops);
+    return Void();
+}
+
+Return<void> DrmPlugin::getSecureStop(const hidl_vec<uint8_t>& secureStopId,
+        getSecureStop_cb _hidl_cb) {
+    SecureStop stop;
+    auto itr = mSecureStops.find(toVector(secureStopId));
+    if (itr != mSecureStops.end()) {
+        ClearkeySecureStop clearkeyStop = itr->second;
+        std::vector<uint8_t> stopVec;
+        stopVec.insert(stopVec.end(), clearkeyStop.id.begin(), clearkeyStop.id.end());
+        stopVec.insert(stopVec.end(), clearkeyStop.data.begin(), clearkeyStop.data.end());
+
+        stop.opaqueData = toHidlVec(stopVec);
+        _hidl_cb(Status::OK, stop);
+    } else {
+        _hidl_cb(Status::BAD_VALUE, stop);
+    }
+
+    return Void();
+}
+
+Return<Status> DrmPlugin::releaseSecureStop(const hidl_vec<uint8_t>& secureStopId) {
+    return removeSecureStop(secureStopId);
+}
+
+Return<Status> DrmPlugin::releaseAllSecureStops() {
+    return removeAllSecureStops();
+}
+
+Return<void> DrmPlugin::getSecureStopIds(getSecureStopIds_cb _hidl_cb) {
+    std::vector<SecureStopId> ids;
+    for (auto itr = mSecureStops.begin(); itr != mSecureStops.end(); ++itr) {
+        ids.push_back(itr->first);
+    }
+
+    _hidl_cb(Status::OK, toHidlVec(ids));
+    return Void();
+}
+
+Return<Status> DrmPlugin::releaseSecureStops(const SecureStopRelease& ssRelease) {
+    if (ssRelease.opaqueData.size() == 0) {
+        return Status::BAD_VALUE;
+    }
+
+    Status status = Status::OK;
+    std::vector<uint8_t> input = toVector(ssRelease.opaqueData);
+
+    // The format of opaqueData is shared between the server
+    // and the drm service. The clearkey implementation consists of:
+    //    count - number of secure stops
+    //    list of fixed length secure stops
+    size_t countBufferSize = sizeof(uint32_t);
+    if (input.size() < countBufferSize) {
+        // SafetyNet logging
+        android_errorWriteLog(0x534e4554, "144766455");
+        return Status::BAD_VALUE;
+    }
+    uint32_t count = 0;
+    sscanf(reinterpret_cast<char*>(input.data()), "%04" PRIu32, &count);
+
+    // Avoid divide by 0 below.
+    if (count == 0) {
+        return Status::BAD_VALUE;
+    }
+
+    size_t secureStopSize = (input.size() - countBufferSize) / count;
+    uint8_t buffer[secureStopSize];
+    size_t offset = countBufferSize; // skip the count
+    for (size_t i = 0; i < count; ++i, offset += secureStopSize) {
+        memcpy(buffer, input.data() + offset, secureStopSize);
+        std::vector<uint8_t> id(buffer, buffer + kSecureStopIdSize);
+
+        status = removeSecureStop(toHidlVec(id));
+        if (Status::OK != status) break;
+    }
+
+    return status;
+}
+
+Return<Status> DrmPlugin::removeSecureStop(const hidl_vec<uint8_t>& secureStopId) {
+    if (1 != mSecureStops.erase(toVector(secureStopId))) {
+        return Status::BAD_VALUE;
+    }
+    return Status::OK;
+}
+
+Return<Status> DrmPlugin::removeAllSecureStops() {
+    mSecureStops.clear();
+    mNextSecureStopId = kSecureStopIdStart;
+    return Status::OK;
+}
+
+}  // namespace clearkey
+}  // namespace V1_1
+}  // namespace drm
+}  // namespace hardware
+}  // namespace android

diff --git a/drm/mediadrm/plugins/clearkey/hidl/InitDataParser.cpp b/drm/mediadrm/plugins/clearkey/hidl/InitDataParser.cpp
new file mode 100644
index 0000000..e2bb651
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/InitDataParser.cpp

@@ -0,0 +1,163 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "hidl_InitDataParser"
+
+#include <algorithm>
+#include <utils/Log.h>
+
+#include "InitDataParser.h"
+
+#include "Base64.h"
+
+#include "ClearKeyUUID.h"
+#include "MimeType.h"
+#include "Utils.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+namespace {
+    const size_t kKeyIdSize = 16;
+    const size_t kSystemIdSize = 16;
+}
+
+std::vector<uint8_t> StrToVector(const std::string& str) {
+    std::vector<uint8_t> vec(str.begin(), str.end());
+    return vec;
+}
+
+Status InitDataParser::parse(const std::vector<uint8_t>& initData,
+        const std::string& type,
+        std::vector<uint8_t>* licenseRequest) {
+    // Build a list of the key IDs
+    std::vector<const uint8_t*> keyIds;
+
+    if (type == kIsoBmffVideoMimeType ||
+        type == kIsoBmffAudioMimeType ||
+        type == kCencInitDataFormat) {
+        Status res = parsePssh(initData, &keyIds);
+        if (res != Status::OK) {
+            return res;
+        }
+    } else if (type == kWebmVideoMimeType ||
+        type == kWebmAudioMimeType ||
+        type == kWebmInitDataFormat) {
+        // WebM "init data" is just a single key ID
+        if (initData.size() != kKeyIdSize) {
+            return Status::ERROR_DRM_CANNOT_HANDLE;
+        }
+        keyIds.push_back(initData.data());
+    } else {
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    // Build the request
+    std::string requestJson = generateRequest(keyIds);
+    std::vector<uint8_t> requestJsonVec = StrToVector(requestJson);
+
+    licenseRequest->clear();
+    licenseRequest->insert(licenseRequest->end(), requestJsonVec.begin(), requestJsonVec.end());
+    return Status::OK;
+}
+
+Status InitDataParser::parsePssh(const std::vector<uint8_t>& initData,
+        std::vector<const uint8_t*>* keyIds) {
+    size_t readPosition = 0;
+
+    // Validate size field
+    uint32_t expectedSize = initData.size();
+    expectedSize = htonl(expectedSize);
+    if (memcmp(&initData[readPosition], &expectedSize,
+               sizeof(expectedSize)) != 0) {
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+    readPosition += sizeof(expectedSize);
+
+    // Validate PSSH box identifier
+    const char psshIdentifier[4] = {'p', 's', 's', 'h'};
+    if (memcmp(&initData[readPosition], psshIdentifier,
+               sizeof(psshIdentifier)) != 0) {
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+    readPosition += sizeof(psshIdentifier);
+
+    // Validate EME version number
+    const uint8_t psshVersion1[4] = {1, 0, 0, 0};
+    if (memcmp(&initData[readPosition], psshVersion1,
+               sizeof(psshVersion1)) != 0) {
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+    readPosition += sizeof(psshVersion1);
+
+    // Validate system ID
+    if (!clearkeydrm::isClearKeyUUID(&initData[readPosition])) {
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+    readPosition += kSystemIdSize;
+
+    // Read key ID count
+    uint32_t keyIdCount;
+    memcpy(&keyIdCount, &initData[readPosition], sizeof(keyIdCount));
+    keyIdCount = ntohl(keyIdCount);
+    readPosition += sizeof(keyIdCount);
+    if (readPosition + ((uint64_t)keyIdCount * kKeyIdSize) !=
+            initData.size() - sizeof(uint32_t)) {
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    // Calculate the key ID offsets
+    for (uint32_t i = 0; i < keyIdCount; ++i) {
+        size_t keyIdPosition = readPosition + (i * kKeyIdSize);
+        keyIds->push_back(&initData[keyIdPosition]);
+    }
+    return Status::OK;
+}
+
+std::string InitDataParser::generateRequest(const std::vector<const uint8_t*>& keyIds) {
+    const std::string kRequestPrefix("{\"kids\":[");
+    const std::string kRequestSuffix("],\"type\":\"temporary\"}");
+
+    std::string request(kRequestPrefix);
+    std::string encodedId;
+    for (size_t i = 0; i < keyIds.size(); ++i) {
+        encodedId.clear();
+        encodeBase64Url(keyIds[i], kKeyIdSize, &encodedId);
+        if (i != 0) {
+            request.append(",");
+        }
+        request.push_back('\"');
+        request.append(encodedId);
+        request.push_back('\"');
+    }
+    request.append(kRequestSuffix);
+
+    // Android's Base64 encoder produces padding. EME forbids padding.
+    const char kBase64Padding = '=';
+    request.erase(std::remove(request.begin(), request.end(), kBase64Padding), request.end());
+
+    return request;
+}
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android

diff --git a/drm/mediadrm/plugins/clearkey/hidl/JsonWebKey.cpp b/drm/mediadrm/plugins/clearkey/hidl/JsonWebKey.cpp
new file mode 100644
index 0000000..cccb41e
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/JsonWebKey.cpp

@@ -0,0 +1,275 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#define LOG_TAG "hidl_JsonWebKey"
+
+#include <utils/Log.h>
+
+#include "JsonWebKey.h"
+
+#include "Base64.h"
+
+namespace {
+const std::string kKeysTag("keys");
+const std::string kKeyTypeTag("kty");
+const std::string kSymmetricKeyValue("oct");
+const std::string kKeyTag("k");
+const std::string kKeyIdTag("kid");
+const std::string kBase64Padding("=");
+}
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+JsonWebKey::JsonWebKey() {
+}
+
+JsonWebKey::~JsonWebKey() {
+}
+
+/*
+ * Parses a JSON Web Key Set string, initializes a KeyMap with key id:key
+ * pairs from the JSON Web Key Set. Both key ids and keys are base64url
+ * encoded. The KeyMap contains base64url decoded key id:key pairs.
+ *
+ * @return Returns false for errors, true for success.
+ */
+bool JsonWebKey::extractKeysFromJsonWebKeySet(const std::string& jsonWebKeySet,
+        KeyMap* keys) {
+
+    keys->clear();
+
+    if (!parseJsonWebKeySet(jsonWebKeySet, &mJsonObjects)) {
+        return false;
+    }
+
+    // mJsonObjects[0] contains the entire JSON Web Key Set, including
+    // all the base64 encoded keys. Each key is also stored separately as
+    // a JSON object in mJsonObjects[1..n] where n is the total
+    // number of keys in the set.
+    if (!isJsonWebKeySet(mJsonObjects[0])) {
+        return false;
+    }
+
+    std::string encodedKey, encodedKeyId;
+    std::vector<uint8_t> decodedKey, decodedKeyId;
+
+    // mJsonObjects[1] contains the first JSON Web Key in the set
+    for (size_t i = 1; i < mJsonObjects.size(); ++i) {
+        encodedKeyId.clear();
+        encodedKey.clear();
+
+        if (!parseJsonObject(mJsonObjects[i], &mTokens))
+            return false;
+
+        if (findKey(mJsonObjects[i], &encodedKeyId, &encodedKey)) {
+            if (encodedKeyId.empty() || encodedKey.empty()) {
+                ALOGE("Must have both key id and key in the JsonWebKey set.");
+                continue;
+            }
+
+            if (!decodeBase64String(encodedKeyId, &decodedKeyId)) {
+                ALOGE("Failed to decode key id(%s)", encodedKeyId.c_str());
+                continue;
+            }
+
+            if (!decodeBase64String(encodedKey, &decodedKey)) {
+                ALOGE("Failed to decode key(%s)", encodedKey.c_str());
+                continue;
+            }
+
+            keys->insert(std::pair<std::vector<uint8_t>,
+                    std::vector<uint8_t> >(decodedKeyId, decodedKey));
+        }
+    }
+    return true;
+}
+
+bool JsonWebKey::decodeBase64String(const std::string& encodedText,
+        std::vector<uint8_t>* decodedText) {
+
+    decodedText->clear();
+
+    // encodedText should not contain padding characters as per EME spec.
+    if (encodedText.find(kBase64Padding) != std::string::npos) {
+        return false;
+    }
+
+    // Since decodeBase64() requires padding characters,
+    // add them so length of encodedText is exactly a multiple of 4.
+    int remainder = encodedText.length() % 4;
+    std::string paddedText(encodedText);
+    if (remainder > 0) {
+        for (int i = 0; i < 4 - remainder; ++i) {
+            paddedText.append(kBase64Padding);
+        }
+    }
+
+    sp<Buffer> buffer = decodeBase64(paddedText);
+    if (buffer == nullptr) {
+        ALOGE("Malformed base64 encoded content found.");
+        return false;
+    }
+
+    decodedText->insert(decodedText->end(), buffer->base(), buffer->base() + buffer->size());
+    return true;
+}
+
+bool JsonWebKey::findKey(const std::string& jsonObject, std::string* keyId,
+        std::string* encodedKey) {
+
+    std::string key, value;
+
+    // Only allow symmetric key, i.e. "kty":"oct" pair.
+    if (jsonObject.find(kKeyTypeTag) != std::string::npos) {
+        findValue(kKeyTypeTag, &value);
+        if (0 != value.compare(kSymmetricKeyValue))
+            return false;
+    }
+
+    if (jsonObject.find(kKeyIdTag) != std::string::npos) {
+        findValue(kKeyIdTag, keyId);
+    }
+
+    if (jsonObject.find(kKeyTag) != std::string::npos) {
+        findValue(kKeyTag, encodedKey);
+    }
+    return true;
+}
+
+void JsonWebKey::findValue(const std::string &key, std::string* value) {
+    value->clear();
+    const char* valueToken;
+    for (std::vector<std::string>::const_iterator nextToken = mTokens.begin();
+        nextToken != mTokens.end(); ++nextToken) {
+        if (0 == (*nextToken).compare(key)) {
+            if (nextToken + 1 == mTokens.end())
+                break;
+            valueToken = (*(nextToken + 1)).c_str();
+            value->assign(valueToken);
+            nextToken++;
+            break;
+        }
+    }
+}
+
+bool JsonWebKey::isJsonWebKeySet(const std::string& jsonObject) const {
+    if (jsonObject.find(kKeysTag) == std::string::npos) {
+        ALOGE("JSON Web Key does not contain keys.");
+        return false;
+    }
+    return true;
+}
+
+/*
+ * Parses a JSON objects string and initializes a vector of tokens.
+ *
+ * @return Returns false for errors, true for success.
+ */
+bool JsonWebKey::parseJsonObject(const std::string& jsonObject,
+        std::vector<std::string>* tokens) {
+    jsmn_parser parser;
+
+    jsmn_init(&parser);
+    int numTokens = jsmn_parse(&parser,
+        jsonObject.c_str(), jsonObject.size(), nullptr, 0);
+    if (numTokens < 0) {
+        ALOGE("Parser returns error code=%d", numTokens);
+        return false;
+    }
+
+    unsigned int jsmnTokensSize = numTokens * sizeof(jsmntok_t);
+    mJsmnTokens.clear();
+    mJsmnTokens.resize(jsmnTokensSize);
+
+    jsmn_init(&parser);
+    int status = jsmn_parse(&parser, jsonObject.c_str(),
+        jsonObject.size(), mJsmnTokens.data(), numTokens);
+    if (status < 0) {
+        ALOGE("Parser returns error code=%d", status);
+        return false;
+    }
+
+    tokens->clear();
+    std::string token;
+    const char *pjs;
+    for (int j = 0; j < numTokens; ++j) {
+        pjs = jsonObject.c_str() + mJsmnTokens[j].start;
+        if (mJsmnTokens[j].type == JSMN_STRING ||
+                mJsmnTokens[j].type == JSMN_PRIMITIVE) {
+            token.assign(pjs, mJsmnTokens[j].end - mJsmnTokens[j].start);
+            tokens->push_back(token);
+        }
+    }
+    return true;
+}
+
+/*
+ * Parses JSON Web Key Set string and initializes a vector of JSON objects.
+ *
+ * @return Returns false for errors, true for success.
+ */
+bool JsonWebKey::parseJsonWebKeySet(const std::string& jsonWebKeySet,
+        std::vector<std::string>* jsonObjects) {
+    if (jsonWebKeySet.empty()) {
+        ALOGE("Empty JSON Web Key");
+        return false;
+    }
+
+    // The jsmn parser only supports unicode encoding.
+    jsmn_parser parser;
+
+    // Computes number of tokens. A token marks the type, offset in
+    // the original string.
+    jsmn_init(&parser);
+    int numTokens = jsmn_parse(&parser,
+            jsonWebKeySet.c_str(), jsonWebKeySet.size(), nullptr, 0);
+    if (numTokens < 0) {
+        ALOGE("Parser returns error code=%d", numTokens);
+        return false;
+    }
+
+    unsigned int jsmnTokensSize = numTokens * sizeof(jsmntok_t);
+    mJsmnTokens.resize(jsmnTokensSize);
+
+    jsmn_init(&parser);
+    int status = jsmn_parse(&parser, jsonWebKeySet.c_str(),
+            jsonWebKeySet.size(), mJsmnTokens.data(), numTokens);
+    if (status < 0) {
+        ALOGE("Parser returns error code=%d", status);
+        return false;
+    }
+
+    std::string token;
+    const char *pjs;
+    for (int i = 0; i < numTokens; ++i) {
+        pjs = jsonWebKeySet.c_str() + mJsmnTokens[i].start;
+        if (mJsmnTokens[i].type == JSMN_OBJECT) {
+            token.assign(pjs, mJsmnTokens[i].end - mJsmnTokens[i].start);
+            jsonObjects->push_back(token);
+        }
+    }
+    return true;
+}
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+

diff --git a/drm/mediadrm/plugins/clearkey/hidl/Session.cpp b/drm/mediadrm/plugins/clearkey/hidl/Session.cpp
new file mode 100644
index 0000000..07c9269
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/Session.cpp

@@ -0,0 +1,95 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "hidl_ClearKeySession"
+#include <utils/Log.h>
+
+#include "Session.h"
+#include "Utils.h"
+
+#include "AesCtrDecryptor.h"
+#include "InitDataParser.h"
+#include "JsonWebKey.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_0::KeyValue;
+using ::android::hardware::drm::V1_0::Status;
+using ::android::hardware::drm::V1_0::SubSample;
+using ::android::hardware::Return;
+using ::android::sp;
+
+using android::Mutex;
+
+Status Session::getKeyRequest(
+        const std::vector<uint8_t>& initData,
+        const std::string& mimeType,
+        std::vector<uint8_t>* keyRequest) const {
+    InitDataParser parser;
+    return parser.parse(initData, mimeType, keyRequest);
+}
+
+Status Session::provideKeyResponse(const std::vector<uint8_t>& response) {
+    std::string responseString(
+            reinterpret_cast<const char*>(response.data()), response.size());
+    KeyMap keys;
+
+    Mutex::Autolock lock(mMapLock);
+    JsonWebKey parser;
+    if (parser.extractKeysFromJsonWebKeySet(responseString, &keys)) {
+        for (auto &key : keys) {
+            std::string first(key.first.begin(), key.first.end());
+            std::string second(key.second.begin(), key.second.end());
+            mKeyMap.insert(std::pair<std::vector<uint8_t>,
+                    std::vector<uint8_t> >(key.first, key.second));
+        }
+        return Status::OK;
+    } else {
+        return Status::ERROR_DRM_UNKNOWN;
+    }
+}
+
+Status Session::decrypt(
+        const KeyId keyId, const Iv iv, const uint8_t* srcPtr,
+        uint8_t* destPtr, const std::vector<SubSample> subSamples,
+        size_t* bytesDecryptedOut) {
+    Mutex::Autolock lock(mMapLock);
+
+    std::vector<uint8_t> keyIdVector;
+    keyIdVector.clear();
+    keyIdVector.insert(keyIdVector.end(), keyId, keyId + kBlockSize);
+    std::map<std::vector<uint8_t>, std::vector<uint8_t> >::iterator itr;
+    itr = mKeyMap.find(keyIdVector);
+    if (itr == mKeyMap.end()) {
+        return Status::ERROR_DRM_NO_LICENSE;
+    }
+
+    AesCtrDecryptor decryptor;
+    return decryptor.decrypt(
+            itr->second /*key*/, iv, srcPtr, destPtr, subSamples,
+            subSamples.size(), bytesDecryptedOut);
+}
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android

diff --git a/drm/mediadrm/plugins/clearkey/hidl/SessionLibrary.cpp b/drm/mediadrm/plugins/clearkey/hidl/SessionLibrary.cpp
new file mode 100644
index 0000000..b4319e6
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/SessionLibrary.cpp

@@ -0,0 +1,90 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "hidl_ClearKeySessionLibrary"
+#include <utils/Log.h>
+
+#include "SessionLibrary.h"
+#include "Utils.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::hidl_string;
+using ::android::hardware::hidl_vec;
+using ::android::sp;
+
+Mutex SessionLibrary::sSingletonLock;
+SessionLibrary* SessionLibrary::sSingleton = NULL;
+
+SessionLibrary* SessionLibrary::get() {
+    Mutex::Autolock lock(sSingletonLock);
+
+    if (sSingleton == NULL) {
+        ALOGD("Instantiating Session Library Singleton.");
+        sSingleton = new SessionLibrary();
+    }
+
+    return sSingleton;
+}
+
+sp<Session> SessionLibrary::createSession() {
+    Mutex::Autolock lock(mSessionsLock);
+
+    char sessionIdRaw[16];
+    snprintf(sessionIdRaw, sizeof(sessionIdRaw), "%u", mNextSessionId);
+
+    mNextSessionId += 1;
+
+    std::vector<uint8_t> sessionId;
+    sessionId.insert(sessionId.end(), sessionIdRaw,
+            sessionIdRaw + sizeof(sessionIdRaw) / sizeof(uint8_t));
+
+    mSessions.insert(std::pair<std::vector<uint8_t>,
+            sp<Session> >(sessionId, new Session(sessionId)));
+    std::map<std::vector<uint8_t>, sp<Session> >::iterator itr = mSessions.find(sessionId);
+    if (itr != mSessions.end()) {
+        return itr->second;
+    } else {
+        return nullptr;
+    }
+}
+
+sp<Session> SessionLibrary::findSession(
+        const std::vector<uint8_t>& sessionId) {
+    Mutex::Autolock lock(mSessionsLock);
+    std::map<std::vector<uint8_t>, sp<Session> >::iterator itr = mSessions.find(sessionId);
+    if (itr != mSessions.end()) {
+        return itr->second;
+    } else {
+        return nullptr;
+    }
+}
+
+void SessionLibrary::destroySession(const sp<Session>& session) {
+    Mutex::Autolock lock(mSessionsLock);
+    mSessions.erase(session->sessionId());
+}
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android

diff --git a/drm/mediadrm/plugins/clearkey/hidl/android.hardware.drm@1.1-service.clearkey.rc b/drm/mediadrm/plugins/clearkey/hidl/android.hardware.drm@1.1-service.clearkey.rc
new file mode 100644
index 0000000..ffe856a
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/android.hardware.drm@1.1-service.clearkey.rc

@@ -0,0 +1,6 @@
+service vendor.drm-clearkey-hal-1-1 /vendor/bin/hw/android.hardware.drm@1.1-service.clearkey
+    class hal
+    user media
+    group media mediadrm
+    ioprio rt 4
+    writepid /dev/cpuset/foreground/tasks

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/AesCtrDecryptor.h b/drm/mediadrm/plugins/clearkey/hidl/include/AesCtrDecryptor.h
new file mode 100644
index 0000000..0c7ef20
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/AesCtrDecryptor.h

@@ -0,0 +1,50 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_AES_CTR_DECRYPTOR_H_
+#define CLEARKEY_AES_CTR_DECRYPTOR_H_
+
+#include "ClearKeyTypes.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_0::Status;
+using ::android::hardware::drm::V1_0::SubSample;
+
+class AesCtrDecryptor {
+public:
+    AesCtrDecryptor() {}
+
+    Status decrypt(const std::vector<uint8_t>& key, const Iv iv,
+            const uint8_t* source, uint8_t* destination,
+            const std::vector<SubSample> subSamples, size_t numSubSamples,
+            size_t* bytesDecryptedOut);
+
+private:
+    CLEARKEY_DISALLOW_COPY_AND_ASSIGN(AesCtrDecryptor);
+};
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif // CLEARKEY_AES_CTR_DECRYPTOR_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/Base64.h b/drm/mediadrm/plugins/clearkey/hidl/include/Base64.h
new file mode 100644
index 0000000..4a385bd
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/Base64.h

@@ -0,0 +1,46 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef BASE_64_H_
+
+#define BASE_64_H_
+
+#include <android/hardware/drm/1.0/types.h>
+
+#include "Buffer.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::sp;
+
+struct Buffer;
+
+sp<Buffer> decodeBase64(const std::string &s);
+void encodeBase64(const void *data, size_t size, std::string *out);
+
+void encodeBase64Url(const void *data, size_t size, std::string *out);
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif  // BASE_64_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/Buffer.h b/drm/mediadrm/plugins/clearkey/hidl/include/Buffer.h
new file mode 100644
index 0000000..5bbb28a
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/Buffer.h

@@ -0,0 +1,62 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef BUFFER_H_
+#define BUFFER_H_
+
+#include <android/hardware/drm/1.0/types.h>
+#include <utils/RefBase.h>
+
+#include "ClearKeyTypes.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::sp;
+
+struct Buffer : public RefBase {
+    explicit Buffer(size_t capacity);
+
+    uint8_t *base() { return reinterpret_cast<uint8_t *>(mData); }
+    uint8_t *data() { return reinterpret_cast<uint8_t *>(mData) + mRangeOffset; }
+    size_t capacity() const { return mCapacity; }
+    size_t size() const { return mRangeLength; }
+    size_t offset() const { return mRangeOffset; }
+
+protected:
+    virtual ~Buffer();
+
+private:
+    void *mData;
+    size_t mCapacity;
+    size_t mRangeOffset;
+    size_t mRangeLength;
+
+    bool mOwnsData;
+
+    CLEARKEY_DISALLOW_COPY_AND_ASSIGN(Buffer);
+};
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif  // BUFFER_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/ClearKeyDrmProperties.h b/drm/mediadrm/plugins/clearkey/hidl/include/ClearKeyDrmProperties.h
new file mode 100644
index 0000000..d65b25c
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/ClearKeyDrmProperties.h

@@ -0,0 +1,54 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_DRM_PROPERTIES_H_
+#define CLEARKEY_DRM_PROPERTIES_H_
+
+#include <string.h>
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+static const std::string kVendorKey("vendor");
+static const std::string kVendorValue("Google");
+static const std::string kVersionKey("version");
+static const std::string kVersionValue("1.1");
+static const std::string kPluginDescriptionKey("description");
+static const std::string kPluginDescriptionValue("ClearKey CDM");
+static const std::string kAlgorithmsKey("algorithms");
+static const std::string kAlgorithmsValue("");
+static const std::string kListenerTestSupportKey("listenerTestSupport");
+static const std::string kListenerTestSupportValue("true");
+
+static const std::string kDeviceIdKey("deviceId");
+static const uint8_t kTestDeviceIdData[] =
+        {0x0, 0x1, 0x2, 0x3, 0x4, 0x5, 0x6, 0x7,
+         0x8, 0x9, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf};
+// TODO stub out metrics for nw
+static const std::string kMetricsKey("metrics");
+static const uint8_t kMetricsData[] = { 0 };
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif // CLEARKEY_DRM_PROPERTIES_H_
+

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/ClearKeyTypes.h b/drm/mediadrm/plugins/clearkey/hidl/include/ClearKeyTypes.h
new file mode 100644
index 0000000..46cb5e4
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/ClearKeyTypes.h

@@ -0,0 +1,55 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_MACROS_H_
+#define CLEARKEY_MACROS_H_
+
+#include <android/hardware/drm/1.0/types.h>
+
+#include <map>
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_0::KeyValue;
+using ::android::hardware::hidl_vec;
+
+const uint8_t kBlockSize = 16; //AES_BLOCK_SIZE;
+typedef uint8_t KeyId[kBlockSize];
+typedef uint8_t Iv[kBlockSize];
+
+typedef ::android::hardware::drm::V1_0::SubSample SubSample;
+typedef std::map<std::vector<uint8_t>, std::vector<uint8_t> > KeyMap;
+
+#define CLEARKEY_DISALLOW_COPY_AND_ASSIGN(TypeName) \
+  TypeName(const TypeName&) = delete;      \
+  void operator=(const TypeName&) = delete;
+
+#define CLEARKEY_DISALLOW_COPY_AND_ASSIGN_AND_NEW(TypeName) \
+  TypeName() = delete;                     \
+  TypeName(const TypeName&) = delete;      \
+  void operator=(const TypeName&) = delete;
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif // CLEARKEY_MACROS_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/CreatePluginFactories.h b/drm/mediadrm/plugins/clearkey/hidl/include/CreatePluginFactories.h
new file mode 100644
index 0000000..9952027
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/CreatePluginFactories.h

@@ -0,0 +1,42 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_CREATE_PLUGIN_FACTORIES_H_
+#define CLEARKEY_CREATE_PLUGIN_FACTORIES_H_
+
+#include <android/hardware/drm/1.1/ICryptoFactory.h>
+#include <android/hardware/drm/1.1/IDrmFactory.h>
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_1::ICryptoFactory;
+using ::android::hardware::drm::V1_1::IDrmFactory;
+
+extern "C" {
+    IDrmFactory* createDrmFactory();
+    ICryptoFactory* createCryptoFactory();
+}
+
+}  // namespace clearkey
+}  // namespace V1_1
+}  // namespace drm
+}  // namespace hardware
+}  // namespace android
+#endif // CLEARKEY_CREATE_PLUGIN_FACTORIES_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/CryptoFactory.h b/drm/mediadrm/plugins/clearkey/hidl/include/CryptoFactory.h
new file mode 100644
index 0000000..175ab76
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/CryptoFactory.h

@@ -0,0 +1,60 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_CRYPTO_FACTORY_H_
+#define CLEARKEY_CRYPTO_FACTORY_H_
+
+#include <android/hardware/drm/1.0/ICryptoPlugin.h>
+#include <android/hardware/drm/1.1/ICryptoFactory.h>
+
+#include "ClearKeyTypes.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_1::ICryptoFactory;
+using ::android::hardware::drm::V1_0::ICryptoPlugin;
+using ::android::hardware::hidl_array;
+using ::android::hardware::hidl_string;
+using ::android::hardware::Return;
+
+struct CryptoFactory : public ICryptoFactory {
+    CryptoFactory() {}
+    virtual ~CryptoFactory() {}
+
+    Return<bool> isCryptoSchemeSupported(const hidl_array<uint8_t, 16>& uuid)
+            override;
+
+    Return<void> createPlugin(
+            const hidl_array<uint8_t, 16>& uuid,
+            const hidl_vec<uint8_t>& initData,
+            createPlugin_cb _hidl_cb) override;
+
+private:
+    CLEARKEY_DISALLOW_COPY_AND_ASSIGN(CryptoFactory);
+
+};
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif // CLEARKEY_CRYPTO_FACTORY_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/CryptoPlugin.h b/drm/mediadrm/plugins/clearkey/hidl/include/CryptoPlugin.h
new file mode 100644
index 0000000..6a73806
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/CryptoPlugin.h

@@ -0,0 +1,104 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_CRYPTO_PLUGIN_H_
+#define CLEARKEY_CRYPTO_PLUGIN_H_
+
+#include <android/hardware/drm/1.0/ICryptoPlugin.h>
+#include <android/hidl/memory/1.0/IMemory.h>
+
+#include "ClearKeyTypes.h"
+#include "Session.h"
+#include "Utils.h"
+
+namespace {
+    static const size_t KEY_ID_SIZE = 16;
+    static const size_t KEY_IV_SIZE = 16;
+}
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_0::DestinationBuffer;
+using ::android::hardware::drm::V1_0::ICryptoPlugin;
+using ::android::hardware::drm::V1_0::Mode;
+using ::android::hardware::drm::V1_0::Pattern;
+using ::android::hardware::drm::V1_0::SharedBuffer;
+using ::android::hardware::drm::V1_0::Status;
+using ::android::hardware::drm::V1_0::SubSample;
+using ::android::hardware::hidl_array;
+using ::android::hardware::hidl_memory;
+using ::android::hardware::hidl_string;
+using ::android::hardware::hidl_vec;
+using ::android::hardware::Return;
+using ::android::hardware::Void;
+using ::android::hidl::memory::V1_0::IMemory;
+using ::android::sp;
+
+struct CryptoPlugin : public ICryptoPlugin {
+    explicit CryptoPlugin(const hidl_vec<uint8_t>& sessionId) {
+        mInitStatus = setMediaDrmSession(sessionId);
+    }
+    virtual ~CryptoPlugin() {}
+
+    Return<bool> requiresSecureDecoderComponent(const hidl_string& mime) {
+        UNUSED(mime);
+        return false;
+    }
+
+    Return<void> notifyResolution(uint32_t width, uint32_t height) {
+        UNUSED(width);
+        UNUSED(height);
+        return Void();
+    }
+
+    Return<void> decrypt(
+            bool secure,
+            const hidl_array<uint8_t, KEY_ID_SIZE>& keyId,
+            const hidl_array<uint8_t, KEY_IV_SIZE>& iv,
+            Mode mode,
+            const Pattern& pattern,
+            const hidl_vec<SubSample>& subSamples,
+            const SharedBuffer& source,
+            uint64_t offset,
+            const DestinationBuffer& destination,
+            decrypt_cb _hidl_cb);
+
+    Return<void> setSharedBufferBase(const hidl_memory& base,
+            uint32_t bufferId);
+
+    Return<Status> setMediaDrmSession(const hidl_vec<uint8_t>& sessionId);
+
+    Return<Status> getInitStatus() const { return mInitStatus; }
+
+private:
+    CLEARKEY_DISALLOW_COPY_AND_ASSIGN(CryptoPlugin);
+
+    std::map<uint32_t, sp<IMemory> > mSharedBufferMap;
+    sp<Session> mSession;
+    Status mInitStatus;
+};
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif // CLEARKEY_CRYPTO_PLUGIN_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/DrmFactory.h b/drm/mediadrm/plugins/clearkey/hidl/include/DrmFactory.h
new file mode 100644
index 0000000..6f58195
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/DrmFactory.h

@@ -0,0 +1,60 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_DRM_FACTORY_H_
+#define CLEARKEY_DRM_FACTORY_H_
+
+#include <android/hardware/drm/1.1/IDrmPlugin.h>
+#include <android/hardware/drm/1.1/IDrmFactory.h>
+
+#include "ClearKeyTypes.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::hidl_array;
+using ::android::hardware::hidl_string;
+using ::android::hardware::Return;
+
+struct DrmFactory : public IDrmFactory {
+    DrmFactory() {}
+    virtual ~DrmFactory() {}
+
+    Return<bool> isCryptoSchemeSupported(const hidl_array<uint8_t, 16>& uuid)
+            override;
+
+    Return<bool> isContentTypeSupported(const hidl_string &mimeType)
+            override;
+
+    Return<void> createPlugin(
+            const hidl_array<uint8_t, 16>& uuid,
+            const hidl_string& appPackageName,
+            createPlugin_cb _hidl_cb) override;
+
+private:
+    CLEARKEY_DISALLOW_COPY_AND_ASSIGN(DrmFactory);
+};
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif // CLEARKEY_DRM_FACTORY_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
new file mode 100644
index 0000000..fb0695a
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h

@@ -0,0 +1,343 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_DRM_PLUGIN_H_
+#define CLEARKEY_DRM_PLUGIN_H_
+
+#include <android/hardware/drm/1.1/IDrmPlugin.h>
+
+#include <stdio.h>
+#include <map>
+
+#include "SessionLibrary.h"
+#include "Utils.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_0::EventType;
+using ::android::hardware::drm::V1_0::IDrmPluginListener;
+using ::android::hardware::drm::V1_0::KeyStatus;
+using ::android::hardware::drm::V1_0::KeyType;
+using ::android::hardware::drm::V1_0::KeyValue;
+using ::android::hardware::drm::V1_0::SecureStop;
+using ::android::hardware::drm::V1_0::SecureStopId;
+using ::android::hardware::drm::V1_0::SessionId;
+using ::android::hardware::drm::V1_0::Status;
+using ::android::hardware::drm::V1_1::DrmMetricGroup;
+using ::android::hardware::drm::V1_1::IDrmPlugin;
+using ::android::hardware::drm::V1_1::KeyRequestType;
+
+using ::android::hardware::hidl_string;
+using ::android::hardware::hidl_vec;
+using ::android::hardware::Return;
+using ::android::hardware::Void;
+using ::android::sp;
+
+struct DrmPlugin : public IDrmPlugin {
+    explicit DrmPlugin(SessionLibrary* sessionLibrary);
+
+    virtual ~DrmPlugin() {}
+
+    Return<void> openSession(openSession_cb _hidl_cb) override;
+    Return<void> openSession_1_1(SecurityLevel securityLevel,
+            openSession_cb _hidl_cb) override;
+
+    Return<Status> closeSession(const hidl_vec<uint8_t>& sessionId) override;
+
+    Return<void> getKeyRequest(
+        const hidl_vec<uint8_t>& scope,
+        const hidl_vec<uint8_t>& initData,
+        const hidl_string& mimeType,
+        KeyType keyType,
+        const hidl_vec<KeyValue>& optionalParameters,
+        getKeyRequest_cb _hidl_cb) override;
+
+    Return<void> getKeyRequest_1_1(
+        const hidl_vec<uint8_t>& scope,
+        const hidl_vec<uint8_t>& initData,
+        const hidl_string& mimeType,
+        KeyType keyType,
+        const hidl_vec<KeyValue>& optionalParameters,
+        getKeyRequest_1_1_cb _hidl_cb) override;
+
+    Return<void> provideKeyResponse(
+        const hidl_vec<uint8_t>& scope,
+        const hidl_vec<uint8_t>& response,
+        provideKeyResponse_cb _hidl_cb) override;
+
+    Return<Status> removeKeys(const hidl_vec<uint8_t>& sessionId) {
+        if (sessionId.size() == 0) {
+            return Status::BAD_VALUE;
+        }
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    Return<Status> restoreKeys(
+        const hidl_vec<uint8_t>& sessionId,
+        const hidl_vec<uint8_t>& keySetId) {
+
+        if (sessionId.size() == 0 || keySetId.size() == 0) {
+            return Status::BAD_VALUE;
+        }
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    Return<void> queryKeyStatus(
+        const hidl_vec<uint8_t>& sessionId,
+        queryKeyStatus_cb _hidl_cb) override;
+
+    Return<void> getProvisionRequest(
+        const hidl_string& certificateType,
+        const hidl_string& certificateAuthority,
+        getProvisionRequest_cb _hidl_cb) {
+        UNUSED(certificateType);
+        UNUSED(certificateAuthority);
+
+        hidl_string defaultUrl;
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, hidl_vec<uint8_t>(), defaultUrl);
+        return Void();
+    }
+
+    Return<void> provideProvisionResponse(
+        const hidl_vec<uint8_t>& response,
+        provideProvisionResponse_cb _hidl_cb) {
+
+        if (response.size() == 0) {
+            _hidl_cb(Status::BAD_VALUE, hidl_vec<uint8_t>(), hidl_vec<uint8_t>());
+            return Void();
+        }
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, hidl_vec<uint8_t>(), hidl_vec<uint8_t>());
+        return Void();
+    }
+
+    Return<void> getHdcpLevels(getHdcpLevels_cb _hidl_cb) {
+        HdcpLevel connectedLevel = HdcpLevel::HDCP_NONE;
+        HdcpLevel maxLevel = HdcpLevel::HDCP_NO_OUTPUT;
+        _hidl_cb(Status::OK, connectedLevel, maxLevel);
+        return Void();
+    }
+
+    Return<void> getNumberOfSessions(getNumberOfSessions_cb _hidl_cb) override;
+
+    Return<void> getSecurityLevel(const hidl_vec<uint8_t>& sessionId,
+            getSecurityLevel_cb _hidl_cb) override;
+
+    Return<void> getMetrics(getMetrics_cb _hidl_cb) override;
+
+    Return<void> getPropertyString(
+        const hidl_string& name,
+        getPropertyString_cb _hidl_cb) override;
+
+    Return<void> getPropertyByteArray(
+        const hidl_string& name,
+        getPropertyByteArray_cb _hidl_cb) override;
+
+    Return<Status> setPropertyString(
+            const hidl_string& name, const hidl_string& value) override;
+
+    Return<Status> setPropertyByteArray(
+            const hidl_string& name, const hidl_vec<uint8_t>& value) override;
+
+    Return<Status> setCipherAlgorithm(
+            const hidl_vec<uint8_t>& sessionId, const hidl_string& algorithm) {
+        if (sessionId.size() == 0 || algorithm.size() == 0) {
+            return Status::BAD_VALUE;
+        }
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    Return<Status> setMacAlgorithm(
+            const hidl_vec<uint8_t>& sessionId, const hidl_string& algorithm) {
+        if (sessionId.size() == 0 || algorithm.size() == 0) {
+            return Status::BAD_VALUE;
+        }
+        return Status::ERROR_DRM_CANNOT_HANDLE;
+    }
+
+    Return<void> encrypt(
+            const hidl_vec<uint8_t>& sessionId,
+            const hidl_vec<uint8_t>& keyId,
+            const hidl_vec<uint8_t>& input,
+            const hidl_vec<uint8_t>& iv,
+            encrypt_cb _hidl_cb) {
+        if (sessionId.size() == 0 || keyId.size() == 0 ||
+                input.size() == 0 || iv.size() == 0) {
+            _hidl_cb(Status::BAD_VALUE, hidl_vec<uint8_t>());
+            return Void();
+        }
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, hidl_vec<uint8_t>());
+        return Void();
+    }
+
+    Return<void> decrypt(
+            const hidl_vec<uint8_t>& sessionId,
+            const hidl_vec<uint8_t>& keyId,
+            const hidl_vec<uint8_t>& input,
+            const hidl_vec<uint8_t>& iv,
+            decrypt_cb _hidl_cb) {
+        if (sessionId.size() == 0 || keyId.size() == 0 ||
+                input.size() == 0 || iv.size() == 0) {
+            _hidl_cb(Status::BAD_VALUE, hidl_vec<uint8_t>());
+            return Void();
+        }
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, hidl_vec<uint8_t>());
+        return Void();
+    }
+
+    Return<void> sign(
+            const hidl_vec<uint8_t>& sessionId,
+            const hidl_vec<uint8_t>& keyId,
+            const hidl_vec<uint8_t>& message,
+            sign_cb _hidl_cb) {
+        if (sessionId.size() == 0 || keyId.size() == 0 ||
+                message.size() == 0) {
+            _hidl_cb(Status::BAD_VALUE, hidl_vec<uint8_t>());
+            return Void();
+        }
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, hidl_vec<uint8_t>());
+        return Void();
+    }
+
+    Return<void> verify(
+            const hidl_vec<uint8_t>& sessionId,
+            const hidl_vec<uint8_t>& keyId,
+            const hidl_vec<uint8_t>& message,
+            const hidl_vec<uint8_t>& signature,
+            verify_cb _hidl_cb) {
+
+        if (sessionId.size() == 0 || keyId.size() == 0 ||
+                message.size() == 0 || signature.size() == 0) {
+            _hidl_cb(Status::BAD_VALUE, false);
+            return Void();
+        }
+        _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, false);
+        return Void();
+    }
+
+    Return<void> signRSA(
+            const hidl_vec<uint8_t>& sessionId,
+            const hidl_string& algorithm,
+            const hidl_vec<uint8_t>& message,
+            const hidl_vec<uint8_t>& wrappedKey,
+            signRSA_cb _hidl_cb) {
+        if (sessionId.size() == 0 || algorithm.size() == 0 ||
+                message.size() == 0 || wrappedKey.size() == 0) {
+             _hidl_cb(Status::BAD_VALUE, hidl_vec<uint8_t>());
+             return Void();
+         }
+         _hidl_cb(Status::ERROR_DRM_CANNOT_HANDLE, hidl_vec<uint8_t>());
+         return Void();
+    }
+
+    Return<void> setListener(const sp<IDrmPluginListener>& listener) {
+        mListener = listener;
+        return Void();
+    };
+
+    Return<void> sendEvent(EventType eventType, const hidl_vec<uint8_t>& sessionId,
+            const hidl_vec<uint8_t>& data) {
+        if (mListener != NULL) {
+            mListener->sendEvent(eventType, sessionId, data);
+        } else {
+            ALOGE("Null event listener, event not sent");
+        }
+        return Void();
+    }
+
+    Return<void> sendExpirationUpdate(const hidl_vec<uint8_t>& sessionId, int64_t expiryTimeInMS) {
+        if (mListener != NULL) {
+            mListener->sendExpirationUpdate(sessionId, expiryTimeInMS);
+        } else {
+            ALOGE("Null event listener, event not sent");
+        }
+        return Void();
+    }
+
+    Return<void> sendKeysChange(const hidl_vec<uint8_t>& sessionId,
+            const hidl_vec<KeyStatus>& keyStatusList, bool hasNewUsableKey) {
+        if (mListener != NULL) {
+            mListener->sendKeysChange(sessionId, keyStatusList, hasNewUsableKey);
+        } else {
+            ALOGE("Null event listener, event not sent");
+        }
+        return Void();
+    }
+
+    Return<void> getSecureStops(getSecureStops_cb _hidl_cb);
+
+    Return<void> getSecureStop(const hidl_vec<uint8_t>& secureStopId,
+            getSecureStop_cb _hidl_cb);
+
+    Return<Status> releaseSecureStop(const hidl_vec<uint8_t>& ssRelease);
+
+    Return<Status> releaseAllSecureStops();
+
+    Return<void> getSecureStopIds(getSecureStopIds_cb _hidl_cb);
+
+    Return<Status> releaseSecureStops(const SecureStopRelease& ssRelease);
+
+    Return<Status> removeSecureStop(const hidl_vec<uint8_t>& secureStopId);
+
+    Return<Status> removeAllSecureStops();
+
+private:
+    void initProperties();
+    void installSecureStop(const hidl_vec<uint8_t>& sessionId);
+    void setPlayPolicy();
+
+    Return<Status> setSecurityLevel(const hidl_vec<uint8_t>& sessionId,
+            SecurityLevel level);
+
+    Status getKeyRequestCommon(const hidl_vec<uint8_t>& scope,
+            const hidl_vec<uint8_t>& initData,
+            const hidl_string& mimeType,
+            KeyType keyType,
+            const hidl_vec<KeyValue>& optionalParameters,
+            std::vector<uint8_t> *request,
+            KeyRequestType *getKeyRequestType,
+            std::string *defaultUrl);
+
+    struct ClearkeySecureStop {
+        std::vector<uint8_t> id;
+        std::vector<uint8_t> data;
+    };
+
+    std::map<std::vector<uint8_t>, ClearkeySecureStop> mSecureStops;
+    std::vector<KeyValue> mPlayPolicy;
+    std::map<std::string, std::string> mStringProperties;
+    std::map<std::string, std::vector<uint8_t> > mByteArrayProperties;
+    std::map<std::vector<uint8_t>, SecurityLevel> mSecurityLevel;
+    sp<IDrmPluginListener> mListener;
+    SessionLibrary *mSessionLibrary;
+    int64_t mOpenSessionOkCount;
+    int64_t mCloseSessionOkCount;
+    int64_t mCloseSessionNotOpenedCount;
+    uint32_t mNextSecureStopId;
+
+    CLEARKEY_DISALLOW_COPY_AND_ASSIGN_AND_NEW(DrmPlugin);
+};
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif // CLEARKEY_DRM_PLUGIN_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/InitDataParser.h b/drm/mediadrm/plugins/clearkey/hidl/include/InitDataParser.h
new file mode 100644
index 0000000..3189c4a
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/InitDataParser.h

@@ -0,0 +1,56 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *            http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_INIT_DATA_PARSER_H_
+#define CLEARKEY_INIT_DATA_PARSER_H_
+
+#include <android/hardware/drm/1.0/types.h>
+
+#include "ClearKeyTypes.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_0::Status;
+
+class InitDataParser {
+public:
+    InitDataParser() {}
+
+    Status parse(const std::vector<uint8_t>& initData,
+            const std::string& type,
+            std::vector<uint8_t>* licenseRequest);
+
+private:
+    CLEARKEY_DISALLOW_COPY_AND_ASSIGN(InitDataParser);
+
+    Status parsePssh(const std::vector<uint8_t>& initData,
+            std::vector<const uint8_t*>* keyIds);
+
+    std::string generateRequest(
+            const std::vector<const uint8_t*>& keyIds);
+};
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif // CLEARKEY_INIT_DATA_PARSER_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/JsonWebKey.h b/drm/mediadrm/plugins/clearkey/hidl/include/JsonWebKey.h
new file mode 100644
index 0000000..4ab034c
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/JsonWebKey.h

@@ -0,0 +1,62 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#ifndef CLEARKEY_JSON_WEB_KEY_H_
+#define CLEARKEY_JSON_WEB_KEY_H_
+
+#include "jsmn.h"
+#include "Utils.h"
+#include "ClearKeyTypes.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+class JsonWebKey {
+ public:
+    JsonWebKey();
+    virtual ~JsonWebKey();
+
+    bool extractKeysFromJsonWebKeySet(const std::string& jsonWebKeySet,
+            KeyMap* keys);
+
+ private:
+    std::vector<jsmntok_t> mJsmnTokens;
+    std::vector<std::string> mJsonObjects;
+    std::vector<std::string> mTokens;
+
+    bool decodeBase64String(const std::string& encodedText,
+            std::vector<uint8_t>* decodedText);
+    bool findKey(const std::string& jsonObject, std::string* keyId,
+            std::string* encodedKey);
+    void findValue(const std::string &key, std::string* value);
+    bool isJsonWebKeySet(const std::string& jsonObject) const;
+    bool parseJsonObject(const std::string& jsonObject,
+            std::vector<std::string>* tokens);
+    bool parseJsonWebKeySet(const std::string& jsonWebKeySet,
+            std::vector<std::string>* jsonObjects);
+
+    CLEARKEY_DISALLOW_COPY_AND_ASSIGN(JsonWebKey);
+};
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif  // CLEARKEY_JSON_WEB_KEY_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/Session.h b/drm/mediadrm/plugins/clearkey/hidl/include/Session.h
new file mode 100644
index 0000000..cddfca5
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/Session.h

@@ -0,0 +1,71 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_SESSION_H_
+#define CLEARKEY_SESSION_H_
+
+#include <utils/Mutex.h>
+#include <utils/RefBase.h>
+#include <vector>
+
+#include "ClearKeyTypes.h"
+
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::drm::V1_0::Status;
+using ::android::hardware::drm::V1_0::SubSample;
+
+class Session : public RefBase {
+public:
+    explicit Session(const std::vector<uint8_t>& sessionId)
+            : mSessionId(sessionId) {}
+    virtual ~Session() {}
+
+    const std::vector<uint8_t>& sessionId() const { return mSessionId; }
+
+    Status getKeyRequest(
+            const std::vector<uint8_t>& mimeType,
+            const std::string& initDataType,
+            std::vector<uint8_t>* keyRequest) const;
+
+    Status provideKeyResponse(
+            const std::vector<uint8_t>& response);
+
+    Status decrypt(
+            const KeyId keyId, const Iv iv, const uint8_t* srcPtr,
+            uint8_t* dstPtr, const std::vector<SubSample> subSamples,
+            size_t* bytesDecryptedOut);
+
+private:
+    CLEARKEY_DISALLOW_COPY_AND_ASSIGN(Session);
+
+    const std::vector<uint8_t> mSessionId;
+    KeyMap mKeyMap;
+    Mutex mMapLock;
+};
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif // CLEARKEY_SESSION_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/SessionLibrary.h b/drm/mediadrm/plugins/clearkey/hidl/include/SessionLibrary.h
new file mode 100644
index 0000000..326a0c1
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/SessionLibrary.h

@@ -0,0 +1,66 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_SESSION_LIBRARY_H_
+#define CLEARKEY_SESSION_LIBRARY_H_
+
+#include <utils/RefBase.h>
+#include <utils/Mutex.h>
+
+#include "ClearKeyTypes.h"
+#include "Session.h"
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::sp;
+
+class SessionLibrary : public RefBase {
+public:
+    static SessionLibrary* get();
+
+    sp<Session> createSession();
+
+    sp<Session> findSession(
+            const std::vector<uint8_t>& sessionId);
+
+    void destroySession(const sp<Session>& session);
+
+    size_t numOpenSessions() const { return mSessions.size(); }
+
+private:
+    CLEARKEY_DISALLOW_COPY_AND_ASSIGN(SessionLibrary);
+
+    SessionLibrary() : mNextSessionId(1) {}
+
+    static Mutex sSingletonLock;
+    static SessionLibrary* sSingleton;
+
+    Mutex mSessionsLock;
+    uint32_t mNextSessionId;
+    std::map<std::vector<uint8_t>, sp<Session> > mSessions;
+};
+
+} // namespace clearkey
+} // namespace V1_1
+} // namespace drm
+} // namespace hardware
+} // namespace android
+
+#endif // CLEARKEY_SESSION_LIBRARY_H_

diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/TypeConvert.h b/drm/mediadrm/plugins/clearkey/hidl/include/TypeConvert.h
new file mode 100644
index 0000000..cc06329
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/TypeConvert.h

@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef CLEARKEY_ANDROID_HARDWARE_DRM_V1_1_TYPECONVERT
+#define CLEARKEY_ANDROID_HARDWARE_DRM_V1_1_TYPECONVERT
+
+#include <vector>
+
+#include <android/hardware/drm/1.0/types.h>
+
+namespace android {
+namespace hardware {
+namespace drm {
+namespace V1_1 {
+namespace clearkey {
+
+using ::android::hardware::hidl_array;
+using ::android::hardware::hidl_vec;
+
+template<typename T> const hidl_vec<T> toHidlVec(const std::vector<T> &vec) {
+    hidl_vec<T> hVec;
+    hVec.setToExternal(const_cast<T *>(vec.data()), vec.size());
+    return hVec;
+}
+
+template<typename T> hidl_vec<T> toHidlVec(std::vector<T> &vec) {
+    hidl_vec<T> hVec;
+    hVec.setToExternal(vec.data(), vec.size());
+    return hVec;
+}
+
+template<typename T> const std::vector<T> toVector(const hidl_vec<T> &hVec) {
+    std::vector<T> vec;
+    vec.assign(hVec.data(), hVec.data() + hVec.size());
+    return *const_cast<const std::vector<T> *>(&vec);
+}
+
+template<typename T> std::vector<T> toVector(hidl_vec<T> &hVec) {
+    std::vector<T> vec;
+    vec.assign(hVec.data(), hVec.data() + hVec.size());
+    return vec;
+}
+
+template<typename T, size_t SIZE> const std::vector<T> toVector(
+        const hidl_array<T, SIZE> &hArray) {
+    std::vector<T> vec;
+    vec.assign(hArray.data(), hArray.data() + hArray.size());
+    return vec;
+}
+
+template<typename T, size_t SIZE> std::vector<T> toVector(
+        hidl_array<T, SIZE> &hArray) {
+    std::vector<T> vec;
+    vec.assign(hArray.data(), hArray.data() + hArray.size());
+    return vec;
+}
+
+}  // namespace clearkey
+}  // namespace V1_1
+}  // namespace drm
+}  // namespace hardware
+}  // namespace android
+
+#endif // CLEARKEY_ANDROID_HARDWARE_DRM_V1_1_TYPECONVERT

diff --git a/drm/mediadrm/plugins/clearkey/hidl/service.cpp b/drm/mediadrm/plugins/clearkey/hidl/service.cpp
new file mode 100644
index 0000000..6a97b72
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/hidl/service.cpp

@@ -0,0 +1,54 @@
+/*
+ * Copyright 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#define LOG_TAG "android.hardware.drm@1.1-service.clearkey"
+
+#include <CryptoFactory.h>
+#include <DrmFactory.h>
+
+#include <android-base/logging.h>
+#include <binder/ProcessState.h>
+#include <hidl/HidlTransportSupport.h>
+
+using ::android::hardware::configureRpcThreadpool;
+using ::android::hardware::joinRpcThreadpool;
+using ::android::sp;
+
+using android::hardware::drm::V1_1::ICryptoFactory;
+using android::hardware::drm::V1_1::IDrmFactory;
+using android::hardware::drm::V1_1::clearkey::CryptoFactory;
+using android::hardware::drm::V1_1::clearkey::DrmFactory;
+
+
+int main(int /* argc */, char** /* argv */) {
+    ALOGD("android.hardware.drm@1.1-service.clearkey starting...");
+
+    // The DRM HAL may communicate to other vendor components via
+    // /dev/vndbinder
+    android::ProcessState::initWithDriver("/dev/vndbinder");
+
+    sp<IDrmFactory> drmFactory = new DrmFactory;
+    sp<ICryptoFactory> cryptoFactory = new CryptoFactory;
+
+    configureRpcThreadpool(8, true /* callerWillJoin */);
+
+    // Setup hwbinder service
+    CHECK_EQ(drmFactory->registerAsService("clearkey"), android::NO_ERROR)
+        << "Failed to register Clearkey Factory HAL";
+    CHECK_EQ(cryptoFactory->registerAsService("clearkey"), android::NO_ERROR)
+        << "Failed to register Clearkey Crypto  HAL";
+
+    joinRpcThreadpool();
+}

diff --git a/drm/mediadrm/plugins/clearkey/tests/Android.bp b/drm/mediadrm/plugins/clearkey/tests/Android.bp
deleted file mode 100644
index 0fcfc64..0000000
--- a/drm/mediadrm/plugins/clearkey/tests/Android.bp
+++ /dev/null

@@ -1,38 +0,0 @@
-//
-// Copyright (C) 2014 The Android Open Source Project
-//
-// Licensed under the Apache License, Version 2.0 (the "License");
-// you may not use this file except in compliance with the License.
-// You may obtain a copy of the License at
-//
-//      http://www.apache.org/licenses/LICENSE-2.0
-//
-// Unless required by applicable law or agreed to in writing, software
-// distributed under the License is distributed on an "AS IS" BASIS,
-// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-// See the License for the specific language governing permissions and
-// limitations under the License.
-//
-// ----------------------------------------------------------------
-// Builds ClearKey Drm Tests
-//
-
-cc_test {
-    name: "ClearKeyDrmUnitTest",
-    vendor: true,
-
-    srcs: [
-        "AesCtrDecryptorUnittest.cpp",
-        "InitDataParserUnittest.cpp",
-        "JsonWebKeyUnittest.cpp",
-    ],
-
-    shared_libs: [
-        "libcrypto",
-        "libdrmclearkeyplugin",
-        "liblog",
-        "libstagefright_foundation",
-        "libutils",
-    ],
-    header_libs: ["media_plugin_headers"],
-}

diff --git a/drm/mediadrm/plugins/mock/Android.bp b/drm/mediadrm/plugins/mock/Android.bp
index abd1884..dd2ad7b 100644
--- a/drm/mediadrm/plugins/mock/Android.bp
+++ b/drm/mediadrm/plugins/mock/Android.bp

@@ -32,5 +32,7 @@
     cflags: [
         // Set the following flag to enable the decryption passthru flow
         //"-DENABLE_PASSTHRU_DECRYPTION",
+        "-Wall",
+        "-Werror",
     ],
 }

diff --git a/include/OWNERS b/include/OWNERS
index 3cb6d9c..d6bd998 100644
--- a/include/OWNERS
+++ b/include/OWNERS

@@ -1,5 +1,5 @@
 elaurent@google.com
-gkasten@android.com
+gkasten@google.com
 hunga@google.com
 jtinker@google.com
 lajos@google.com

diff --git a/include/common_time/OWNERS b/include/common_time/OWNERS
new file mode 100644
index 0000000..f9cb567
--- /dev/null
+++ b/include/common_time/OWNERS

@@ -0,0 +1 @@
+gkasten@google.com

diff --git a/include/media/AudioClient.h b/include/media/AudioClient.h
deleted file mode 100644
index 9efd76d..0000000
--- a/include/media/AudioClient.h
+++ /dev/null

@@ -1,38 +0,0 @@
-/*
- * Copyright (C) 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-
-#ifndef ANDROID_AUDIO_CLIENT_H
-#define ANDROID_AUDIO_CLIENT_H
-
-#include <system/audio.h>
-#include <utils/String16.h>
-
-namespace android {
-
-class AudioClient {
- public:
-    AudioClient() :
-        clientUid(-1), clientPid(-1), packageName("") {}
-
-    uid_t clientUid;
-    pid_t clientPid;
-    String16 packageName;
-};
-
-}; // namespace android
-
-#endif  // ANDROID_AUDIO_CLIENT_H

diff --git a/include/media/AudioClient.h b/include/media/AudioClient.h
new file mode 120000
index 0000000..a0530e4
--- /dev/null
+++ b/include/media/AudioClient.h

@@ -0,0 +1 @@
+../../media/libaudioclient/include/media/AudioClient.h
\ No newline at end of file

diff --git a/include/media/AudioPresentationInfo.h b/include/media/AudioPresentationInfo.h
new file mode 100644
index 0000000..e91a992
--- /dev/null
+++ b/include/media/AudioPresentationInfo.h

@@ -0,0 +1,79 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AUDIO_PRESENTATION_INFO_H_
+#define AUDIO_PRESENTATION_INFO_H_
+
+#include <sstream>
+#include <stdint.h>
+
+#include <utils/KeyedVector.h>
+#include <utils/RefBase.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+enum mastering_indication {
+    MASTERING_NOT_INDICATED,
+    MASTERED_FOR_STEREO,
+    MASTERED_FOR_SURROUND,
+    MASTERED_FOR_3D,
+    MASTERED_FOR_HEADPHONE,
+};
+
+struct AudioPresentation : public RefBase {
+    int32_t mPresentationId;
+    int32_t mProgramId;
+    KeyedVector<String8, String8> mLabels;
+    String8 mLanguage;
+    int32_t mMasteringIndication;
+    bool mAudioDescriptionAvailable;
+    bool mSpokenSubtitlesAvailable;
+    bool mDialogueEnhancementAvailable;
+
+    AudioPresentation() {
+        mPresentationId = -1;
+        mProgramId = -1;
+        mLanguage = "";
+        mMasteringIndication = MASTERING_NOT_INDICATED;
+        mAudioDescriptionAvailable = false;
+        mSpokenSubtitlesAvailable = false;
+        mDialogueEnhancementAvailable = false;
+    }
+};
+
+typedef Vector<sp<AudioPresentation>> AudioPresentations;
+
+class AudioPresentationInfo : public RefBase {
+ public:
+    AudioPresentationInfo();
+
+    ~AudioPresentationInfo();
+
+    void addPresentation(sp<AudioPresentation> presentation);
+
+    size_t countPresentations() const;
+
+    const sp<AudioPresentation> getPresentation(size_t index) const;
+
+ private:
+    AudioPresentations mPresentations;
+};
+
+}  // namespace android
+
+#endif  // AUDIO_PRESENTATION_INFO_H_

diff --git a/include/media/CounterMetric.h b/include/media/CounterMetric.h
new file mode 120000
index 0000000..baba043
--- /dev/null
+++ b/include/media/CounterMetric.h

@@ -0,0 +1 @@
+../../media/libmedia/include/media/CounterMetric.h
\ No newline at end of file

diff --git a/include/media/DataSource.h b/include/media/DataSource.h
new file mode 120000
index 0000000..905bec1
--- /dev/null
+++ b/include/media/DataSource.h

@@ -0,0 +1 @@
+../../media/libmediaextractor/include/media/DataSource.h
\ No newline at end of file

diff --git a/include/media/DataSourceBase.h b/include/media/DataSourceBase.h
new file mode 120000
index 0000000..54c8047
--- /dev/null
+++ b/include/media/DataSourceBase.h

@@ -0,0 +1 @@
+../../media/libmediaextractor/include/media/DataSourceBase.h
\ No newline at end of file

diff --git a/include/media/EventMetric.h b/include/media/EventMetric.h
new file mode 120000
index 0000000..5707d9a
--- /dev/null
+++ b/include/media/EventMetric.h

@@ -0,0 +1 @@
+../../media/libmedia/include/media/EventMetric.h
\ No newline at end of file

diff --git a/include/media/ExtractorUtils.h b/include/media/ExtractorUtils.h
new file mode 120000
index 0000000..e2dd082
--- /dev/null
+++ b/include/media/ExtractorUtils.h

@@ -0,0 +1 @@
+../../media/libmediaextractor/include/media/ExtractorUtils.h
\ No newline at end of file

diff --git a/include/media/IAudioRecord.h b/include/media/IAudioRecord.h
deleted file mode 120000
index 7fbf8f2..0000000
--- a/include/media/IAudioRecord.h
+++ /dev/null

@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IAudioRecord.h
\ No newline at end of file

diff --git a/include/media/IHDCP.h b/include/media/IHDCP.h
deleted file mode 120000
index 9d4568e..0000000
--- a/include/media/IHDCP.h
+++ /dev/null

@@ -1 +0,0 @@
-../../media/libmedia/include/media/IHDCP.h
\ No newline at end of file

diff --git a/include/media/IMediaCodecService.h b/include/media/IMediaCodecService.h
deleted file mode 120000
index 37f6822..0000000
--- a/include/media/IMediaCodecService.h
+++ /dev/null

@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaCodecService.h
\ No newline at end of file

diff --git a/include/media/MediaDefs.h b/include/media/MediaDefs.h
deleted file mode 120000
index 9850603..0000000
--- a/include/media/MediaDefs.h
+++ /dev/null

@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaDefs.h
\ No newline at end of file

diff --git a/include/media/MediaExtractor.h b/include/media/MediaExtractor.h
new file mode 120000
index 0000000..4b35fe1
--- /dev/null
+++ b/include/media/MediaExtractor.h

@@ -0,0 +1 @@
+../../media/libmediaextractor/include/media/MediaExtractor.h
\ No newline at end of file

diff --git a/include/media/MediaSource.h b/include/media/MediaSource.h
new file mode 120000
index 0000000..2e147c4
--- /dev/null
+++ b/include/media/MediaSource.h

@@ -0,0 +1 @@
+../../media/libmediaextractor/include/media/MediaSource.h
\ No newline at end of file

diff --git a/include/media/MediaTrack.h b/include/media/MediaTrack.h
new file mode 120000
index 0000000..5a63287a
--- /dev/null
+++ b/include/media/MediaTrack.h

@@ -0,0 +1 @@
+../../media/libmediaextractor/include/media/MediaTrack.h
\ No newline at end of file

diff --git a/include/media/MicrophoneInfo.h b/include/media/MicrophoneInfo.h
new file mode 100644
index 0000000..2287aca
--- /dev/null
+++ b/include/media/MicrophoneInfo.h

@@ -0,0 +1,249 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_MICROPHONE_INFO_H
+#define ANDROID_MICROPHONE_INFO_H
+
+#include <binder/Parcel.h>
+#include <binder/Parcelable.h>
+#include <system/audio.h>
+#include <utils/String16.h>
+#include <utils/Vector.h>
+
+namespace android {
+namespace media {
+
+#define RETURN_IF_FAILED(calledOnce)                                     \
+    {                                                                    \
+        status_t returnStatus = calledOnce;                              \
+        if (returnStatus) {                                              \
+            ALOGE("Failed at %s:%d (%s)", __FILE__, __LINE__, __func__); \
+            return returnStatus;                                         \
+         }                                                               \
+    }
+
+class MicrophoneInfo : public Parcelable {
+public:
+    MicrophoneInfo() = default;
+    MicrophoneInfo(const MicrophoneInfo& microphoneInfo) = default;
+    MicrophoneInfo(audio_microphone_characteristic_t& characteristic) {
+        mDeviceId = String16(&characteristic.device_id[0]);
+        mPortId = characteristic.id;
+        mType = characteristic.device;
+        mAddress = String16(&characteristic.address[0]);
+        mDeviceLocation = characteristic.location;
+        mDeviceGroup = characteristic.group;
+        mIndexInTheGroup = characteristic.index_in_the_group;
+        mGeometricLocation.push_back(characteristic.geometric_location.x);
+        mGeometricLocation.push_back(characteristic.geometric_location.y);
+        mGeometricLocation.push_back(characteristic.geometric_location.z);
+        mOrientation.push_back(characteristic.orientation.x);
+        mOrientation.push_back(characteristic.orientation.y);
+        mOrientation.push_back(characteristic.orientation.z);
+        Vector<float> frequencies;
+        Vector<float> responses;
+        for (size_t i = 0; i < characteristic.num_frequency_responses; i++) {
+            frequencies.push_back(characteristic.frequency_responses[0][i]);
+            responses.push_back(characteristic.frequency_responses[1][i]);
+        }
+        mFrequencyResponses.push_back(frequencies);
+        mFrequencyResponses.push_back(responses);
+        for (size_t i = 0; i < AUDIO_CHANNEL_COUNT_MAX; i++) {
+            mChannelMapping.push_back(characteristic.channel_mapping[i]);
+        }
+        mSensitivity = characteristic.sensitivity;
+        mMaxSpl = characteristic.max_spl;
+        mMinSpl = characteristic.min_spl;
+        mDirectionality = characteristic.directionality;
+    }
+
+    virtual ~MicrophoneInfo() = default;
+
+    virtual status_t writeToParcel(Parcel* parcel) const {
+        RETURN_IF_FAILED(parcel->writeString16(mDeviceId));
+        RETURN_IF_FAILED(parcel->writeInt32(mPortId));
+        RETURN_IF_FAILED(parcel->writeUint32(mType));
+        RETURN_IF_FAILED(parcel->writeString16(mAddress));
+        RETURN_IF_FAILED(parcel->writeInt32(mDeviceLocation));
+        RETURN_IF_FAILED(parcel->writeInt32(mDeviceGroup));
+        RETURN_IF_FAILED(parcel->writeInt32(mIndexInTheGroup));
+        RETURN_IF_FAILED(writeFloatVector(parcel, mGeometricLocation));
+        RETURN_IF_FAILED(writeFloatVector(parcel, mOrientation));
+        if (mFrequencyResponses.size() != 2) {
+            return BAD_VALUE;
+        }
+        for (size_t i = 0; i < mFrequencyResponses.size(); i++) {
+            RETURN_IF_FAILED(parcel->writeInt32(mFrequencyResponses[i].size()));
+            RETURN_IF_FAILED(writeFloatVector(parcel, mFrequencyResponses[i]));
+        }
+        std::vector<int> channelMapping;
+        for (size_t i = 0; i < mChannelMapping.size(); ++i) {
+            channelMapping.push_back(mChannelMapping[i]);
+        }
+        RETURN_IF_FAILED(parcel->writeInt32Vector(channelMapping));
+        RETURN_IF_FAILED(parcel->writeFloat(mSensitivity));
+        RETURN_IF_FAILED(parcel->writeFloat(mMaxSpl));
+        RETURN_IF_FAILED(parcel->writeFloat(mMinSpl));
+        RETURN_IF_FAILED(parcel->writeInt32(mDirectionality));
+        return OK;
+    }
+
+    virtual status_t readFromParcel(const Parcel* parcel) {
+        RETURN_IF_FAILED(parcel->readString16(&mDeviceId));
+        RETURN_IF_FAILED(parcel->readInt32(&mPortId));
+        RETURN_IF_FAILED(parcel->readUint32(&mType));
+        RETURN_IF_FAILED(parcel->readString16(&mAddress));
+        RETURN_IF_FAILED(parcel->readInt32(&mDeviceLocation));
+        RETURN_IF_FAILED(parcel->readInt32(&mDeviceGroup));
+        RETURN_IF_FAILED(parcel->readInt32(&mIndexInTheGroup));
+        RETURN_IF_FAILED(readFloatVector(parcel, &mGeometricLocation, 3));
+        RETURN_IF_FAILED(readFloatVector(parcel, &mOrientation, 3));
+        int32_t frequenciesNum;
+        RETURN_IF_FAILED(parcel->readInt32(&frequenciesNum));
+        Vector<float> frequencies;
+        RETURN_IF_FAILED(readFloatVector(parcel, &frequencies, frequenciesNum));
+        int32_t responsesNum;
+        RETURN_IF_FAILED(parcel->readInt32(&responsesNum));
+        Vector<float> responses;
+        RETURN_IF_FAILED(readFloatVector(parcel, &responses, responsesNum));
+        if (frequencies.size() != responses.size()) {
+            return BAD_VALUE;
+        }
+        mFrequencyResponses.push_back(frequencies);
+        mFrequencyResponses.push_back(responses);
+        std::vector<int> channelMapping;
+        status_t result = parcel->readInt32Vector(&channelMapping);
+        if (result != OK) {
+            return result;
+        }
+        if (channelMapping.size() != AUDIO_CHANNEL_COUNT_MAX) {
+            return BAD_VALUE;
+        }
+        for (size_t i = 0; i < channelMapping.size(); i++) {
+            mChannelMapping.push_back(channelMapping[i]);
+        }
+        RETURN_IF_FAILED(parcel->readFloat(&mSensitivity));
+        RETURN_IF_FAILED(parcel->readFloat(&mMaxSpl));
+        RETURN_IF_FAILED(parcel->readFloat(&mMinSpl));
+        RETURN_IF_FAILED(parcel->readInt32(&mDirectionality));
+        return OK;
+    }
+
+    String16 getDeviceId() const {
+        return mDeviceId;
+    }
+
+    int getPortId() const {
+        return mPortId;
+    }
+
+    unsigned int getType() const {
+        return mType;
+    }
+
+    String16 getAddress() const {
+        return mAddress;
+    }
+
+    int getDeviceLocation() const {
+        return mDeviceLocation;
+    }
+
+    int getDeviceGroup() const {
+        return mDeviceGroup;
+    }
+
+    int getIndexInTheGroup() const {
+        return mIndexInTheGroup;
+    }
+
+    const Vector<float>& getGeometricLocation() const {
+        return mGeometricLocation;
+    }
+
+    const Vector<float>& getOrientation() const {
+        return mOrientation;
+    }
+
+    const Vector<Vector<float>>& getFrequencyResponses() const {
+        return mFrequencyResponses;
+    }
+
+    const Vector<int>& getChannelMapping() const {
+        return mChannelMapping;
+    }
+
+    float getSensitivity() const {
+        return mSensitivity;
+    }
+
+    float getMaxSpl() const {
+        return mMaxSpl;
+    }
+
+    float getMinSpl() const {
+        return mMinSpl;
+    }
+
+    int getDirectionality() const {
+        return mDirectionality;
+    }
+
+private:
+    status_t readFloatVector(
+            const Parcel* parcel, Vector<float> *vectorPtr, size_t defaultLength) {
+        std::unique_ptr<std::vector<float>> v;
+        status_t result = parcel->readFloatVector(&v);
+        if (result != OK) return result;
+        vectorPtr->clear();
+        if (v.get() != nullptr) {
+            for (const auto& iter : *v) {
+                vectorPtr->push_back(iter);
+            }
+        } else {
+            vectorPtr->resize(defaultLength);
+        }
+        return OK;
+    }
+    status_t writeFloatVector(Parcel* parcel, const Vector<float>& vector) const {
+        std::vector<float> v;
+        for (size_t i = 0; i < vector.size(); i++) {
+            v.push_back(vector[i]);
+        }
+        return parcel->writeFloatVector(v);
+    }
+
+    String16 mDeviceId;
+    int32_t mPortId;
+    uint32_t mType;
+    String16 mAddress;
+    int32_t mDeviceLocation;
+    int32_t mDeviceGroup;
+    int32_t mIndexInTheGroup;
+    Vector<float> mGeometricLocation;
+    Vector<float> mOrientation;
+    Vector<Vector<float>> mFrequencyResponses;
+    Vector<int> mChannelMapping;
+    float mSensitivity;
+    float mMaxSpl;
+    float mMinSpl;
+    int32_t mDirectionality;
+};
+
+} // namespace media
+} // namespace android
+
+#endif

diff --git a/include/media/MmapStreamCallback.h b/include/media/MmapStreamCallback.h
index 8098e79..31b8eb5 100644
--- a/include/media/MmapStreamCallback.h
+++ b/include/media/MmapStreamCallback.h

@@ -31,8 +31,9 @@
      * The mmap stream should be torn down because conditions that permitted its creation with
      * the requested parameters have changed and do not allow it to operate with the requested
      * constraints any more.
+     * \param[in] handle handle for the client stream to tear down.
      */
-    virtual void onTearDown() = 0;
+    virtual void onTearDown(audio_port_handle_t handle) = 0;
 
     /**
      * The volume to be applied to the use case specified when opening the stream has changed

diff --git a/include/media/MmapStreamInterface.h b/include/media/MmapStreamInterface.h
index d689e25..0196a0c 100644
--- a/include/media/MmapStreamInterface.h
+++ b/include/media/MmapStreamInterface.h

@@ -52,6 +52,9 @@
      * \param[in,out] deviceId audio device the stream should preferably be routed to/from
      *                       Requested as input,
      *                       Actual as output
+     * \param[in,out] sessionId audio sessionId for the stream
+     *                       Requested as input, may be AUDIO_SESSION_ALLOCATE
+     *                       Actual as output
      * \param[in] callback the MmapStreamCallback interface used by AudioFlinger to notify
      *                     condition changes affecting the stream operation
      * \param[out] interface the MmapStreamInterface interface controlling the created stream
@@ -66,6 +69,7 @@
                                            audio_config_base_t *config,
                                            const AudioClient& client,
                                            audio_port_handle_t *deviceId,
+                                           audio_session_t *sessionId,
                                            const sp<MmapStreamCallback>& callback,
                                            sp<MmapStreamInterface>& interface,
                                            audio_port_handle_t *handle);

diff --git a/include/media/TimeCheck.h b/include/media/TimeCheck.h
new file mode 120000
index 0000000..e3ef134
--- /dev/null
+++ b/include/media/TimeCheck.h

@@ -0,0 +1 @@
+../../media/libmedia/include/media/TimeCheck.h
\ No newline at end of file

diff --git a/include/media/VolumeShaper.h b/include/media/VolumeShaper.h
index 302641f..a3aaece 100644
--- a/include/media/VolumeShaper.h
+++ b/include/media/VolumeShaper.h

@@ -37,6 +37,8 @@
 
 namespace android {
 
+namespace media {
+
 // The native VolumeShaper class mirrors the java VolumeShaper class;
 // in addition, the native class contains implementation for actual operation.
 //
@@ -101,7 +103,7 @@
      * See "frameworks/base/media/java/android/media/VolumeShaper.java" for
      * details on the Java implementation.
      */
-    class Configuration : public Interpolator<S, T>, public RefBase {
+    class Configuration : public Interpolator<S, T>, public RefBase, public Parcelable {
     public:
         // Must match with VolumeShaper.java in frameworks/base.
         enum Type : int32_t {
@@ -283,7 +285,7 @@
         }
 
         // The parcel layout must match VolumeShaper.java
-        status_t writeToParcel(Parcel *parcel) const {
+        status_t writeToParcel(Parcel *parcel) const override {
             if (parcel == nullptr) return BAD_VALUE;
             return parcel->writeInt32((int32_t)mType)
                     ?: parcel->writeInt32(mId)
@@ -294,17 +296,17 @@
                             ?: Interpolator<S, T>::writeToParcel(parcel);
         }
 
-        status_t readFromParcel(const Parcel &parcel) {
+        status_t readFromParcel(const Parcel *parcel) override {
             int32_t type, optionFlags;
-            return parcel.readInt32(&type)
+            return parcel->readInt32(&type)
                     ?: setType((Type)type)
-                    ?: parcel.readInt32(&mId)
+                    ?: parcel->readInt32(&mId)
                     ?: mType == TYPE_ID
                         ? NO_ERROR
-                        : parcel.readInt32(&optionFlags)
+                        : parcel->readInt32(&optionFlags)
                             ?: setOptionFlags((OptionFlag)optionFlags)
-                            ?: parcel.readDouble(&mDurationMs)
-                            ?: Interpolator<S, T>::readFromParcel(parcel)
+                            ?: parcel->readDouble(&mDurationMs)
+                            ?: Interpolator<S, T>::readFromParcel(*parcel)
                             ?: checkCurve();
         }
 
@@ -336,7 +338,7 @@
      * See "frameworks/base/media/java/android/media/VolumeShaper.java" for
      * details on the Java implementation.
      */
-    class Operation : public RefBase {
+    class Operation : public RefBase, public Parcelable {
     public:
         // Must match with VolumeShaper.java.
         enum Flag : int32_t {
@@ -418,18 +420,18 @@
             return NO_ERROR;
         }
 
-        status_t writeToParcel(Parcel *parcel) const {
+        status_t writeToParcel(Parcel *parcel) const override {
             if (parcel == nullptr) return BAD_VALUE;
             return parcel->writeInt32((int32_t)mFlags)
                     ?: parcel->writeInt32(mReplaceId)
                     ?: parcel->writeFloat(mXOffset);
         }
 
-        status_t readFromParcel(const Parcel &parcel) {
+        status_t readFromParcel(const Parcel *parcel) override {
             int32_t flags;
-            return parcel.readInt32(&flags)
-                    ?: parcel.readInt32(&mReplaceId)
-                    ?: parcel.readFloat(&mXOffset)
+            return parcel->readInt32(&flags)
+                    ?: parcel->readInt32(&mReplaceId)
+                    ?: parcel->readFloat(&mXOffset)
                     ?: setFlags((Flag)flags);
         }
 
@@ -455,7 +457,7 @@
      * See "frameworks/base/media/java/android/media/VolumeShaper.java" for
      * details on the Java implementation.
      */
-    class State : public RefBase {
+    class State : public RefBase, public Parcelable {
     public:
         State(T volume, S xOffset)
             : mVolume(volume)
@@ -481,15 +483,15 @@
             mXOffset = xOffset;
         }
 
-        status_t writeToParcel(Parcel *parcel) const {
+        status_t writeToParcel(Parcel *parcel) const override {
             if (parcel == nullptr) return BAD_VALUE;
             return parcel->writeFloat(mVolume)
                     ?: parcel->writeFloat(mXOffset);
         }
 
-        status_t readFromParcel(const Parcel &parcel) {
-            return parcel.readFloat(&mVolume)
-                     ?: parcel.readFloat(&mXOffset);
+        status_t readFromParcel(const Parcel *parcel) override {
+            return parcel->readFloat(&mVolume)
+                     ?: parcel->readFloat(&mXOffset);
         }
 
         std::string toString() const {
@@ -1020,6 +1022,8 @@
     std::list<VolumeShaper> mVolumeShapers; // list provides stable iterators on erase
 }; // VolumeHandler
 
+} // namespace media
+
 } // namespace android
 
 #pragma pop_macro("LOG_TAG")

diff --git a/include/media/VorbisComment.h b/include/media/VorbisComment.h
new file mode 120000
index 0000000..adaa489
--- /dev/null
+++ b/include/media/VorbisComment.h

@@ -0,0 +1 @@
+../../media/libmediaextractor/include/media/VorbisComment.h
\ No newline at end of file

diff --git a/include/media/nbaio/NBLog.h b/include/media/nbaio/NBLog.h
deleted file mode 120000
index c35401e..0000000
--- a/include/media/nbaio/NBLog.h
+++ /dev/null

@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/NBLog.h
\ No newline at end of file

diff --git a/include/media/nbaio/PerformanceAnalysis.h b/include/media/nbaio/PerformanceAnalysis.h
deleted file mode 120000
index 7acfc90..0000000
--- a/include/media/nbaio/PerformanceAnalysis.h
+++ /dev/null

@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/PerformanceAnalysis.h
\ No newline at end of file

diff --git a/include/media/nblog/NBLog.h b/include/media/nblog/NBLog.h
new file mode 120000
index 0000000..3cc366c
--- /dev/null
+++ b/include/media/nblog/NBLog.h

@@ -0,0 +1 @@
+../../../media/libnblog/include/media/nblog/NBLog.h
\ No newline at end of file

diff --git a/include/media/nblog/PerformanceAnalysis.h b/include/media/nblog/PerformanceAnalysis.h
new file mode 120000
index 0000000..6ead3bc
--- /dev/null
+++ b/include/media/nblog/PerformanceAnalysis.h

@@ -0,0 +1 @@
+../../../media/libnblog/include/media/nblog/PerformanceAnalysis.h
\ No newline at end of file

diff --git a/include/media/nblog/ReportPerformance.h b/include/media/nblog/ReportPerformance.h
new file mode 120000
index 0000000..e9b8e80
--- /dev/null
+++ b/include/media/nblog/ReportPerformance.h

@@ -0,0 +1 @@
+../../../media/libnblog/include/media/nblog/ReportPerformance.h
\ No newline at end of file

diff --git a/include/media/Crypto.h b/include/mediadrm/Crypto.h
similarity index 100%
rename from include/media/Crypto.h
rename to include/mediadrm/Crypto.h


diff --git a/include/media/CryptoHal.h b/include/mediadrm/CryptoHal.h
similarity index 100%
rename from include/media/CryptoHal.h
rename to include/mediadrm/CryptoHal.h


diff --git a/include/media/Drm.h b/include/mediadrm/Drm.h
similarity index 100%
rename from include/media/Drm.h
rename to include/mediadrm/Drm.h


diff --git a/include/media/DrmHal.h b/include/mediadrm/DrmHal.h
similarity index 100%
rename from include/media/DrmHal.h
rename to include/mediadrm/DrmHal.h


diff --git a/include/mediadrm/DrmMetrics.h b/include/mediadrm/DrmMetrics.h
new file mode 120000
index 0000000..abc966b
--- /dev/null
+++ b/include/mediadrm/DrmMetrics.h

@@ -0,0 +1 @@
+../../media/libmedia/include/media/DrmMetrics.h
\ No newline at end of file

diff --git a/include/media/DrmPluginPath.h b/include/mediadrm/DrmPluginPath.h
similarity index 100%
rename from include/media/DrmPluginPath.h
rename to include/mediadrm/DrmPluginPath.h


diff --git a/include/media/DrmSessionClientInterface.h b/include/mediadrm/DrmSessionClientInterface.h
similarity index 100%
rename from include/media/DrmSessionClientInterface.h
rename to include/mediadrm/DrmSessionClientInterface.h


diff --git a/include/media/DrmSessionManager.h b/include/mediadrm/DrmSessionManager.h
similarity index 100%
rename from include/media/DrmSessionManager.h
rename to include/mediadrm/DrmSessionManager.h


diff --git a/include/media/ICrypto.h b/include/mediadrm/ICrypto.h
similarity index 100%
rename from include/media/ICrypto.h
rename to include/mediadrm/ICrypto.h


diff --git a/include/media/IDrm.h b/include/mediadrm/IDrm.h
similarity index 100%
rename from include/media/IDrm.h
rename to include/mediadrm/IDrm.h


diff --git a/include/media/IDrmClient.h b/include/mediadrm/IDrmClient.h
similarity index 100%
rename from include/media/IDrmClient.h
rename to include/mediadrm/IDrmClient.h


diff --git a/include/media/IMediaDrmService.h b/include/mediadrm/IMediaDrmService.h
similarity index 100%
rename from include/media/IMediaDrmService.h
rename to include/mediadrm/IMediaDrmService.h


diff --git a/include/mediadrm/OWNERS b/include/mediadrm/OWNERS
new file mode 100644
index 0000000..e788754
--- /dev/null
+++ b/include/mediadrm/OWNERS

@@ -0,0 +1 @@
+jtinker@google.com

diff --git a/include/media/SharedLibrary.h b/include/mediadrm/SharedLibrary.h
similarity index 100%
rename from include/media/SharedLibrary.h
rename to include/mediadrm/SharedLibrary.h


diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index ff440bc..ca119d5 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h

@@ -60,6 +60,8 @@
     volatile int32_t mRear;     // written by producer (output: client, input: server)
     volatile int32_t mFlush;    // incremented by client to indicate a request to flush;
                                 // server notices and discards all data between mFront and mRear
+    volatile int32_t mStop;     // set by client to indicate a stop frame position; server
+                                // will not read beyond this position until start is called.
     volatile uint32_t mUnderrunFrames; // server increments for each unavailable but desired frame
     volatile uint32_t mUnderrunCount;  // server increments for each underrun occurrence
 };
@@ -335,6 +337,8 @@
         mTimestamp.clear();
     }
 
+    virtual void stop() { }; // called by client in AudioTrack::stop()
+
 private:
     // This is a copy of mCblk->mBufferSizeInFrames
     uint32_t   mBufferSizeInFrames;  // effective size of the buffer
@@ -383,8 +387,14 @@
         mPlaybackRateMutator.push(playbackRate);
     }
 
+    // Sends flush and stop position information from the client to the server,
+    // used by streaming AudioTrack flush() or stop().
+    void sendStreamingFlushStop(bool flush);
+
     virtual void flush();
 
+            void stop() override;
+
     virtual uint32_t    getUnderrunFrames() const {
         return mCblk->u.mStreaming.mUnderrunFrames;
     }
@@ -410,6 +420,8 @@
 
     virtual void    flush();
 
+    void stop() override;
+
 #define MIN_LOOP    16  // minimum length of each loop iteration in frames
 
             // setLoop(), setBufferPosition(), and setBufferPositionAndLoop() set the
@@ -438,7 +450,11 @@
         return 0;
     }
 
-    virtual uint32_t    getUnderrunFrames() const {
+    virtual uint32_t getUnderrunFrames() const override {
+        return 0;
+    }
+
+    virtual uint32_t getUnderrunCount() const override {
         return 0;
     }
 
@@ -528,6 +544,10 @@
     //   client will be notified via Futex
     virtual void    flushBufferIfNeeded();
 
+    // Returns the rear position of the AudioTrack shared ring buffer, limited by
+    // the stop frame position level.
+    virtual int32_t getRear() const = 0;
+
     // Total count of the number of flushed frames since creation (never reset).
     virtual int64_t     framesFlushed() const { return mFlushed; }
 
@@ -603,10 +623,18 @@
         return mDrained.load();
     }
 
+    int32_t             getRear() const override;
+
+    // Called on server side track start().
+    virtual void        start();
+
 private:
     AudioPlaybackRate             mPlaybackRate;  // last observed playback rate
     PlaybackRateQueue::Observer   mPlaybackRateObserver;
 
+    // Last client stop-at position when start() was called. Used for streaming AudioTracks.
+    std::atomic<int32_t>          mStopLast{0};
+
     // The server keeps a copy here where it is safe from the client.
     uint32_t                      mUnderrunCount; // echoed to mCblk
     bool                          mUnderrunning;  // used to detect edge of underrun
@@ -630,6 +658,10 @@
     virtual void        tallyUnderrunFrames(uint32_t frameCount);
     virtual uint32_t    getUnderrunFrames() const { return 0; }
 
+    int32_t getRear() const override;
+
+    void start() override { } // ignore for static tracks
+
 private:
     status_t            updateStateWithLoop(StaticAudioTrackState *localState,
                                             const StaticAudioTrackState &update) const;
@@ -657,6 +689,10 @@
             size_t frameSize, bool clientInServer)
         : ServerProxy(cblk, buffers, frameCount, frameSize, false /*isOut*/, clientInServer) { }
 
+    int32_t getRear() const override {
+        return mCblk->u.mStreaming.mRear; // For completeness only; mRear written by server.
+    }
+
 protected:
     virtual ~AudioRecordServerProxy() { }
 };

diff --git a/include/private/media/OWNERS b/include/private/media/OWNERS
new file mode 100644
index 0000000..21723ba
--- /dev/null
+++ b/include/private/media/OWNERS

@@ -0,0 +1,3 @@
+elaurent@google.com
+gkasten@google.com
+hunga@google.com

diff --git a/include/private/media/VideoFrame.h b/include/private/media/VideoFrame.h
index a9d4dd1..712f118 100644
--- a/include/private/media/VideoFrame.h
+++ b/include/private/media/VideoFrame.h

@@ -25,94 +25,34 @@
 
 namespace android {
 
-// Represents a color converted (RGB-based) video frame
-// with bitmap pixels stored in FrameBuffer
+// Represents a color converted (RGB-based) video frame with bitmap
+// pixels stored in FrameBuffer.
+// In a VideoFrame struct stored in IMemory, frame data and ICC data
+// come after the VideoFrame structure. Their locations can be retrieved
+// by getFlattenedData() and getFlattenedIccData();
 class VideoFrame
 {
 public:
     // Construct a VideoFrame object with the specified parameters,
-    // will allocate frame buffer if |allocate| is set to true, will
-    // allocate buffer to hold ICC data if |iccData| and |iccSize|
-    // indicate its presence.
+    // will calculate frame buffer size if |hasData| is set to true.
     VideoFrame(uint32_t width, uint32_t height,
             uint32_t displayWidth, uint32_t displayHeight,
-            uint32_t angle, uint32_t bpp, bool allocate,
-            const void *iccData, size_t iccSize):
+            uint32_t tileWidth, uint32_t tileHeight,
+            uint32_t angle, uint32_t bpp, bool hasData, size_t iccSize):
         mWidth(width), mHeight(height),
         mDisplayWidth(displayWidth), mDisplayHeight(displayHeight),
+        mTileWidth(tileWidth), mTileHeight(tileHeight),
         mRotationAngle(angle), mBytesPerPixel(bpp), mRowBytes(bpp * width),
-        mSize(0), mIccSize(0), mReserved(0), mData(0), mIccData(0) {
-        if (allocate) {
-            mSize = mRowBytes * mHeight;
-            mData = new uint8_t[mSize];
-            if (mData == NULL) {
-                mSize = 0;
-            }
-        }
-
-        if (iccData != NULL && iccSize > 0) {
-            mIccSize = iccSize;
-            mIccData = new uint8_t[iccSize];
-            if (mIccData != NULL) {
-                memcpy(mIccData, iccData, iccSize);
-            } else {
-                mIccSize = 0;
-            }
-        }
+        mSize(hasData ? (bpp * width * height) : 0),
+        mIccSize(iccSize), mReserved(0) {
     }
 
-    // Deep copy of both the information fields and the frame data
-    VideoFrame(const VideoFrame& copy) {
-        copyInfoOnly(copy);
-
-        mSize = copy.mSize;
-        mData = NULL;  // initialize it first
-        if (mSize > 0 && copy.mData != NULL) {
-            mData = new uint8_t[mSize];
-            if (mData != NULL) {
-                memcpy(mData, copy.mData, mSize);
-            } else {
-                mSize = 0;
-            }
-        }
-
-        mIccSize = copy.mIccSize;
-        mIccData = NULL;  // initialize it first
-        if (mIccSize > 0 && copy.mIccData != NULL) {
-            mIccData = new uint8_t[mIccSize];
-            if (mIccData != NULL) {
-                memcpy(mIccData, copy.mIccData, mIccSize);
-            } else {
-                mIccSize = 0;
-            }
-        }
-    }
-
-    ~VideoFrame() {
-        if (mData != 0) {
-            delete[] mData;
-        }
-        if (mIccData != 0) {
-            delete[] mIccData;
-        }
-    }
-
-    // Copy |copy| to a flattened VideoFrame in IMemory, 'this' must point to
-    // a chunk of memory back by IMemory of size at least getFlattenedSize()
-    // of |copy|.
-    void copyFlattened(const VideoFrame& copy) {
-        copyInfoOnly(copy);
-
-        mSize = copy.mSize;
-        mData = NULL;  // initialize it first
-        if (copy.mSize > 0 && copy.mData != NULL) {
-            memcpy(getFlattenedData(), copy.mData, copy.mSize);
-        }
-
-        mIccSize = copy.mIccSize;
-        mIccData = NULL;  // initialize it first
-        if (copy.mIccSize > 0 && copy.mIccData != NULL) {
-            memcpy(getFlattenedIccData(), copy.mIccData, copy.mIccSize);
+    void init(const VideoFrame& copy, const void* iccData, size_t iccSize) {
+        *this = copy;
+        if (mIccSize == iccSize && iccSize > 0 && iccData != NULL) {
+            memcpy(getFlattenedIccData(), iccData, iccSize);
+        } else {
+            mIccSize = 0;
         }
     }
 
@@ -136,38 +76,14 @@
     uint32_t mHeight;          // Decoded image height before rotation
     uint32_t mDisplayWidth;    // Display width before rotation
     uint32_t mDisplayHeight;   // Display height before rotation
+    uint32_t mTileWidth;       // Tile width (0 if image doesn't have grid)
+    uint32_t mTileHeight;      // Tile height (0 if image doesn't have grid)
     int32_t  mRotationAngle;   // Rotation angle, clockwise, should be multiple of 90
     uint32_t mBytesPerPixel;   // Number of bytes per pixel
     uint32_t mRowBytes;        // Number of bytes per row before rotation
-    uint32_t mSize;            // Number of bytes in mData
-    uint32_t mIccSize;         // Number of bytes in mIccData
+    uint32_t mSize;            // Number of bytes of frame data
+    uint32_t mIccSize;         // Number of bytes of ICC data
     uint32_t mReserved;        // (padding to make mData 64-bit aligned)
-
-    // mData should be 64-bit aligned to prevent additional padding
-    uint8_t* mData;            // Actual binary data
-    // pad structure so it's the same size on 64-bit and 32-bit
-    char     mPadding[8 - sizeof(mData)];
-
-    // mIccData should be 64-bit aligned to prevent additional padding
-    uint8_t* mIccData;            // Actual binary data
-    // pad structure so it's the same size on 64-bit and 32-bit
-    char     mIccPadding[8 - sizeof(mIccData)];
-
-private:
-    //
-    // Utility methods used only within VideoFrame struct
-    //
-
-    // Copy the information fields only
-    void copyInfoOnly(const VideoFrame& copy) {
-        mWidth = copy.mWidth;
-        mHeight = copy.mHeight;
-        mDisplayWidth = copy.mDisplayWidth;
-        mDisplayHeight = copy.mDisplayHeight;
-        mRotationAngle = copy.mRotationAngle;
-        mBytesPerPixel = copy.mBytesPerPixel;
-        mRowBytes = copy.mRowBytes;
-    }
 };
 
 }; // namespace android

diff --git a/include/soundtrigger/OWNERS b/include/soundtrigger/OWNERS
new file mode 100644
index 0000000..e83f6b9
--- /dev/null
+++ b/include/soundtrigger/OWNERS

@@ -0,0 +1,2 @@
+elaurent@google.com
+thorntonc@google.com

diff --git a/media/OWNERS b/media/OWNERS
index 1605efd..d49eb8d 100644
--- a/media/OWNERS
+++ b/media/OWNERS

@@ -2,6 +2,7 @@
 dwkang@google.com
 elaurent@google.com
 essick@google.com
+hkuang@google.com
 hunga@google.com
 jmtrivi@google.com
 krocard@google.com

diff --git a/media/audioserver/Android.mk b/media/audioserver/Android.mk
index 3ee7494..70c281a 100644
--- a/media/audioserver/Android.mk
+++ b/media/audioserver/Android.mk

@@ -3,7 +3,8 @@
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES := \
-	main_audioserver.cpp
+	main_audioserver.cpp \
+	../libaudioclient/aidl/android/media/IAudioRecord.aidl
 
 LOCAL_SHARED_LIBRARIES := \
 	libaaudioservice \
@@ -12,11 +13,13 @@
 	libbinder \
 	libcutils \
 	liblog \
+	libhidltransport \
+	libhwbinder \
+	libmedia \
 	libmedialogservice \
 	libnbaio \
 	libsoundtriggerservice \
-	libutils \
-	libhwbinder
+	libutils
 
 # TODO oboeservice is the old folder name for aaudioservice. It will be changed.
 LOCAL_C_INCLUDES := \
@@ -33,9 +36,13 @@
 	frameworks/av/media/libaaudio/include \
 	frameworks/av/media/libaaudio/src \
 	frameworks/av/media/libaaudio/src/binding \
+	frameworks/av/media/libmedia \
 	$(call include-path-for, audio-utils) \
 	external/sonic \
 
+LOCAL_AIDL_INCLUDES := \
+        frameworks/av/media/libaudioclient/aidl
+
 # If AUDIOSERVER_MULTILIB in device.mk is non-empty then it is used to control
 # the LOCAL_MULTILIB for all audioserver exclusive libraries.
 # This is relevant for 64 bit architectures where either or both

diff --git a/media/audioserver/OWNERS b/media/audioserver/OWNERS
new file mode 100644
index 0000000..f9cb567
--- /dev/null
+++ b/media/audioserver/OWNERS

@@ -0,0 +1 @@
+gkasten@google.com

diff --git a/media/audioserver/audioserver.rc b/media/audioserver/audioserver.rc
index 9d42bce..75675a9 100644
--- a/media/audioserver/audioserver.rc
+++ b/media/audioserver/audioserver.rc

@@ -1,10 +1,13 @@
 service audioserver /system/bin/audioserver
-    class main
+    class core
     user audioserver
     # media gid needed for /dev/fm (radio) and for /data/misc/media (tee)
     group audio camera drmrpc inet media mediadrm net_bt net_bt_admin net_bw_acct
     ioprio rt 4
     writepid /dev/cpuset/foreground/tasks /dev/stune/foreground/tasks
+    onrestart restart vendor.audio-hal-2-0
+    # Keep the original service name for backward compatibility when upgrading
+    # O-MR1 devices with framework-only.
     onrestart restart audio-hal-2-0
 
 on property:vts.native_server.on=1

diff --git a/media/audioserver/main_audioserver.cpp b/media/audioserver/main_audioserver.cpp
index 474ef97..db57248 100644
--- a/media/audioserver/main_audioserver.cpp
+++ b/media/audioserver/main_audioserver.cpp

@@ -25,12 +25,9 @@
 #include <binder/IPCThreadState.h>
 #include <binder/ProcessState.h>
 #include <binder/IServiceManager.h>
+#include <hidl/HidlTransportSupport.h>
 #include <utils/Log.h>
 
-// FIXME: remove when BUG 31748996 is fixed
-#include <hwbinder/IPCThreadState.h>
-#include <hwbinder/ProcessState.h>
-
 // from LOCAL_C_INCLUDES
 #include "aaudio/AAudioTesting.h"
 #include "AudioFlinger.h"
@@ -38,12 +35,19 @@
 #include "AAudioService.h"
 #include "utility/AAudioUtilities.h"
 #include "MediaLogService.h"
+#include "MediaUtils.h"
 #include "SoundTriggerHwService.h"
 
 using namespace android;
 
 int main(int argc __unused, char **argv)
 {
+    // TODO: update with refined parameters
+    limitProcessMemory(
+        "audio.maxmem", /* "ro.audio.maxmem", property that defines limit */
+        (size_t)512 * (1 << 20), /* SIZE_MAX, upper limit in bytes */
+        20 /* upper limit as percentage of physical RAM */);
+
     signal(SIGPIPE, SIG_IGN);
 
     bool doLog = (bool) property_get_bool("ro.test_harness", 0);
@@ -128,6 +132,7 @@
             prctl(PR_SET_PDEATHSIG, SIGKILL);   // if parent media.log dies before me, kill me also
             setpgid(0, 0);                      // but if I die first, don't kill my parent
         }
+        android::hardware::configureRpcThreadpool(4, false /*callerWillJoin*/);
         sp<ProcessState> proc(ProcessState::self());
         sp<IServiceManager> sm = defaultServiceManager();
         ALOGI("ServiceManager: %p", sm.get());
@@ -145,10 +150,6 @@
 
         SoundTriggerHwService::instantiate();
         ProcessState::self()->startThreadPool();
-
-// FIXME: remove when BUG 31748996 is fixed
-        android::hardware::ProcessState::self()->startThreadPool();
-
         IPCThreadState::self()->joinThreadPool();
     }
 }

diff --git a/media/common_time/OWNERS b/media/common_time/OWNERS
new file mode 100644
index 0000000..f9cb567
--- /dev/null
+++ b/media/common_time/OWNERS

@@ -0,0 +1 @@
+gkasten@google.com

diff --git a/media/extractors/Android.bp b/media/extractors/Android.bp
new file mode 100644
index 0000000..e8176cf
--- /dev/null
+++ b/media/extractors/Android.bp

@@ -0,0 +1,3 @@
+subdirs = [
+    "*",
+]

diff --git a/media/extractors/aac/AACExtractor.cpp b/media/extractors/aac/AACExtractor.cpp
new file mode 100644
index 0000000..9fc5a76
--- /dev/null
+++ b/media/extractors/aac/AACExtractor.cpp

@@ -0,0 +1,402 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AACExtractor"
+#include <utils/Log.h>
+
+#include "AACExtractor.h"
+#include <media/DataSourceBase.h>
+#include <media/MediaTrack.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/MetaDataUtils.h>
+#include <utils/String8.h>
+
+namespace android {
+
+class AACSource : public MediaTrack {
+public:
+    AACSource(
+            DataSourceBase *source,
+            MetaDataBase &meta,
+            const Vector<uint64_t> &offset_vector,
+            int64_t frame_duration_us);
+
+    virtual status_t start(MetaDataBase *params = NULL);
+    virtual status_t stop();
+
+    virtual status_t getFormat(MetaDataBase&);
+
+    virtual status_t read(
+            MediaBufferBase **buffer, const ReadOptions *options = NULL);
+
+protected:
+    virtual ~AACSource();
+
+private:
+    static const size_t kMaxFrameSize;
+    DataSourceBase *mDataSource;
+    MetaDataBase mMeta;
+
+    off64_t mOffset;
+    int64_t mCurrentTimeUs;
+    bool mStarted;
+    MediaBufferGroup *mGroup;
+
+    Vector<uint64_t> mOffsetVector;
+    int64_t mFrameDurationUs;
+
+    AACSource(const AACSource &);
+    AACSource &operator=(const AACSource &);
+};
+
+////////////////////////////////////////////////////////////////////////////////
+
+// Returns the sample rate based on the sampling frequency index
+uint32_t get_sample_rate(const uint8_t sf_index)
+{
+    static const uint32_t sample_rates[] =
+    {
+        96000, 88200, 64000, 48000, 44100, 32000,
+        24000, 22050, 16000, 12000, 11025, 8000
+    };
+
+    if (sf_index < sizeof(sample_rates) / sizeof(sample_rates[0])) {
+        return sample_rates[sf_index];
+    }
+
+    return 0;
+}
+
+// Returns the frame length in bytes as described in an ADTS header starting at the given offset,
+//     or 0 if the size can't be read due to an error in the header or a read failure.
+// The returned value is the AAC frame size with the ADTS header length (regardless of
+//     the presence of the CRC).
+// If headerSize is non-NULL, it will be used to return the size of the header of this ADTS frame.
+static size_t getAdtsFrameLength(DataSourceBase *source, off64_t offset, size_t* headerSize) {
+
+    const size_t kAdtsHeaderLengthNoCrc = 7;
+    const size_t kAdtsHeaderLengthWithCrc = 9;
+
+    size_t frameSize = 0;
+
+    uint8_t syncword[2];
+    if (source->readAt(offset, &syncword, 2) != 2) {
+        return 0;
+    }
+    if ((syncword[0] != 0xff) || ((syncword[1] & 0xf6) != 0xf0)) {
+        return 0;
+    }
+
+    uint8_t protectionAbsent;
+    if (source->readAt(offset + 1, &protectionAbsent, 1) < 1) {
+        return 0;
+    }
+    protectionAbsent &= 0x1;
+
+    uint8_t header[3];
+    if (source->readAt(offset + 3, &header, 3) < 3) {
+        return 0;
+    }
+
+    frameSize = (header[0] & 0x3) << 11 | header[1] << 3 | header[2] >> 5;
+
+    // protectionAbsent is 0 if there is CRC
+    size_t headSize = protectionAbsent ? kAdtsHeaderLengthNoCrc : kAdtsHeaderLengthWithCrc;
+    if (headSize > frameSize) {
+        return 0;
+    }
+    if (headerSize != NULL) {
+        *headerSize = headSize;
+    }
+
+    return frameSize;
+}
+
+AACExtractor::AACExtractor(
+        DataSourceBase *source, off64_t offset)
+    : mDataSource(source),
+      mInitCheck(NO_INIT),
+      mFrameDurationUs(0) {
+
+    uint8_t profile, sf_index, channel, header[2];
+    if (mDataSource->readAt(offset + 2, &header, 2) < 2) {
+        return;
+    }
+
+    profile = (header[0] >> 6) & 0x3;
+    sf_index = (header[0] >> 2) & 0xf;
+    uint32_t sr = get_sample_rate(sf_index);
+    if (sr == 0) {
+        return;
+    }
+    channel = (header[0] & 0x1) << 2 | (header[1] >> 6);
+
+    MakeAACCodecSpecificData(mMeta, profile, sf_index, channel);
+
+    off64_t streamSize, numFrames = 0;
+    size_t frameSize = 0;
+    int64_t duration = 0;
+
+    if (mDataSource->getSize(&streamSize) == OK) {
+         while (offset < streamSize) {
+            if ((frameSize = getAdtsFrameLength(source, offset, NULL)) == 0) {
+                ALOGW("prematured AAC stream (%lld vs %lld)",
+                        (long long)offset, (long long)streamSize);
+                break;
+            }
+
+            mOffsetVector.push(offset);
+
+            offset += frameSize;
+            numFrames ++;
+        }
+
+        // Round up and get the duration
+        mFrameDurationUs = (1024 * 1000000ll + (sr - 1)) / sr;
+        duration = numFrames * mFrameDurationUs;
+        mMeta.setInt64(kKeyDuration, duration);
+    }
+
+    mInitCheck = OK;
+}
+
+AACExtractor::~AACExtractor() {
+}
+
+status_t AACExtractor::getMetaData(MetaDataBase &meta) {
+    meta.clear();
+    if (mInitCheck == OK) {
+        meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AAC_ADTS);
+    }
+
+    return OK;
+}
+
+size_t AACExtractor::countTracks() {
+    return mInitCheck == OK ? 1 : 0;
+}
+
+MediaTrack *AACExtractor::getTrack(size_t index) {
+    if (mInitCheck != OK || index != 0) {
+        return NULL;
+    }
+
+    return new AACSource(mDataSource, mMeta, mOffsetVector, mFrameDurationUs);
+}
+
+status_t AACExtractor::getTrackMetaData(MetaDataBase &meta, size_t index, uint32_t /* flags */) {
+    if (mInitCheck != OK || index != 0) {
+        return UNKNOWN_ERROR;
+    }
+
+    meta = mMeta;
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+// 8192 = 2^13, 13bit AAC frame size (in bytes)
+const size_t AACSource::kMaxFrameSize = 8192;
+
+AACSource::AACSource(
+        DataSourceBase *source,
+        MetaDataBase &meta,
+        const Vector<uint64_t> &offset_vector,
+        int64_t frame_duration_us)
+    : mDataSource(source),
+      mMeta(meta),
+      mOffset(0),
+      mCurrentTimeUs(0),
+      mStarted(false),
+      mGroup(NULL),
+      mOffsetVector(offset_vector),
+      mFrameDurationUs(frame_duration_us) {
+}
+
+AACSource::~AACSource() {
+    if (mStarted) {
+        stop();
+    }
+}
+
+status_t AACSource::start(MetaDataBase * /* params */) {
+    CHECK(!mStarted);
+
+    if (mOffsetVector.empty()) {
+        mOffset = 0;
+    } else {
+        mOffset = mOffsetVector.itemAt(0);
+    }
+
+    mCurrentTimeUs = 0;
+    mGroup = new MediaBufferGroup;
+    mGroup->add_buffer(MediaBufferBase::Create(kMaxFrameSize));
+    mStarted = true;
+
+    return OK;
+}
+
+status_t AACSource::stop() {
+    CHECK(mStarted);
+
+    delete mGroup;
+    mGroup = NULL;
+
+    mStarted = false;
+    return OK;
+}
+
+status_t AACSource::getFormat(MetaDataBase &meta) {
+    meta = mMeta;
+    return OK;
+}
+
+status_t AACSource::read(
+        MediaBufferBase **out, const ReadOptions *options) {
+    *out = NULL;
+
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode mode;
+    if (options && options->getSeekTo(&seekTimeUs, &mode)) {
+        if (mFrameDurationUs > 0) {
+            int64_t seekFrame = seekTimeUs / mFrameDurationUs;
+            if (seekFrame < 0 || seekFrame >= (int64_t)mOffsetVector.size()) {
+                android_errorWriteLog(0x534e4554, "70239507");
+                return ERROR_MALFORMED;
+            }
+            mCurrentTimeUs = seekFrame * mFrameDurationUs;
+
+            mOffset = mOffsetVector.itemAt(seekFrame);
+        }
+    }
+
+    size_t frameSize, frameSizeWithoutHeader, headerSize;
+    if ((frameSize = getAdtsFrameLength(mDataSource, mOffset, &headerSize)) == 0) {
+        return ERROR_END_OF_STREAM;
+    }
+
+    MediaBufferBase *buffer;
+    status_t err = mGroup->acquire_buffer(&buffer);
+    if (err != OK) {
+        return err;
+    }
+
+    frameSizeWithoutHeader = frameSize - headerSize;
+    if (mDataSource->readAt(mOffset + headerSize, buffer->data(),
+                frameSizeWithoutHeader) != (ssize_t)frameSizeWithoutHeader) {
+        buffer->release();
+        buffer = NULL;
+
+        return ERROR_IO;
+    }
+
+    buffer->set_range(0, frameSizeWithoutHeader);
+    buffer->meta_data().setInt64(kKeyTime, mCurrentTimeUs);
+    buffer->meta_data().setInt32(kKeyIsSyncFrame, 1);
+
+    mOffset += frameSize;
+    mCurrentTimeUs += mFrameDurationUs;
+
+    *out = buffer;
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+static MediaExtractor* CreateExtractor(
+        DataSourceBase *source,
+        void *meta) {
+    off64_t offset = *static_cast<off64_t*>(meta);
+    return new AACExtractor(source, offset);
+}
+
+static MediaExtractor::CreatorFunc Sniff(
+        DataSourceBase *source, float *confidence, void **meta,
+        MediaExtractor::FreeMetaFunc *freeMeta) {
+    off64_t pos = 0;
+
+    for (;;) {
+        uint8_t id3header[10];
+        if (source->readAt(pos, id3header, sizeof(id3header))
+                < (ssize_t)sizeof(id3header)) {
+            return NULL;
+        }
+
+        if (memcmp("ID3", id3header, 3)) {
+            break;
+        }
+
+        // Skip the ID3v2 header.
+
+        size_t len =
+            ((id3header[6] & 0x7f) << 21)
+            | ((id3header[7] & 0x7f) << 14)
+            | ((id3header[8] & 0x7f) << 7)
+            | (id3header[9] & 0x7f);
+
+        len += 10;
+
+        pos += len;
+
+        ALOGV("skipped ID3 tag, new starting offset is %lld (0x%016llx)",
+                (long long)pos, (long long)pos);
+    }
+
+    uint8_t header[2];
+
+    if (source->readAt(pos, &header, 2) != 2) {
+        return NULL;
+    }
+
+    // ADTS syncword
+    if ((header[0] == 0xff) && ((header[1] & 0xf6) == 0xf0)) {
+        *confidence = 0.2;
+
+        off64_t *offPtr = (off64_t*) malloc(sizeof(off64_t));
+        *offPtr = pos;
+        *meta = offPtr;
+        *freeMeta = ::free;
+
+        return CreateExtractor;
+    }
+
+    return NULL;
+}
+
+
+extern "C" {
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
+MediaExtractor::ExtractorDef GETEXTRACTORDEF() {
+    return {
+        MediaExtractor::EXTRACTORDEF_VERSION,
+        UUID("4fd80eae-03d2-4d72-9eb9-48fa6bb54613"),
+        1, // version
+        "AAC Extractor",
+        Sniff
+    };
+}
+
+} // extern "C"
+
+} // namespace android

diff --git a/media/extractors/aac/AACExtractor.h b/media/extractors/aac/AACExtractor.h
new file mode 100644
index 0000000..9dadbed
--- /dev/null
+++ b/media/extractors/aac/AACExtractor.h

@@ -0,0 +1,62 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AAC_EXTRACTOR_H_
+
+#define AAC_EXTRACTOR_H_
+
+#include <media/MediaExtractor.h>
+#include <media/stagefright/MetaDataBase.h>
+
+#include <utils/Vector.h>
+
+namespace android {
+
+struct AMessage;
+class String8;
+
+class AACExtractor : public MediaExtractor {
+public:
+    AACExtractor(DataSourceBase *source, off64_t offset);
+
+    virtual size_t countTracks();
+    virtual MediaTrack *getTrack(size_t index);
+    virtual status_t getTrackMetaData(MetaDataBase& meta, size_t index, uint32_t flags);
+
+    virtual status_t getMetaData(MetaDataBase& meta);
+    virtual const char * name() { return "AACExtractor"; }
+
+protected:
+    virtual ~AACExtractor();
+
+private:
+    DataSourceBase *mDataSource;
+    MetaDataBase mMeta;
+    status_t mInitCheck;
+
+    Vector<uint64_t> mOffsetVector;
+    int64_t mFrameDurationUs;
+
+    AACExtractor(const AACExtractor &);
+    AACExtractor &operator=(const AACExtractor &);
+};
+
+bool SniffAAC(
+        DataSourceBase *source, String8 *mimeType, float *confidence, off64_t *offset);
+
+}  // namespace android
+
+#endif  // AAC_EXTRACTOR_H_

diff --git a/media/extractors/aac/Android.bp b/media/extractors/aac/Android.bp
new file mode 100644
index 0000000..5f05b42
--- /dev/null
+++ b/media/extractors/aac/Android.bp

@@ -0,0 +1,43 @@
+cc_library_shared {
+
+    srcs: ["AACExtractor.cpp"],
+
+    include_dirs: [
+        "frameworks/av/media/libstagefright/",
+    ],
+
+    shared_libs: [
+        "liblog",
+        "libmediaextractor",
+    ],
+
+    static_libs: [
+        "libstagefright_foundation",
+        "libstagefright_metadatautils",
+        "libutils",
+    ],
+
+    name: "libaacextractor",
+    relative_install_path: "extractors",
+
+    compile_multilib: "first",
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+        "-fvisibility=hidden",
+    ],
+    version_script: "exports.lds",
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+        diag: {
+            cfi: true,
+        },
+    },
+
+}

diff --git a/media/libstagefright/codecs/on2/h264dec/MODULE_LICENSE_APACHE2 b/media/extractors/aac/MODULE_LICENSE_APACHE2
similarity index 100%
rename from media/libstagefright/codecs/on2/h264dec/MODULE_LICENSE_APACHE2
rename to media/extractors/aac/MODULE_LICENSE_APACHE2


diff --git a/media/libstagefright/codecs/on2/h264dec/NOTICE b/media/extractors/aac/NOTICE
similarity index 100%
rename from media/libstagefright/codecs/on2/h264dec/NOTICE
rename to media/extractors/aac/NOTICE


diff --git a/media/extractors/aac/exports.lds b/media/extractors/aac/exports.lds
new file mode 100644
index 0000000..b1309ee
--- /dev/null
+++ b/media/extractors/aac/exports.lds

@@ -0,0 +1 @@
+{ global: GETEXTRACTORDEF; local: *; };

diff --git a/media/extractors/amr/AMRExtractor.cpp b/media/extractors/amr/AMRExtractor.cpp
new file mode 100644
index 0000000..f56d5ef
--- /dev/null
+++ b/media/extractors/amr/AMRExtractor.cpp

@@ -0,0 +1,392 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AMRExtractor"
+#include <utils/Log.h>
+
+#include "AMRExtractor.h"
+
+#include <media/DataSourceBase.h>
+#include <media/MediaTrack.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <utils/String8.h>
+
+namespace android {
+
+class AMRSource : public MediaTrack {
+public:
+    AMRSource(
+            DataSourceBase *source,
+            MetaDataBase &meta,
+            bool isWide,
+            const off64_t *offset_table,
+            size_t offset_table_length);
+
+    virtual status_t start(MetaDataBase *params = NULL);
+    virtual status_t stop();
+
+    virtual status_t getFormat(MetaDataBase &);
+
+    virtual status_t read(
+            MediaBufferBase **buffer, const ReadOptions *options = NULL);
+
+protected:
+    virtual ~AMRSource();
+
+private:
+    DataSourceBase *mDataSource;
+    MetaDataBase mMeta;
+    bool mIsWide;
+
+    off64_t mOffset;
+    int64_t mCurrentTimeUs;
+    bool mStarted;
+    MediaBufferGroup *mGroup;
+
+    off64_t mOffsetTable[OFFSET_TABLE_LEN];
+    size_t mOffsetTableLength;
+
+    AMRSource(const AMRSource &);
+    AMRSource &operator=(const AMRSource &);
+};
+
+////////////////////////////////////////////////////////////////////////////////
+
+static size_t getFrameSize(bool isWide, unsigned FT) {
+    static const size_t kFrameSizeNB[16] = {
+        95, 103, 118, 134, 148, 159, 204, 244,
+        39, 43, 38, 37, // SID
+        0, 0, 0, // future use
+        0 // no data
+    };
+    static const size_t kFrameSizeWB[16] = {
+        132, 177, 253, 285, 317, 365, 397, 461, 477,
+        40, // SID
+        0, 0, 0, 0, // future use
+        0, // speech lost
+        0 // no data
+    };
+
+    if (FT > 15 || (isWide && FT > 9 && FT < 14) || (!isWide && FT > 11 && FT < 15)) {
+        ALOGE("illegal AMR frame type %d", FT);
+        return 0;
+    }
+
+    size_t frameSize = isWide ? kFrameSizeWB[FT] : kFrameSizeNB[FT];
+
+    // Round up bits to bytes and add 1 for the header byte.
+    frameSize = (frameSize + 7) / 8 + 1;
+
+    return frameSize;
+}
+
+static status_t getFrameSizeByOffset(DataSourceBase *source,
+        off64_t offset, bool isWide, size_t *frameSize) {
+    uint8_t header;
+    ssize_t count = source->readAt(offset, &header, 1);
+    if (count == 0) {
+        return ERROR_END_OF_STREAM;
+    } else if (count < 0) {
+        return ERROR_IO;
+    }
+
+    unsigned FT = (header >> 3) & 0x0f;
+
+    *frameSize = getFrameSize(isWide, FT);
+    if (*frameSize == 0) {
+        return ERROR_MALFORMED;
+    }
+    return OK;
+}
+
+static bool SniffAMR(
+        DataSourceBase *source, bool *isWide, float *confidence) {
+    char header[9];
+
+    if (source->readAt(0, header, sizeof(header)) != sizeof(header)) {
+        return false;
+    }
+
+    if (!memcmp(header, "#!AMR\n", 6)) {
+        if (isWide != nullptr) {
+            *isWide = false;
+        }
+        *confidence = 0.5;
+
+        return true;
+    } else if (!memcmp(header, "#!AMR-WB\n", 9)) {
+        if (isWide != nullptr) {
+            *isWide = true;
+        }
+        *confidence = 0.5;
+
+        return true;
+    }
+
+    return false;
+}
+
+AMRExtractor::AMRExtractor(DataSourceBase *source)
+    : mDataSource(source),
+      mInitCheck(NO_INIT),
+      mOffsetTableLength(0) {
+    float confidence;
+    if (!SniffAMR(mDataSource, &mIsWide, &confidence)) {
+        return;
+    }
+
+    mMeta.setCString(
+            kKeyMIMEType, mIsWide ? MEDIA_MIMETYPE_AUDIO_AMR_WB
+                                  : MEDIA_MIMETYPE_AUDIO_AMR_NB);
+
+    mMeta.setInt32(kKeyChannelCount, 1);
+    mMeta.setInt32(kKeySampleRate, mIsWide ? 16000 : 8000);
+
+    off64_t offset = mIsWide ? 9 : 6;
+    off64_t streamSize;
+    size_t frameSize, numFrames = 0;
+    int64_t duration = 0;
+
+    if (mDataSource->getSize(&streamSize) == OK) {
+         while (offset < streamSize) {
+             status_t status = getFrameSizeByOffset(source, offset, mIsWide, &frameSize);
+             if (status == ERROR_END_OF_STREAM) {
+                 break;
+             } else if (status != OK) {
+                return;
+            }
+
+            if ((numFrames % 50 == 0) && (numFrames / 50 < OFFSET_TABLE_LEN)) {
+                CHECK_EQ(mOffsetTableLength, numFrames / 50);
+                mOffsetTable[mOffsetTableLength] = offset - (mIsWide ? 9: 6);
+                mOffsetTableLength ++;
+            }
+
+            offset += frameSize;
+            duration += 20000;  // Each frame is 20ms
+            numFrames ++;
+        }
+
+        mMeta.setInt64(kKeyDuration, duration);
+    }
+
+    mInitCheck = OK;
+}
+
+AMRExtractor::~AMRExtractor() {
+}
+
+status_t AMRExtractor::getMetaData(MetaDataBase &meta) {
+    meta.clear();
+
+    if (mInitCheck == OK) {
+        meta.setCString(kKeyMIMEType, mIsWide ? "audio/amr-wb" : "audio/amr");
+    }
+
+    return OK;
+}
+
+size_t AMRExtractor::countTracks() {
+    return mInitCheck == OK ? 1 : 0;
+}
+
+MediaTrack *AMRExtractor::getTrack(size_t index) {
+    if (mInitCheck != OK || index != 0) {
+        return NULL;
+    }
+
+    return new AMRSource(mDataSource, mMeta, mIsWide,
+            mOffsetTable, mOffsetTableLength);
+}
+
+status_t AMRExtractor::getTrackMetaData(MetaDataBase &meta, size_t index, uint32_t /* flags */) {
+    if (mInitCheck != OK || index != 0) {
+        return UNKNOWN_ERROR;
+    }
+
+    meta = mMeta;
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+AMRSource::AMRSource(
+        DataSourceBase *source, MetaDataBase &meta,
+        bool isWide, const off64_t *offset_table, size_t offset_table_length)
+    : mDataSource(source),
+      mMeta(meta),
+      mIsWide(isWide),
+      mOffset(mIsWide ? 9 : 6),
+      mCurrentTimeUs(0),
+      mStarted(false),
+      mGroup(NULL),
+      mOffsetTableLength(offset_table_length) {
+    if (mOffsetTableLength > 0 && mOffsetTableLength <= OFFSET_TABLE_LEN) {
+        memcpy ((char*)mOffsetTable, (char*)offset_table, sizeof(off64_t) * mOffsetTableLength);
+    }
+}
+
+AMRSource::~AMRSource() {
+    if (mStarted) {
+        stop();
+    }
+}
+
+status_t AMRSource::start(MetaDataBase * /* params */) {
+    CHECK(!mStarted);
+
+    mOffset = mIsWide ? 9 : 6;
+    mCurrentTimeUs = 0;
+    mGroup = new MediaBufferGroup;
+    mGroup->add_buffer(MediaBufferBase::Create(128));
+    mStarted = true;
+
+    return OK;
+}
+
+status_t AMRSource::stop() {
+    CHECK(mStarted);
+
+    delete mGroup;
+    mGroup = NULL;
+
+    mStarted = false;
+    return OK;
+}
+
+status_t AMRSource::getFormat(MetaDataBase &meta) {
+    meta = mMeta;
+    return OK;
+}
+
+status_t AMRSource::read(
+        MediaBufferBase **out, const ReadOptions *options) {
+    *out = NULL;
+
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode mode;
+    if (mOffsetTableLength > 0 && options && options->getSeekTo(&seekTimeUs, &mode)) {
+        size_t size;
+        int64_t seekFrame = seekTimeUs / 20000ll;  // 20ms per frame.
+        mCurrentTimeUs = seekFrame * 20000ll;
+
+        size_t index = seekFrame < 0 ? 0 : seekFrame / 50;
+        if (index >= mOffsetTableLength) {
+            index = mOffsetTableLength - 1;
+        }
+
+        mOffset = mOffsetTable[index] + (mIsWide ? 9 : 6);
+
+        for (size_t i = 0; i< seekFrame - index * 50; i++) {
+            status_t err;
+            if ((err = getFrameSizeByOffset(mDataSource, mOffset,
+                            mIsWide, &size)) != OK) {
+                return err;
+            }
+            mOffset += size;
+        }
+    }
+
+    uint8_t header;
+    ssize_t n = mDataSource->readAt(mOffset, &header, 1);
+
+    if (n < 1) {
+        return ERROR_END_OF_STREAM;
+    }
+
+    if (header & 0x83) {
+        // Padding bits must be 0.
+
+        ALOGE("padding bits must be 0, header is 0x%02x", header);
+
+        return ERROR_MALFORMED;
+    }
+
+    unsigned FT = (header >> 3) & 0x0f;
+
+    size_t frameSize = getFrameSize(mIsWide, FT);
+    if (frameSize == 0) {
+        return ERROR_MALFORMED;
+    }
+
+    MediaBufferBase *buffer;
+    status_t err = mGroup->acquire_buffer(&buffer);
+    if (err != OK) {
+        return err;
+    }
+
+    n = mDataSource->readAt(mOffset, buffer->data(), frameSize);
+
+    if (n != (ssize_t)frameSize) {
+        buffer->release();
+        buffer = NULL;
+
+        if (n < 0) {
+            return ERROR_IO;
+        } else {
+            // only partial frame is available, treat it as EOS.
+            mOffset += n;
+            return ERROR_END_OF_STREAM;
+        }
+    }
+
+    buffer->set_range(0, frameSize);
+    buffer->meta_data().setInt64(kKeyTime, mCurrentTimeUs);
+    buffer->meta_data().setInt32(kKeyIsSyncFrame, 1);
+
+    mOffset += frameSize;
+    mCurrentTimeUs += 20000;  // Each frame is 20ms
+
+    *out = buffer;
+
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+extern "C" {
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
+MediaExtractor::ExtractorDef GETEXTRACTORDEF() {
+    return {
+        MediaExtractor::EXTRACTORDEF_VERSION,
+        UUID("c86639c9-2f31-40ac-a715-fa01b4493aaf"),
+        1,
+        "AMR Extractor",
+        [](
+                DataSourceBase *source,
+                float *confidence,
+                void **,
+                MediaExtractor::FreeMetaFunc *) -> MediaExtractor::CreatorFunc {
+            if (SniffAMR(source, nullptr, confidence)) {
+                return [](
+                        DataSourceBase *source,
+                        void *) -> MediaExtractor* {
+                    return new AMRExtractor(source);};
+            }
+            return NULL;
+        }
+    };
+}
+
+} // extern "C"
+
+}  // namespace android

diff --git a/media/extractors/amr/AMRExtractor.h b/media/extractors/amr/AMRExtractor.h
new file mode 100644
index 0000000..c90b325
--- /dev/null
+++ b/media/extractors/amr/AMRExtractor.h

@@ -0,0 +1,60 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AMR_EXTRACTOR_H_
+
+#define AMR_EXTRACTOR_H_
+
+#include <utils/Errors.h>
+#include <media/MediaExtractor.h>
+#include <media/stagefright/MetaDataBase.h>
+
+namespace android {
+
+struct AMessage;
+class String8;
+#define OFFSET_TABLE_LEN    300
+
+class AMRExtractor : public MediaExtractor {
+public:
+    explicit AMRExtractor(DataSourceBase *source);
+
+    virtual size_t countTracks();
+    virtual MediaTrack *getTrack(size_t index);
+    virtual status_t getTrackMetaData(MetaDataBase& meta, size_t index, uint32_t flags);
+
+    virtual status_t getMetaData(MetaDataBase& meta);
+    virtual const char * name() { return "AMRExtractor"; }
+
+protected:
+    virtual ~AMRExtractor();
+
+private:
+    DataSourceBase *mDataSource;
+    MetaDataBase mMeta;
+    status_t mInitCheck;
+    bool mIsWide;
+
+    off64_t mOffsetTable[OFFSET_TABLE_LEN]; //5 min
+    size_t mOffsetTableLength;
+
+    AMRExtractor(const AMRExtractor &);
+    AMRExtractor &operator=(const AMRExtractor &);
+};
+
+}  // namespace android
+
+#endif  // AMR_EXTRACTOR_H_

diff --git a/media/extractors/amr/Android.bp b/media/extractors/amr/Android.bp
new file mode 100644
index 0000000..d962b93
--- /dev/null
+++ b/media/extractors/amr/Android.bp

@@ -0,0 +1,41 @@
+cc_library_shared {
+
+    srcs: ["AMRExtractor.cpp"],
+
+    include_dirs: [
+        "frameworks/av/media/libstagefright/include",
+    ],
+
+    shared_libs: [
+        "liblog",
+        "libmediaextractor",
+    ],
+
+    static_libs: [
+        "libstagefright_foundation",
+    ],
+
+    name: "libamrextractor",
+    relative_install_path: "extractors",
+
+    compile_multilib: "first",
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+        "-fvisibility=hidden",
+    ],
+    version_script: "exports.lds",
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+        diag: {
+            cfi: true,
+        },
+    },
+
+}

diff --git a/media/libstagefright/matroska/MODULE_LICENSE_APACHE2 b/media/extractors/amr/MODULE_LICENSE_APACHE2
similarity index 100%
copy from media/libstagefright/matroska/MODULE_LICENSE_APACHE2
copy to media/extractors/amr/MODULE_LICENSE_APACHE2


diff --git a/media/libstagefright/matroska/NOTICE b/media/extractors/amr/NOTICE
similarity index 100%
copy from media/libstagefright/matroska/NOTICE
copy to media/extractors/amr/NOTICE


diff --git a/media/extractors/amr/exports.lds b/media/extractors/amr/exports.lds
new file mode 100644
index 0000000..b1309ee
--- /dev/null
+++ b/media/extractors/amr/exports.lds

@@ -0,0 +1 @@
+{ global: GETEXTRACTORDEF; local: *; };

diff --git a/media/extractors/flac/Android.bp b/media/extractors/flac/Android.bp
new file mode 100644
index 0000000..6282793
--- /dev/null
+++ b/media/extractors/flac/Android.bp

@@ -0,0 +1,43 @@
+cc_library_shared {
+
+    srcs: ["FLACExtractor.cpp"],
+
+    include_dirs: [
+        "frameworks/av/media/libstagefright/include",
+        "external/flac/include",
+    ],
+
+    shared_libs: [
+        "liblog",
+        "libmediaextractor",
+    ],
+
+    static_libs: [
+        "libFLAC",
+        "libstagefright_foundation",
+    ],
+
+    name: "libflacextractor",
+    relative_install_path: "extractors",
+
+    compile_multilib: "first",
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+        "-fvisibility=hidden",
+    ],
+    version_script: "exports.lds",
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+        diag: {
+            cfi: true,
+        },
+    },
+
+}

diff --git a/media/extractors/flac/FLACExtractor.cpp b/media/extractors/flac/FLACExtractor.cpp
new file mode 100644
index 0000000..e3da259
--- /dev/null
+++ b/media/extractors/flac/FLACExtractor.cpp

@@ -0,0 +1,884 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "FLACExtractor"
+#include <utils/Log.h>
+
+#include "FLACExtractor.h"
+// libFLAC parser
+#include "FLAC/stream_decoder.h"
+
+#include <media/DataSourceBase.h>
+#include <media/MediaTrack.h>
+#include <media/VorbisComment.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/base64.h>
+#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/MediaBufferBase.h>
+
+namespace android {
+
+class FLACParser;
+
+class FLACSource : public MediaTrack {
+
+public:
+    FLACSource(
+            DataSourceBase *dataSource,
+            MetaDataBase &meta);
+
+    virtual status_t start(MetaDataBase *params);
+    virtual status_t stop();
+    virtual status_t getFormat(MetaDataBase &meta);
+
+    virtual status_t read(
+            MediaBufferBase **buffer, const ReadOptions *options = NULL);
+
+protected:
+    virtual ~FLACSource();
+
+private:
+    DataSourceBase *mDataSource;
+    MetaDataBase mTrackMetadata;
+    FLACParser *mParser;
+    bool mInitCheck;
+    bool mStarted;
+
+    // no copy constructor or assignment
+    FLACSource(const FLACSource &);
+    FLACSource &operator=(const FLACSource &);
+
+};
+
+// FLACParser wraps a C libFLAC parser aka stream decoder
+
+class FLACParser {
+
+public:
+    enum {
+        kMaxChannels = 8,
+    };
+
+    explicit FLACParser(
+        DataSourceBase *dataSource,
+        // If metadata pointers aren't provided, we don't fill them
+        MetaDataBase *fileMetadata = 0,
+        MetaDataBase *trackMetadata = 0);
+
+    virtual ~FLACParser();
+
+    status_t initCheck() const {
+        return mInitCheck;
+    }
+
+    // stream properties
+    unsigned getMaxBlockSize() const {
+        return mStreamInfo.max_blocksize;
+    }
+    unsigned getSampleRate() const {
+        return mStreamInfo.sample_rate;
+    }
+    unsigned getChannels() const {
+        return mStreamInfo.channels;
+    }
+    unsigned getBitsPerSample() const {
+        return mStreamInfo.bits_per_sample;
+    }
+    FLAC__uint64 getTotalSamples() const {
+        return mStreamInfo.total_samples;
+    }
+
+    // media buffers
+    void allocateBuffers();
+    void releaseBuffers();
+    MediaBufferBase *readBuffer() {
+        return readBuffer(false, 0LL);
+    }
+    MediaBufferBase *readBuffer(FLAC__uint64 sample) {
+        return readBuffer(true, sample);
+    }
+
+private:
+    DataSourceBase *mDataSource;
+    MetaDataBase *mFileMetadata;
+    MetaDataBase *mTrackMetadata;
+    bool mInitCheck;
+
+    // media buffers
+    size_t mMaxBufferSize;
+    MediaBufferGroup *mGroup;
+    void (*mCopy)(short *dst, const int * src[kMaxChannels], unsigned nSamples, unsigned nChannels);
+
+    // handle to underlying libFLAC parser
+    FLAC__StreamDecoder *mDecoder;
+
+    // current position within the data source
+    off64_t mCurrentPos;
+    bool mEOF;
+
+    // cached when the STREAMINFO metadata is parsed by libFLAC
+    FLAC__StreamMetadata_StreamInfo mStreamInfo;
+    bool mStreamInfoValid;
+
+    // cached when a decoded PCM block is "written" by libFLAC parser
+    bool mWriteRequested;
+    bool mWriteCompleted;
+    FLAC__FrameHeader mWriteHeader;
+    FLAC__int32 const * mWriteBuffer[kMaxChannels];
+
+    // most recent error reported by libFLAC parser
+    FLAC__StreamDecoderErrorStatus mErrorStatus;
+
+    status_t init();
+    MediaBufferBase *readBuffer(bool doSeek, FLAC__uint64 sample);
+
+    // no copy constructor or assignment
+    FLACParser(const FLACParser &);
+    FLACParser &operator=(const FLACParser &);
+
+    // FLAC parser callbacks as C++ instance methods
+    FLAC__StreamDecoderReadStatus readCallback(
+            FLAC__byte buffer[], size_t *bytes);
+    FLAC__StreamDecoderSeekStatus seekCallback(
+            FLAC__uint64 absolute_byte_offset);
+    FLAC__StreamDecoderTellStatus tellCallback(
+            FLAC__uint64 *absolute_byte_offset);
+    FLAC__StreamDecoderLengthStatus lengthCallback(
+            FLAC__uint64 *stream_length);
+    FLAC__bool eofCallback();
+    FLAC__StreamDecoderWriteStatus writeCallback(
+            const FLAC__Frame *frame, const FLAC__int32 * const buffer[]);
+    void metadataCallback(const FLAC__StreamMetadata *metadata);
+    void errorCallback(FLAC__StreamDecoderErrorStatus status);
+
+    // FLAC parser callbacks as C-callable functions
+    static FLAC__StreamDecoderReadStatus read_callback(
+            const FLAC__StreamDecoder *decoder,
+            FLAC__byte buffer[], size_t *bytes,
+            void *client_data);
+    static FLAC__StreamDecoderSeekStatus seek_callback(
+            const FLAC__StreamDecoder *decoder,
+            FLAC__uint64 absolute_byte_offset,
+            void *client_data);
+    static FLAC__StreamDecoderTellStatus tell_callback(
+            const FLAC__StreamDecoder *decoder,
+            FLAC__uint64 *absolute_byte_offset,
+            void *client_data);
+    static FLAC__StreamDecoderLengthStatus length_callback(
+            const FLAC__StreamDecoder *decoder,
+            FLAC__uint64 *stream_length,
+            void *client_data);
+    static FLAC__bool eof_callback(
+            const FLAC__StreamDecoder *decoder,
+            void *client_data);
+    static FLAC__StreamDecoderWriteStatus write_callback(
+            const FLAC__StreamDecoder *decoder,
+            const FLAC__Frame *frame, const FLAC__int32 * const buffer[],
+            void *client_data);
+    static void metadata_callback(
+            const FLAC__StreamDecoder *decoder,
+            const FLAC__StreamMetadata *metadata,
+            void *client_data);
+    static void error_callback(
+            const FLAC__StreamDecoder *decoder,
+            FLAC__StreamDecoderErrorStatus status,
+            void *client_data);
+
+};
+
+// The FLAC parser calls our C++ static callbacks using C calling conventions,
+// inside FLAC__stream_decoder_process_until_end_of_metadata
+// and FLAC__stream_decoder_process_single.
+// We immediately then call our corresponding C++ instance methods
+// with the same parameter list, but discard redundant information.
+
+FLAC__StreamDecoderReadStatus FLACParser::read_callback(
+        const FLAC__StreamDecoder * /* decoder */, FLAC__byte buffer[],
+        size_t *bytes, void *client_data)
+{
+    return ((FLACParser *) client_data)->readCallback(buffer, bytes);
+}
+
+FLAC__StreamDecoderSeekStatus FLACParser::seek_callback(
+        const FLAC__StreamDecoder * /* decoder */,
+        FLAC__uint64 absolute_byte_offset, void *client_data)
+{
+    return ((FLACParser *) client_data)->seekCallback(absolute_byte_offset);
+}
+
+FLAC__StreamDecoderTellStatus FLACParser::tell_callback(
+        const FLAC__StreamDecoder * /* decoder */,
+        FLAC__uint64 *absolute_byte_offset, void *client_data)
+{
+    return ((FLACParser *) client_data)->tellCallback(absolute_byte_offset);
+}
+
+FLAC__StreamDecoderLengthStatus FLACParser::length_callback(
+        const FLAC__StreamDecoder * /* decoder */,
+        FLAC__uint64 *stream_length, void *client_data)
+{
+    return ((FLACParser *) client_data)->lengthCallback(stream_length);
+}
+
+FLAC__bool FLACParser::eof_callback(
+        const FLAC__StreamDecoder * /* decoder */, void *client_data)
+{
+    return ((FLACParser *) client_data)->eofCallback();
+}
+
+FLAC__StreamDecoderWriteStatus FLACParser::write_callback(
+        const FLAC__StreamDecoder * /* decoder */, const FLAC__Frame *frame,
+        const FLAC__int32 * const buffer[], void *client_data)
+{
+    return ((FLACParser *) client_data)->writeCallback(frame, buffer);
+}
+
+void FLACParser::metadata_callback(
+        const FLAC__StreamDecoder * /* decoder */,
+        const FLAC__StreamMetadata *metadata, void *client_data)
+{
+    ((FLACParser *) client_data)->metadataCallback(metadata);
+}
+
+void FLACParser::error_callback(
+        const FLAC__StreamDecoder * /* decoder */,
+        FLAC__StreamDecoderErrorStatus status, void *client_data)
+{
+    ((FLACParser *) client_data)->errorCallback(status);
+}
+
+// These are the corresponding callbacks with C++ calling conventions
+
+FLAC__StreamDecoderReadStatus FLACParser::readCallback(
+        FLAC__byte buffer[], size_t *bytes)
+{
+    size_t requested = *bytes;
+    ssize_t actual = mDataSource->readAt(mCurrentPos, buffer, requested);
+    if (0 > actual) {
+        *bytes = 0;
+        return FLAC__STREAM_DECODER_READ_STATUS_ABORT;
+    } else if (0 == actual) {
+        *bytes = 0;
+        mEOF = true;
+        return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
+    } else {
+        assert(actual <= requested);
+        *bytes = actual;
+        mCurrentPos += actual;
+        return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
+    }
+}
+
+FLAC__StreamDecoderSeekStatus FLACParser::seekCallback(
+        FLAC__uint64 absolute_byte_offset)
+{
+    mCurrentPos = absolute_byte_offset;
+    mEOF = false;
+    return FLAC__STREAM_DECODER_SEEK_STATUS_OK;
+}
+
+FLAC__StreamDecoderTellStatus FLACParser::tellCallback(
+        FLAC__uint64 *absolute_byte_offset)
+{
+    *absolute_byte_offset = mCurrentPos;
+    return FLAC__STREAM_DECODER_TELL_STATUS_OK;
+}
+
+FLAC__StreamDecoderLengthStatus FLACParser::lengthCallback(
+        FLAC__uint64 *stream_length)
+{
+    off64_t size;
+    if (OK == mDataSource->getSize(&size)) {
+        *stream_length = size;
+        return FLAC__STREAM_DECODER_LENGTH_STATUS_OK;
+    } else {
+        return FLAC__STREAM_DECODER_LENGTH_STATUS_UNSUPPORTED;
+    }
+}
+
+FLAC__bool FLACParser::eofCallback()
+{
+    return mEOF;
+}
+
+FLAC__StreamDecoderWriteStatus FLACParser::writeCallback(
+        const FLAC__Frame *frame, const FLAC__int32 * const buffer[])
+{
+    if (mWriteRequested) {
+        mWriteRequested = false;
+        // FLAC parser doesn't free or realloc buffer until next frame or finish
+        mWriteHeader = frame->header;
+        memmove(mWriteBuffer, buffer, sizeof(const FLAC__int32 * const) * getChannels());
+        mWriteCompleted = true;
+        return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+    } else {
+        ALOGE("FLACParser::writeCallback unexpected");
+        return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
+    }
+}
+
+void FLACParser::metadataCallback(const FLAC__StreamMetadata *metadata)
+{
+    switch (metadata->type) {
+    case FLAC__METADATA_TYPE_STREAMINFO:
+        if (!mStreamInfoValid) {
+            mStreamInfo = metadata->data.stream_info;
+            mStreamInfoValid = true;
+        } else {
+            ALOGE("FLACParser::metadataCallback unexpected STREAMINFO");
+        }
+        break;
+    case FLAC__METADATA_TYPE_VORBIS_COMMENT:
+        {
+        const FLAC__StreamMetadata_VorbisComment *vc;
+        vc = &metadata->data.vorbis_comment;
+        for (FLAC__uint32 i = 0; i < vc->num_comments; ++i) {
+            FLAC__StreamMetadata_VorbisComment_Entry *vce;
+            vce = &vc->comments[i];
+            if (mFileMetadata != 0 && vce->entry != NULL) {
+                parseVorbisComment(mFileMetadata, (const char *) vce->entry,
+                        vce->length);
+            }
+        }
+        }
+        break;
+    case FLAC__METADATA_TYPE_PICTURE:
+        if (mFileMetadata != 0) {
+            const FLAC__StreamMetadata_Picture *p = &metadata->data.picture;
+            mFileMetadata->setData(kKeyAlbumArt,
+                    MetaData::TYPE_NONE, p->data, p->data_length);
+            mFileMetadata->setCString(kKeyAlbumArtMIME, p->mime_type);
+        }
+        break;
+    default:
+        ALOGW("FLACParser::metadataCallback unexpected type %u", metadata->type);
+        break;
+    }
+}
+
+void FLACParser::errorCallback(FLAC__StreamDecoderErrorStatus status)
+{
+    ALOGE("FLACParser::errorCallback status=%d", status);
+    mErrorStatus = status;
+}
+
+// Copy samples from FLAC native 32-bit non-interleaved to 16-bit interleaved.
+// These are candidates for optimization if needed.
+
+static void copyMono8(
+        short *dst,
+        const int * src[FLACParser::kMaxChannels],
+        unsigned nSamples,
+        unsigned /* nChannels */) {
+    for (unsigned i = 0; i < nSamples; ++i) {
+        *dst++ = src[0][i] << 8;
+    }
+}
+
+static void copyStereo8(
+        short *dst,
+        const int * src[FLACParser::kMaxChannels],
+        unsigned nSamples,
+        unsigned /* nChannels */) {
+    for (unsigned i = 0; i < nSamples; ++i) {
+        *dst++ = src[0][i] << 8;
+        *dst++ = src[1][i] << 8;
+    }
+}
+
+static void copyMultiCh8(short *dst, const int * src[FLACParser::kMaxChannels], unsigned nSamples, unsigned nChannels)
+{
+    for (unsigned i = 0; i < nSamples; ++i) {
+        for (unsigned c = 0; c < nChannels; ++c) {
+            *dst++ = src[c][i] << 8;
+        }
+    }
+}
+
+static void copyMono16(
+        short *dst,
+        const int * src[FLACParser::kMaxChannels],
+        unsigned nSamples,
+        unsigned /* nChannels */) {
+    for (unsigned i = 0; i < nSamples; ++i) {
+        *dst++ = src[0][i];
+    }
+}
+
+static void copyStereo16(
+        short *dst,
+        const int * src[FLACParser::kMaxChannels],
+        unsigned nSamples,
+        unsigned /* nChannels */) {
+    for (unsigned i = 0; i < nSamples; ++i) {
+        *dst++ = src[0][i];
+        *dst++ = src[1][i];
+    }
+}
+
+static void copyMultiCh16(short *dst, const int * src[FLACParser::kMaxChannels], unsigned nSamples, unsigned nChannels)
+{
+    for (unsigned i = 0; i < nSamples; ++i) {
+        for (unsigned c = 0; c < nChannels; ++c) {
+            *dst++ = src[c][i];
+        }
+    }
+}
+
+// 24-bit versions should do dithering or noise-shaping, here or in AudioFlinger
+
+static void copyMono24(
+        short *dst,
+        const int * src[FLACParser::kMaxChannels],
+        unsigned nSamples,
+        unsigned /* nChannels */) {
+    for (unsigned i = 0; i < nSamples; ++i) {
+        *dst++ = src[0][i] >> 8;
+    }
+}
+
+static void copyStereo24(
+        short *dst,
+        const int * src[FLACParser::kMaxChannels],
+        unsigned nSamples,
+        unsigned /* nChannels */) {
+    for (unsigned i = 0; i < nSamples; ++i) {
+        *dst++ = src[0][i] >> 8;
+        *dst++ = src[1][i] >> 8;
+    }
+}
+
+static void copyMultiCh24(short *dst, const int * src[FLACParser::kMaxChannels], unsigned nSamples, unsigned nChannels)
+{
+    for (unsigned i = 0; i < nSamples; ++i) {
+        for (unsigned c = 0; c < nChannels; ++c) {
+            *dst++ = src[c][i] >> 8;
+        }
+    }
+}
+
+static void copyTrespass(
+        short * /* dst */,
+        const int *[FLACParser::kMaxChannels] /* src */,
+        unsigned /* nSamples */,
+        unsigned /* nChannels */) {
+    TRESPASS();
+}
+
+// FLACParser
+
+FLACParser::FLACParser(
+        DataSourceBase *dataSource,
+        MetaDataBase *fileMetadata,
+        MetaDataBase *trackMetadata)
+    : mDataSource(dataSource),
+      mFileMetadata(fileMetadata),
+      mTrackMetadata(trackMetadata),
+      mInitCheck(false),
+      mMaxBufferSize(0),
+      mGroup(NULL),
+      mCopy(copyTrespass),
+      mDecoder(NULL),
+      mCurrentPos(0LL),
+      mEOF(false),
+      mStreamInfoValid(false),
+      mWriteRequested(false),
+      mWriteCompleted(false),
+      mErrorStatus((FLAC__StreamDecoderErrorStatus) -1)
+{
+    ALOGV("FLACParser::FLACParser");
+    memset(&mStreamInfo, 0, sizeof(mStreamInfo));
+    memset(&mWriteHeader, 0, sizeof(mWriteHeader));
+    mInitCheck = init();
+}
+
+FLACParser::~FLACParser()
+{
+    ALOGV("FLACParser::~FLACParser");
+    if (mDecoder != NULL) {
+        FLAC__stream_decoder_delete(mDecoder);
+        mDecoder = NULL;
+    }
+}
+
+status_t FLACParser::init()
+{
+    // setup libFLAC parser
+    mDecoder = FLAC__stream_decoder_new();
+    if (mDecoder == NULL) {
+        // The new should succeed, since probably all it does is a malloc
+        // that always succeeds in Android.  But to avoid dependence on the
+        // libFLAC internals, we check and log here.
+        ALOGE("new failed");
+        return NO_INIT;
+    }
+    FLAC__stream_decoder_set_md5_checking(mDecoder, false);
+    FLAC__stream_decoder_set_metadata_ignore_all(mDecoder);
+    FLAC__stream_decoder_set_metadata_respond(
+            mDecoder, FLAC__METADATA_TYPE_STREAMINFO);
+    FLAC__stream_decoder_set_metadata_respond(
+            mDecoder, FLAC__METADATA_TYPE_PICTURE);
+    FLAC__stream_decoder_set_metadata_respond(
+            mDecoder, FLAC__METADATA_TYPE_VORBIS_COMMENT);
+    FLAC__StreamDecoderInitStatus initStatus;
+    initStatus = FLAC__stream_decoder_init_stream(
+            mDecoder,
+            read_callback, seek_callback, tell_callback,
+            length_callback, eof_callback, write_callback,
+            metadata_callback, error_callback, (void *) this);
+    if (initStatus != FLAC__STREAM_DECODER_INIT_STATUS_OK) {
+        // A failure here probably indicates a programming error and so is
+        // unlikely to happen. But we check and log here similarly to above.
+        ALOGE("init_stream failed %d", initStatus);
+        return NO_INIT;
+    }
+    // parse all metadata
+    if (!FLAC__stream_decoder_process_until_end_of_metadata(mDecoder)) {
+        ALOGE("end_of_metadata failed");
+        return NO_INIT;
+    }
+    if (mStreamInfoValid) {
+        // check channel count
+        if (getChannels() == 0 || getChannels() > kMaxChannels) {
+            ALOGE("unsupported channel count %u", getChannels());
+            return NO_INIT;
+        }
+        // check bit depth
+        switch (getBitsPerSample()) {
+        case 8:
+        case 16:
+        case 24:
+            break;
+        default:
+            ALOGE("unsupported bits per sample %u", getBitsPerSample());
+            return NO_INIT;
+        }
+        // check sample rate
+        switch (getSampleRate()) {
+        case  8000:
+        case 11025:
+        case 12000:
+        case 16000:
+        case 22050:
+        case 24000:
+        case 32000:
+        case 44100:
+        case 48000:
+        case 88200:
+        case 96000:
+            break;
+        default:
+            ALOGE("unsupported sample rate %u", getSampleRate());
+            return NO_INIT;
+        }
+        // configure the appropriate copy function, defaulting to trespass
+        static const struct {
+            unsigned mChannels;
+            unsigned mBitsPerSample;
+            void (*mCopy)(short *dst, const int * src[kMaxChannels], unsigned nSamples, unsigned nChannels);
+        } table[] = {
+            { 1,  8, copyMono8    },
+            { 2,  8, copyStereo8  },
+            { 8,  8, copyMultiCh8  },
+            { 1, 16, copyMono16   },
+            { 2, 16, copyStereo16 },
+            { 8, 16, copyMultiCh16 },
+            { 1, 24, copyMono24   },
+            { 2, 24, copyStereo24 },
+            { 8, 24, copyMultiCh24 },
+        };
+        for (unsigned i = 0; i < sizeof(table)/sizeof(table[0]); ++i) {
+            if (table[i].mChannels >= getChannels() &&
+                    table[i].mBitsPerSample == getBitsPerSample()) {
+                mCopy = table[i].mCopy;
+                break;
+            }
+        }
+        // populate track metadata
+        if (mTrackMetadata != 0) {
+            mTrackMetadata->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
+            mTrackMetadata->setInt32(kKeyChannelCount, getChannels());
+            mTrackMetadata->setInt32(kKeySampleRate, getSampleRate());
+            mTrackMetadata->setInt32(kKeyPcmEncoding, kAudioEncodingPcm16bit);
+            // sample rate is non-zero, so division by zero not possible
+            mTrackMetadata->setInt64(kKeyDuration,
+                    (getTotalSamples() * 1000000LL) / getSampleRate());
+        }
+    } else {
+        ALOGE("missing STREAMINFO");
+        return NO_INIT;
+    }
+    if (mFileMetadata != 0) {
+        mFileMetadata->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_FLAC);
+    }
+    return OK;
+}
+
+void FLACParser::allocateBuffers()
+{
+    CHECK(mGroup == NULL);
+    mGroup = new MediaBufferGroup;
+    mMaxBufferSize = getMaxBlockSize() * getChannels() * sizeof(short);
+    mGroup->add_buffer(MediaBufferBase::Create(mMaxBufferSize));
+}
+
+void FLACParser::releaseBuffers()
+{
+    CHECK(mGroup != NULL);
+    delete mGroup;
+    mGroup = NULL;
+}
+
+MediaBufferBase *FLACParser::readBuffer(bool doSeek, FLAC__uint64 sample)
+{
+    mWriteRequested = true;
+    mWriteCompleted = false;
+    if (doSeek) {
+        // We implement the seek callback, so this works without explicit flush
+        if (!FLAC__stream_decoder_seek_absolute(mDecoder, sample)) {
+            ALOGE("FLACParser::readBuffer seek to sample %lld failed", (long long)sample);
+            return NULL;
+        }
+        ALOGV("FLACParser::readBuffer seek to sample %lld succeeded", (long long)sample);
+    } else {
+        if (!FLAC__stream_decoder_process_single(mDecoder)) {
+            ALOGE("FLACParser::readBuffer process_single failed");
+            return NULL;
+        }
+    }
+    if (!mWriteCompleted) {
+        ALOGV("FLACParser::readBuffer write did not complete");
+        return NULL;
+    }
+    // verify that block header keeps the promises made by STREAMINFO
+    unsigned blocksize = mWriteHeader.blocksize;
+    if (blocksize == 0 || blocksize > getMaxBlockSize()) {
+        ALOGE("FLACParser::readBuffer write invalid blocksize %u", blocksize);
+        return NULL;
+    }
+    if (mWriteHeader.sample_rate != getSampleRate() ||
+        mWriteHeader.channels != getChannels() ||
+        mWriteHeader.bits_per_sample != getBitsPerSample()) {
+        ALOGE("FLACParser::readBuffer write changed parameters mid-stream: %d/%d/%d -> %d/%d/%d",
+                getSampleRate(), getChannels(), getBitsPerSample(),
+                mWriteHeader.sample_rate, mWriteHeader.channels, mWriteHeader.bits_per_sample);
+        return NULL;
+    }
+    // acquire a media buffer
+    CHECK(mGroup != NULL);
+    MediaBufferBase *buffer;
+    status_t err = mGroup->acquire_buffer(&buffer);
+    if (err != OK) {
+        return NULL;
+    }
+    size_t bufferSize = blocksize * getChannels() * sizeof(short);
+    CHECK(bufferSize <= mMaxBufferSize);
+    short *data = (short *) buffer->data();
+    buffer->set_range(0, bufferSize);
+    // copy PCM from FLAC write buffer to our media buffer, with interleaving
+    (*mCopy)(data, mWriteBuffer, blocksize, getChannels());
+    // fill in buffer metadata
+    CHECK(mWriteHeader.number_type == FLAC__FRAME_NUMBER_TYPE_SAMPLE_NUMBER);
+    FLAC__uint64 sampleNumber = mWriteHeader.number.sample_number;
+    int64_t timeUs = (1000000LL * sampleNumber) / getSampleRate();
+    buffer->meta_data().setInt64(kKeyTime, timeUs);
+    buffer->meta_data().setInt32(kKeyIsSyncFrame, 1);
+    return buffer;
+}
+
+// FLACsource
+
+FLACSource::FLACSource(
+        DataSourceBase *dataSource,
+        MetaDataBase &trackMetadata)
+    : mDataSource(dataSource),
+      mTrackMetadata(trackMetadata),
+      mParser(0),
+      mInitCheck(false),
+      mStarted(false)
+{
+    ALOGV("FLACSource::FLACSource");
+    // re-use the same track metadata passed into constructor from FLACExtractor
+    mParser = new FLACParser(mDataSource);
+    mInitCheck  = mParser->initCheck();
+}
+
+FLACSource::~FLACSource()
+{
+    ALOGV("~FLACSource::FLACSource");
+    if (mStarted) {
+        stop();
+    }
+    delete mParser;
+}
+
+status_t FLACSource::start(MetaDataBase * /* params */)
+{
+    ALOGV("FLACSource::start");
+
+    CHECK(!mStarted);
+    mParser->allocateBuffers();
+    mStarted = true;
+
+    return OK;
+}
+
+status_t FLACSource::stop()
+{
+    ALOGV("FLACSource::stop");
+
+    CHECK(mStarted);
+    mParser->releaseBuffers();
+    mStarted = false;
+
+    return OK;
+}
+
+status_t FLACSource::getFormat(MetaDataBase &meta)
+{
+    meta = mTrackMetadata;
+    return OK;
+}
+
+status_t FLACSource::read(
+        MediaBufferBase **outBuffer, const ReadOptions *options)
+{
+    MediaBufferBase *buffer;
+    // process an optional seek request
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode mode;
+    if ((NULL != options) && options->getSeekTo(&seekTimeUs, &mode)) {
+        FLAC__uint64 sample;
+        if (seekTimeUs <= 0LL) {
+            sample = 0LL;
+        } else {
+            // sample and total samples are both zero-based, and seek to EOF ok
+            sample = (seekTimeUs * mParser->getSampleRate()) / 1000000LL;
+            if (sample >= mParser->getTotalSamples()) {
+                sample = mParser->getTotalSamples();
+            }
+        }
+        buffer = mParser->readBuffer(sample);
+    // otherwise read sequentially
+    } else {
+        buffer = mParser->readBuffer();
+    }
+    *outBuffer = buffer;
+    return buffer != NULL ? (status_t) OK : (status_t) ERROR_END_OF_STREAM;
+}
+
+// FLACExtractor
+
+FLACExtractor::FLACExtractor(
+        DataSourceBase *dataSource)
+    : mDataSource(dataSource),
+      mParser(nullptr),
+      mInitCheck(false)
+{
+    ALOGV("FLACExtractor::FLACExtractor");
+    // FLACParser will fill in the metadata for us
+    mParser = new FLACParser(mDataSource, &mFileMetadata, &mTrackMetadata);
+    mInitCheck = mParser->initCheck();
+}
+
+FLACExtractor::~FLACExtractor()
+{
+    ALOGV("~FLACExtractor::FLACExtractor");
+    delete mParser;
+}
+
+size_t FLACExtractor::countTracks()
+{
+    return mInitCheck == OK ? 1 : 0;
+}
+
+MediaTrack *FLACExtractor::getTrack(size_t index)
+{
+    if (mInitCheck != OK || index > 0) {
+        return NULL;
+    }
+    return new FLACSource(mDataSource, mTrackMetadata);
+}
+
+status_t FLACExtractor::getTrackMetaData(
+        MetaDataBase &meta,
+        size_t index, uint32_t /* flags */) {
+    if (mInitCheck != OK || index > 0) {
+        return UNKNOWN_ERROR;
+    }
+    meta = mTrackMetadata;
+    return OK;
+}
+
+status_t FLACExtractor::getMetaData(MetaDataBase &meta)
+{
+    meta = mFileMetadata;
+    return OK;
+}
+
+// Sniffer
+
+bool SniffFLAC(DataSourceBase *source, float *confidence)
+{
+    // first 4 is the signature word
+    // second 4 is the sizeof STREAMINFO
+    // 042 is the mandatory STREAMINFO
+    // no need to read rest of the header, as a premature EOF will be caught later
+    uint8_t header[4+4];
+    if (source->readAt(0, header, sizeof(header)) != sizeof(header)
+            || memcmp("fLaC\0\0\0\042", header, 4+4))
+    {
+        return false;
+    }
+
+    *confidence = 0.5;
+
+    return true;
+}
+
+
+extern "C" {
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
+MediaExtractor::ExtractorDef GETEXTRACTORDEF() {
+    return {
+        MediaExtractor::EXTRACTORDEF_VERSION,
+            UUID("1364b048-cc45-4fda-9934-327d0ebf9829"),
+            1,
+            "FLAC Extractor",
+            [](
+                    DataSourceBase *source,
+                    float *confidence,
+                    void **,
+                    MediaExtractor::FreeMetaFunc *) -> MediaExtractor::CreatorFunc {
+                if (SniffFLAC(source, confidence)) {
+                    return [](
+                            DataSourceBase *source,
+                            void *) -> MediaExtractor* {
+                        return new FLACExtractor(source);};
+                }
+                return NULL;
+            }
+     };
+}
+
+} // extern "C"
+
+}  // namespace android

diff --git a/media/extractors/flac/FLACExtractor.h b/media/extractors/flac/FLACExtractor.h
new file mode 100644
index 0000000..7fb6ec6
--- /dev/null
+++ b/media/extractors/flac/FLACExtractor.h

@@ -0,0 +1,62 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef FLAC_EXTRACTOR_H_
+#define FLAC_EXTRACTOR_H_
+
+#include <media/DataSourceBase.h>
+#include <media/MediaExtractor.h>
+#include <media/stagefright/MetaDataBase.h>
+#include <utils/String8.h>
+
+namespace android {
+
+class FLACParser;
+
+class FLACExtractor : public MediaExtractor {
+
+public:
+    explicit FLACExtractor(DataSourceBase *source);
+
+    virtual size_t countTracks();
+    virtual MediaTrack *getTrack(size_t index);
+    virtual status_t getTrackMetaData(MetaDataBase& meta, size_t index, uint32_t flags);
+
+    virtual status_t getMetaData(MetaDataBase& meta);
+    virtual const char * name() { return "FLACExtractor"; }
+
+protected:
+    virtual ~FLACExtractor();
+
+private:
+    DataSourceBase *mDataSource;
+    FLACParser *mParser;
+    status_t mInitCheck;
+    MetaDataBase mFileMetadata;
+
+    // There is only one track
+    MetaDataBase mTrackMetadata;
+
+    FLACExtractor(const FLACExtractor &);
+    FLACExtractor &operator=(const FLACExtractor &);
+
+};
+
+bool SniffFLAC(DataSourceBase *source, float *confidence);
+
+}  // namespace android
+
+#endif  // FLAC_EXTRACTOR_H_

diff --git a/media/libstagefright/codecs/on2/h264dec/MODULE_LICENSE_APACHE2 b/media/extractors/flac/MODULE_LICENSE_APACHE2
similarity index 100%
copy from media/libstagefright/codecs/on2/h264dec/MODULE_LICENSE_APACHE2
copy to media/extractors/flac/MODULE_LICENSE_APACHE2


diff --git a/media/libstagefright/codecs/on2/h264dec/NOTICE b/media/extractors/flac/NOTICE
similarity index 100%
copy from media/libstagefright/codecs/on2/h264dec/NOTICE
copy to media/extractors/flac/NOTICE


diff --git a/media/extractors/flac/exports.lds b/media/extractors/flac/exports.lds
new file mode 100644
index 0000000..b1309ee
--- /dev/null
+++ b/media/extractors/flac/exports.lds

@@ -0,0 +1 @@
+{ global: GETEXTRACTORDEF; local: *; };

diff --git a/media/extractors/midi/Android.bp b/media/extractors/midi/Android.bp
new file mode 100644
index 0000000..fde09df18
--- /dev/null
+++ b/media/extractors/midi/Android.bp

@@ -0,0 +1,42 @@
+cc_library_shared {
+
+    srcs: ["MidiExtractor.cpp"],
+
+    include_dirs: [
+        "frameworks/av/media/libstagefright/include",
+    ],
+
+    shared_libs: [
+        "liblog",
+        "libmediaextractor",
+    ],
+
+    static_libs: [
+        "libmedia_midiiowrapper",
+        "libsonivox",
+        "libstagefright_foundation"
+    ],
+    name: "libmidiextractor",
+    relative_install_path: "extractors",
+
+    compile_multilib: "first",
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+        "-fvisibility=hidden",
+    ],
+    version_script: "exports.lds",
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+        diag: {
+            cfi: true,
+        },
+    },
+
+}

diff --git a/media/libstagefright/matroska/MODULE_LICENSE_APACHE2 b/media/extractors/midi/MODULE_LICENSE_APACHE2
similarity index 100%
copy from media/libstagefright/matroska/MODULE_LICENSE_APACHE2
copy to media/extractors/midi/MODULE_LICENSE_APACHE2


diff --git a/media/extractors/midi/MidiExtractor.cpp b/media/extractors/midi/MidiExtractor.cpp
new file mode 100644
index 0000000..a30b6f8
--- /dev/null
+++ b/media/extractors/midi/MidiExtractor.cpp

@@ -0,0 +1,353 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MidiExtractor"
+#include <utils/Log.h>
+
+#include "MidiExtractor.h"
+
+#include <media/MidiIoWrapper.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <media/MediaTrack.h>
+#include <libsonivox/eas_reverb.h>
+
+namespace android {
+
+// how many Sonivox output buffers to aggregate into one MediaBufferBase
+static const int NUM_COMBINE_BUFFERS = 4;
+
+class MidiSource : public MediaTrack {
+
+public:
+    MidiSource(
+            MidiEngine &engine,
+            MetaDataBase &trackMetadata);
+
+    virtual status_t start(MetaDataBase *params);
+    virtual status_t stop();
+    virtual status_t getFormat(MetaDataBase&);
+
+    virtual status_t read(
+            MediaBufferBase **buffer, const ReadOptions *options = NULL);
+
+protected:
+    virtual ~MidiSource();
+
+private:
+    MidiEngine &mEngine;
+    MetaDataBase &mTrackMetadata;
+    bool mInitCheck;
+    bool mStarted;
+
+    status_t init();
+
+    // no copy constructor or assignment
+    MidiSource(const MidiSource &);
+    MidiSource &operator=(const MidiSource &);
+
+};
+
+
+// Midisource
+
+MidiSource::MidiSource(
+        MidiEngine &engine,
+        MetaDataBase &trackMetadata)
+    : mEngine(engine),
+      mTrackMetadata(trackMetadata),
+      mInitCheck(false),
+      mStarted(false)
+{
+    ALOGV("MidiSource ctor");
+    mInitCheck = init();
+}
+
+MidiSource::~MidiSource()
+{
+    ALOGV("MidiSource dtor");
+    if (mStarted) {
+        stop();
+    }
+}
+
+status_t MidiSource::start(MetaDataBase * /* params */)
+{
+    ALOGV("MidiSource::start");
+
+    CHECK(!mStarted);
+    mStarted = true;
+    mEngine.allocateBuffers();
+    return OK;
+}
+
+status_t MidiSource::stop()
+{
+    ALOGV("MidiSource::stop");
+
+    CHECK(mStarted);
+    mStarted = false;
+    mEngine.releaseBuffers();
+
+    return OK;
+}
+
+status_t MidiSource::getFormat(MetaDataBase &meta)
+{
+    meta = mTrackMetadata;
+    return OK;
+}
+
+status_t MidiSource::read(
+        MediaBufferBase **outBuffer, const ReadOptions *options)
+{
+    ALOGV("MidiSource::read");
+    MediaBufferBase *buffer;
+    // process an optional seek request
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode mode;
+    if ((NULL != options) && options->getSeekTo(&seekTimeUs, &mode)) {
+        if (seekTimeUs <= 0LL) {
+            seekTimeUs = 0LL;
+        }
+        mEngine.seekTo(seekTimeUs);
+    }
+    buffer = mEngine.readBuffer();
+    *outBuffer = buffer;
+    ALOGV("MidiSource::read %p done", this);
+    return buffer != NULL ? (status_t) OK : (status_t) ERROR_END_OF_STREAM;
+}
+
+status_t MidiSource::init()
+{
+    ALOGV("MidiSource::init");
+    return OK;
+}
+
+// MidiEngine
+
+MidiEngine::MidiEngine(DataSourceBase *dataSource,
+        MetaDataBase *fileMetadata,
+        MetaDataBase *trackMetadata) :
+            mGroup(NULL),
+            mEasData(NULL),
+            mEasHandle(NULL),
+            mEasConfig(NULL),
+            mIsInitialized(false) {
+    mIoWrapper = new MidiIoWrapper(dataSource);
+    // spin up a new EAS engine
+    EAS_I32 temp;
+    EAS_RESULT result = EAS_Init(&mEasData);
+
+    if (result == EAS_SUCCESS) {
+        result = EAS_OpenFile(mEasData, mIoWrapper->getLocator(), &mEasHandle);
+    }
+    if (result == EAS_SUCCESS) {
+        result = EAS_Prepare(mEasData, mEasHandle);
+    }
+    if (result == EAS_SUCCESS) {
+        result = EAS_ParseMetaData(mEasData, mEasHandle, &temp);
+    }
+
+    if (result != EAS_SUCCESS) {
+        return;
+    }
+
+    if (fileMetadata != NULL) {
+        fileMetadata->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MIDI);
+    }
+
+    if (trackMetadata != NULL) {
+        trackMetadata->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
+        trackMetadata->setInt64(kKeyDuration, 1000ll * temp); // milli->micro
+        mEasConfig = EAS_Config();
+        trackMetadata->setInt32(kKeySampleRate, mEasConfig->sampleRate);
+        trackMetadata->setInt32(kKeyChannelCount, mEasConfig->numChannels);
+        trackMetadata->setInt32(kKeyPcmEncoding, kAudioEncodingPcm16bit);
+    }
+    mIsInitialized = true;
+}
+
+MidiEngine::~MidiEngine() {
+    if (mEasHandle) {
+        EAS_CloseFile(mEasData, mEasHandle);
+    }
+    if (mEasData) {
+        EAS_Shutdown(mEasData);
+    }
+    delete mGroup;
+    delete mIoWrapper;
+}
+
+status_t MidiEngine::initCheck() {
+    return mIsInitialized ? OK : UNKNOWN_ERROR;
+}
+
+status_t MidiEngine::allocateBuffers() {
+    // select reverb preset and enable
+    EAS_SetParameter(mEasData, EAS_MODULE_REVERB, EAS_PARAM_REVERB_PRESET, EAS_PARAM_REVERB_CHAMBER);
+    EAS_SetParameter(mEasData, EAS_MODULE_REVERB, EAS_PARAM_REVERB_BYPASS, EAS_FALSE);
+
+    mGroup = new MediaBufferGroup;
+    int bufsize = sizeof(EAS_PCM)
+            * mEasConfig->mixBufferSize * mEasConfig->numChannels * NUM_COMBINE_BUFFERS;
+    ALOGV("using %d byte buffer", bufsize);
+    mGroup->add_buffer(MediaBufferBase::Create(bufsize));
+    return OK;
+}
+
+status_t MidiEngine::releaseBuffers() {
+    delete mGroup;
+    mGroup = NULL;
+    return OK;
+}
+
+status_t MidiEngine::seekTo(int64_t positionUs) {
+    ALOGV("seekTo %lld", (long long)positionUs);
+    EAS_RESULT result = EAS_Locate(mEasData, mEasHandle, positionUs / 1000, false);
+    return result == EAS_SUCCESS ? OK : UNKNOWN_ERROR;
+}
+
+MediaBufferBase* MidiEngine::readBuffer() {
+    EAS_STATE state;
+    EAS_State(mEasData, mEasHandle, &state);
+    if ((state == EAS_STATE_STOPPED) || (state == EAS_STATE_ERROR)) {
+        return NULL;
+    }
+    MediaBufferBase *buffer;
+    status_t err = mGroup->acquire_buffer(&buffer);
+    if (err != OK) {
+        ALOGE("readBuffer: no buffer");
+        return NULL;
+    }
+    EAS_I32 timeMs;
+    EAS_GetLocation(mEasData, mEasHandle, &timeMs);
+    int64_t timeUs = 1000ll * timeMs;
+    buffer->meta_data().setInt64(kKeyTime, timeUs);
+
+    EAS_PCM* p = (EAS_PCM*) buffer->data();
+    int numBytesOutput = 0;
+    for (int i = 0; i < NUM_COMBINE_BUFFERS; i++) {
+        EAS_I32 numRendered;
+        EAS_RESULT result = EAS_Render(mEasData, p, mEasConfig->mixBufferSize, &numRendered);
+        if (result != EAS_SUCCESS) {
+            ALOGE("EAS_Render() returned %ld, numBytesOutput = %d", result, numBytesOutput);
+            buffer->release();
+            return NULL; // Stop processing to prevent infinite loops.
+        }
+        p += numRendered * mEasConfig->numChannels;
+        numBytesOutput += numRendered * mEasConfig->numChannels * sizeof(EAS_PCM);
+    }
+    buffer->set_range(0, numBytesOutput);
+    ALOGV("readBuffer: returning %zd in buffer %p", buffer->range_length(), buffer);
+    return buffer;
+}
+
+
+// MidiExtractor
+
+MidiExtractor::MidiExtractor(
+        DataSourceBase *dataSource)
+    : mDataSource(dataSource),
+      mInitCheck(false)
+{
+    ALOGV("MidiExtractor ctor");
+    mEngine = new MidiEngine(mDataSource, &mFileMetadata, &mTrackMetadata);
+    mInitCheck = mEngine->initCheck();
+}
+
+MidiExtractor::~MidiExtractor()
+{
+    ALOGV("MidiExtractor dtor");
+}
+
+size_t MidiExtractor::countTracks()
+{
+    return mInitCheck == OK ? 1 : 0;
+}
+
+MediaTrack *MidiExtractor::getTrack(size_t index)
+{
+    if (mInitCheck != OK || index > 0) {
+        return NULL;
+    }
+    return new MidiSource(*mEngine, mTrackMetadata);
+}
+
+status_t MidiExtractor::getTrackMetaData(
+        MetaDataBase &meta,
+        size_t index, uint32_t /* flags */) {
+    ALOGV("MidiExtractor::getTrackMetaData");
+    if (mInitCheck != OK || index > 0) {
+        return UNKNOWN_ERROR;
+    }
+    meta = mTrackMetadata;
+    return OK;
+}
+
+status_t MidiExtractor::getMetaData(MetaDataBase &meta)
+{
+    ALOGV("MidiExtractor::getMetaData");
+    meta = mFileMetadata;
+    return OK;
+}
+
+// Sniffer
+
+bool SniffMidi(DataSourceBase *source, float *confidence)
+{
+    MidiEngine p(source, NULL, NULL);
+    if (p.initCheck() == OK) {
+        *confidence = 0.8;
+        ALOGV("SniffMidi: yes");
+        return true;
+    }
+    ALOGV("SniffMidi: no");
+    return false;
+
+}
+
+extern "C" {
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
+MediaExtractor::ExtractorDef GETEXTRACTORDEF() {
+    return {
+        MediaExtractor::EXTRACTORDEF_VERSION,
+        UUID("ef6cca0a-f8a2-43e6-ba5f-dfcd7c9a7ef2"),
+        1,
+        "MIDI Extractor",
+        [](
+                DataSourceBase *source,
+                float *confidence,
+                void **,
+                MediaExtractor::FreeMetaFunc *) -> MediaExtractor::CreatorFunc {
+            if (SniffMidi(source, confidence)) {
+                return [](
+                        DataSourceBase *source,
+                        void *) -> MediaExtractor* {
+                    return new MidiExtractor(source);};
+            }
+            return NULL;
+        }
+    };
+}
+
+} // extern "C"
+
+}  // namespace android

diff --git a/media/extractors/midi/MidiExtractor.h b/media/extractors/midi/MidiExtractor.h
new file mode 100644
index 0000000..244dd0f
--- /dev/null
+++ b/media/extractors/midi/MidiExtractor.h

@@ -0,0 +1,95 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MIDI_EXTRACTOR_H_
+#define MIDI_EXTRACTOR_H_
+
+#include <media/DataSourceBase.h>
+#include <media/MediaExtractor.h>
+#include <media/stagefright/MediaBufferBase.h>
+#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MetaDataBase.h>
+#include <media/MidiIoWrapper.h>
+#include <utils/String8.h>
+#include <libsonivox/eas.h>
+
+namespace android {
+
+class MidiEngine {
+public:
+    explicit MidiEngine(DataSourceBase *dataSource,
+            MetaDataBase *fileMetadata,
+            MetaDataBase *trackMetadata);
+    ~MidiEngine();
+
+    status_t initCheck();
+
+    status_t allocateBuffers();
+    status_t releaseBuffers();
+    status_t seekTo(int64_t positionUs);
+    MediaBufferBase* readBuffer();
+private:
+    MidiIoWrapper *mIoWrapper;
+    MediaBufferGroup *mGroup;
+    EAS_DATA_HANDLE mEasData;
+    EAS_HANDLE mEasHandle;
+    const S_EAS_LIB_CONFIG* mEasConfig;
+    bool mIsInitialized;
+};
+
+class MidiExtractor : public MediaExtractor {
+
+public:
+    explicit MidiExtractor(DataSourceBase *source);
+
+    virtual size_t countTracks();
+    virtual MediaTrack *getTrack(size_t index);
+    virtual status_t getTrackMetaData(MetaDataBase& meta, size_t index, uint32_t flags);
+
+    virtual status_t getMetaData(MetaDataBase& meta);
+    virtual const char * name() { return "MidiExtractor"; }
+
+protected:
+    virtual ~MidiExtractor();
+
+private:
+    DataSourceBase *mDataSource;
+    status_t mInitCheck;
+    MetaDataBase mFileMetadata;
+
+    // There is only one track
+    MetaDataBase mTrackMetadata;
+
+    MidiEngine *mEngine;
+
+    EAS_DATA_HANDLE     mEasData;
+    EAS_HANDLE          mEasHandle;
+    EAS_PCM*            mAudioBuffer;
+    EAS_I32             mPlayTime;
+    EAS_I32             mDuration;
+    EAS_STATE           mState;
+    EAS_FILE            mFileLocator;
+
+    MidiExtractor(const MidiExtractor &);
+    MidiExtractor &operator=(const MidiExtractor &);
+
+};
+
+bool SniffMidi(DataSourceBase *source, float *confidence);
+
+}  // namespace android
+
+#endif  // MIDI_EXTRACTOR_H_

diff --git a/media/libstagefright/matroska/NOTICE b/media/extractors/midi/NOTICE
similarity index 100%
copy from media/libstagefright/matroska/NOTICE
copy to media/extractors/midi/NOTICE


diff --git a/media/extractors/midi/exports.lds b/media/extractors/midi/exports.lds
new file mode 100644
index 0000000..b1309ee
--- /dev/null
+++ b/media/extractors/midi/exports.lds

@@ -0,0 +1 @@
+{ global: GETEXTRACTORDEF; local: *; };

diff --git a/media/extractors/mkv/Android.bp b/media/extractors/mkv/Android.bp
new file mode 100644
index 0000000..681fd35
--- /dev/null
+++ b/media/extractors/mkv/Android.bp

@@ -0,0 +1,48 @@
+cc_library_shared {
+
+    srcs: ["MatroskaExtractor.cpp"],
+
+    include_dirs: [
+        "external/flac/include",
+        "external/libvpx/libwebm",
+        "frameworks/av/media/libstagefright/flac/dec",
+        "frameworks/av/media/libstagefright/include",
+    ],
+
+    shared_libs: [
+        "liblog",
+        "libmediaextractor",
+    ],
+
+    static_libs: [
+        "libstagefright_flacdec",
+        "libstagefright_foundation",
+        "libstagefright_metadatautils",
+        "libwebm",
+        "libutils",
+    ],
+
+    name: "libmkvextractor",
+    relative_install_path: "extractors",
+
+    compile_multilib: "first",
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+        "-fvisibility=hidden",
+    ],
+    version_script: "exports.lds",
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+        diag: {
+            cfi: true,
+        },
+    },
+
+}

diff --git a/media/libstagefright/matroska/MODULE_LICENSE_APACHE2 b/media/extractors/mkv/MODULE_LICENSE_APACHE2
similarity index 100%
copy from media/libstagefright/matroska/MODULE_LICENSE_APACHE2
copy to media/extractors/mkv/MODULE_LICENSE_APACHE2


diff --git a/media/extractors/mkv/MatroskaExtractor.cpp b/media/extractors/mkv/MatroskaExtractor.cpp
new file mode 100644
index 0000000..d657582
--- /dev/null
+++ b/media/extractors/mkv/MatroskaExtractor.cpp

@@ -0,0 +1,1665 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MatroskaExtractor"
+#include <utils/Log.h>
+
+#include "FLACDecoder.h"
+#include "MatroskaExtractor.h"
+
+#include <media/DataSourceBase.h>
+#include <media/ExtractorUtils.h>
+#include <media/MediaTrack.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AUtils.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ByteUtils.h>
+#include <media/stagefright/foundation/ColorUtils.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/MediaBufferBase.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/MetaDataUtils.h>
+#include <utils/String8.h>
+
+#include <arpa/inet.h>
+#include <inttypes.h>
+#include <vector>
+
+namespace android {
+
+struct DataSourceBaseReader : public mkvparser::IMkvReader {
+    explicit DataSourceBaseReader(DataSourceBase *source)
+        : mSource(source) {
+    }
+
+    virtual int Read(long long position, long length, unsigned char* buffer) {
+        CHECK(position >= 0);
+        CHECK(length >= 0);
+
+        if (length == 0) {
+            return 0;
+        }
+
+        ssize_t n = mSource->readAt(position, buffer, length);
+
+        if (n <= 0) {
+            return -1;
+        }
+
+        return 0;
+    }
+
+    virtual int Length(long long* total, long long* available) {
+        off64_t size;
+        if (mSource->getSize(&size) != OK) {
+            *total = -1;
+            *available = (long long)((1ull << 63) - 1);
+
+            return 0;
+        }
+
+        if (total) {
+            *total = size;
+        }
+
+        if (available) {
+            *available = size;
+        }
+
+        return 0;
+    }
+
+private:
+    DataSourceBase *mSource;
+
+    DataSourceBaseReader(const DataSourceBaseReader &);
+    DataSourceBaseReader &operator=(const DataSourceBaseReader &);
+};
+
+////////////////////////////////////////////////////////////////////////////////
+
+struct BlockIterator {
+    BlockIterator(MatroskaExtractor *extractor, unsigned long trackNum, unsigned long index);
+
+    bool eos() const;
+
+    void advance();
+    void reset();
+
+    void seek(
+            int64_t seekTimeUs, bool isAudio,
+            int64_t *actualFrameTimeUs);
+
+    const mkvparser::Block *block() const;
+    int64_t blockTimeUs() const;
+
+private:
+    MatroskaExtractor *mExtractor;
+    long long mTrackNum;
+    unsigned long mIndex;
+
+    const mkvparser::Cluster *mCluster;
+    const mkvparser::BlockEntry *mBlockEntry;
+    long mBlockEntryIndex;
+
+    void advance_l();
+
+    BlockIterator(const BlockIterator &);
+    BlockIterator &operator=(const BlockIterator &);
+};
+
+struct MatroskaSource : public MediaTrack {
+    MatroskaSource(MatroskaExtractor *extractor, size_t index);
+
+    virtual status_t start(MetaDataBase *params);
+    virtual status_t stop();
+
+    virtual status_t getFormat(MetaDataBase &);
+
+    virtual status_t read(
+            MediaBufferBase **buffer, const ReadOptions *options);
+
+protected:
+    virtual ~MatroskaSource();
+
+private:
+    enum Type {
+        AVC,
+        AAC,
+        HEVC,
+        OTHER
+    };
+
+    MatroskaExtractor *mExtractor;
+    size_t mTrackIndex;
+    Type mType;
+    bool mIsAudio;
+    BlockIterator mBlockIter;
+    ssize_t mNALSizeLen;  // for type AVC or HEVC
+
+    List<MediaBufferBase *> mPendingFrames;
+
+    status_t advance();
+
+    status_t setWebmBlockCryptoInfo(MediaBufferBase *mbuf);
+    status_t readBlock();
+    void clearPendingFrames();
+
+    MatroskaSource(const MatroskaSource &);
+    MatroskaSource &operator=(const MatroskaSource &);
+};
+
+const mkvparser::Track* MatroskaExtractor::TrackInfo::getTrack() const {
+    return mExtractor->mSegment->GetTracks()->GetTrackByNumber(mTrackNum);
+}
+
+// This function does exactly the same as mkvparser::Cues::Find, except that it
+// searches in our own track based vectors. We should not need this once mkvparser
+// adds the same functionality.
+const mkvparser::CuePoint::TrackPosition *MatroskaExtractor::TrackInfo::find(
+        long long timeNs) const {
+    ALOGV("mCuePoints.size %zu", mCuePoints.size());
+    if (mCuePoints.empty()) {
+        return NULL;
+    }
+
+    const mkvparser::CuePoint* cp = mCuePoints.itemAt(0);
+    const mkvparser::Track* track = getTrack();
+    if (timeNs <= cp->GetTime(mExtractor->mSegment)) {
+        return cp->Find(track);
+    }
+
+    // Binary searches through relevant cues; assumes cues are ordered by timecode.
+    // If we do detect out-of-order cues, return NULL.
+    size_t lo = 0;
+    size_t hi = mCuePoints.size();
+    while (lo < hi) {
+        const size_t mid = lo + (hi - lo) / 2;
+        const mkvparser::CuePoint* const midCp = mCuePoints.itemAt(mid);
+        const long long cueTimeNs = midCp->GetTime(mExtractor->mSegment);
+        if (cueTimeNs <= timeNs) {
+            lo = mid + 1;
+        } else {
+            hi = mid;
+        }
+    }
+
+    if (lo == 0) {
+        return NULL;
+    }
+
+    cp = mCuePoints.itemAt(lo - 1);
+    if (cp->GetTime(mExtractor->mSegment) > timeNs) {
+        return NULL;
+    }
+
+    return cp->Find(track);
+}
+
+MatroskaSource::MatroskaSource(
+        MatroskaExtractor *extractor, size_t index)
+    : mExtractor(extractor),
+      mTrackIndex(index),
+      mType(OTHER),
+      mIsAudio(false),
+      mBlockIter(mExtractor,
+                 mExtractor->mTracks.itemAt(index).mTrackNum,
+                 index),
+      mNALSizeLen(-1) {
+    MetaDataBase &meta = mExtractor->mTracks.editItemAt(index).mMeta;
+
+    const char *mime;
+    CHECK(meta.findCString(kKeyMIMEType, &mime));
+
+    mIsAudio = !strncasecmp("audio/", mime, 6);
+
+    if (!strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC)) {
+        mType = AVC;
+
+        uint32_t dummy;
+        const uint8_t *avcc;
+        size_t avccSize;
+        int32_t nalSizeLen = 0;
+        if (meta.findInt32(kKeyNalLengthSize, &nalSizeLen)) {
+            if (nalSizeLen >= 0 && nalSizeLen <= 4) {
+                mNALSizeLen = nalSizeLen;
+            }
+        } else if (meta.findData(kKeyAVCC, &dummy, (const void **)&avcc, &avccSize)
+                && avccSize >= 5u) {
+            mNALSizeLen = 1 + (avcc[4] & 3);
+            ALOGV("mNALSizeLen = %zd", mNALSizeLen);
+        } else {
+            ALOGE("No mNALSizeLen");
+        }
+    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_HEVC)) {
+        mType = HEVC;
+
+        uint32_t dummy;
+        const uint8_t *hvcc;
+        size_t hvccSize;
+        if (meta.findData(kKeyHVCC, &dummy, (const void **)&hvcc, &hvccSize)
+                && hvccSize >= 22u) {
+            mNALSizeLen = 1 + (hvcc[14+7] & 3);
+            ALOGV("mNALSizeLen = %zu", mNALSizeLen);
+        } else {
+            ALOGE("No mNALSizeLen");
+        }
+    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC)) {
+        mType = AAC;
+    }
+}
+
+MatroskaSource::~MatroskaSource() {
+    clearPendingFrames();
+}
+
+status_t MatroskaSource::start(MetaDataBase * /* params */) {
+    if (mType == AVC && mNALSizeLen < 0) {
+        return ERROR_MALFORMED;
+    }
+
+    mBlockIter.reset();
+
+    return OK;
+}
+
+status_t MatroskaSource::stop() {
+    clearPendingFrames();
+
+    return OK;
+}
+
+status_t MatroskaSource::getFormat(MetaDataBase &meta) {
+    meta = mExtractor->mTracks.itemAt(mTrackIndex).mMeta;
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+BlockIterator::BlockIterator(
+        MatroskaExtractor *extractor, unsigned long trackNum, unsigned long index)
+    : mExtractor(extractor),
+      mTrackNum(trackNum),
+      mIndex(index),
+      mCluster(NULL),
+      mBlockEntry(NULL),
+      mBlockEntryIndex(0) {
+    reset();
+}
+
+bool BlockIterator::eos() const {
+    return mCluster == NULL || mCluster->EOS();
+}
+
+void BlockIterator::advance() {
+    Mutex::Autolock autoLock(mExtractor->mLock);
+    advance_l();
+}
+
+void BlockIterator::advance_l() {
+    for (;;) {
+        long res = mCluster->GetEntry(mBlockEntryIndex, mBlockEntry);
+        ALOGV("GetEntry returned %ld", res);
+
+        long long pos;
+        long len;
+        if (res < 0) {
+            // Need to parse this cluster some more
+
+            CHECK_EQ(res, mkvparser::E_BUFFER_NOT_FULL);
+
+            res = mCluster->Parse(pos, len);
+            ALOGV("Parse returned %ld", res);
+
+            if (res < 0) {
+                // I/O error
+
+                ALOGE("Cluster::Parse returned result %ld", res);
+
+                mCluster = NULL;
+                break;
+            }
+
+            continue;
+        } else if (res == 0) {
+            // We're done with this cluster
+
+            const mkvparser::Cluster *nextCluster;
+            res = mExtractor->mSegment->ParseNext(
+                    mCluster, nextCluster, pos, len);
+            ALOGV("ParseNext returned %ld", res);
+
+            if (res != 0) {
+                // EOF or error
+
+                mCluster = NULL;
+                break;
+            }
+
+            CHECK_EQ(res, 0);
+            CHECK(nextCluster != NULL);
+            CHECK(!nextCluster->EOS());
+
+            mCluster = nextCluster;
+
+            res = mCluster->Parse(pos, len);
+            ALOGV("Parse (2) returned %ld", res);
+
+            if (res < 0) {
+                // I/O error
+
+                ALOGE("Cluster::Parse returned result %ld", res);
+
+                mCluster = NULL;
+                break;
+            }
+
+            mBlockEntryIndex = 0;
+            continue;
+        }
+
+        CHECK(mBlockEntry != NULL);
+        CHECK(mBlockEntry->GetBlock() != NULL);
+        ++mBlockEntryIndex;
+
+        if (mBlockEntry->GetBlock()->GetTrackNumber() == mTrackNum) {
+            break;
+        }
+    }
+}
+
+void BlockIterator::reset() {
+    Mutex::Autolock autoLock(mExtractor->mLock);
+
+    mCluster = mExtractor->mSegment->GetFirst();
+    mBlockEntry = NULL;
+    mBlockEntryIndex = 0;
+
+    do {
+        advance_l();
+    } while (!eos() && block()->GetTrackNumber() != mTrackNum);
+}
+
+void BlockIterator::seek(
+        int64_t seekTimeUs, bool isAudio,
+        int64_t *actualFrameTimeUs) {
+    Mutex::Autolock autoLock(mExtractor->mLock);
+
+    *actualFrameTimeUs = -1ll;
+
+    if (seekTimeUs > INT64_MAX / 1000ll ||
+            seekTimeUs < INT64_MIN / 1000ll ||
+            (mExtractor->mSeekPreRollNs > 0 &&
+                    (seekTimeUs * 1000ll) < INT64_MIN + mExtractor->mSeekPreRollNs) ||
+            (mExtractor->mSeekPreRollNs < 0 &&
+                    (seekTimeUs * 1000ll) > INT64_MAX + mExtractor->mSeekPreRollNs)) {
+        ALOGE("cannot seek to %lld", (long long) seekTimeUs);
+        return;
+    }
+
+    const int64_t seekTimeNs = seekTimeUs * 1000ll - mExtractor->mSeekPreRollNs;
+
+    mkvparser::Segment* const pSegment = mExtractor->mSegment;
+
+    // Special case the 0 seek to avoid loading Cues when the application
+    // extraneously seeks to 0 before playing.
+    if (seekTimeNs <= 0) {
+        ALOGV("Seek to beginning: %" PRId64, seekTimeUs);
+        mCluster = pSegment->GetFirst();
+        mBlockEntryIndex = 0;
+        do {
+            advance_l();
+        } while (!eos() && block()->GetTrackNumber() != mTrackNum);
+        return;
+    }
+
+    ALOGV("Seeking to: %" PRId64, seekTimeUs);
+
+    // If the Cues have not been located then find them.
+    const mkvparser::Cues* pCues = pSegment->GetCues();
+    const mkvparser::SeekHead* pSH = pSegment->GetSeekHead();
+    if (!pCues && pSH) {
+        const size_t count = pSH->GetCount();
+        const mkvparser::SeekHead::Entry* pEntry;
+        ALOGV("No Cues yet");
+
+        for (size_t index = 0; index < count; index++) {
+            pEntry = pSH->GetEntry(index);
+
+            if (pEntry->id == 0x0C53BB6B) { // Cues ID
+                long len; long long pos;
+                pSegment->ParseCues(pEntry->pos, pos, len);
+                pCues = pSegment->GetCues();
+                ALOGV("Cues found");
+                break;
+            }
+        }
+
+        if (!pCues) {
+            ALOGE("No Cues in file");
+            return;
+        }
+    }
+    else if (!pSH) {
+        ALOGE("No SeekHead");
+        return;
+    }
+
+    const mkvparser::CuePoint* pCP;
+    mkvparser::Tracks const *pTracks = pSegment->GetTracks();
+    while (!pCues->DoneParsing()) {
+        pCues->LoadCuePoint();
+        pCP = pCues->GetLast();
+        CHECK(pCP);
+
+        size_t trackCount = mExtractor->mTracks.size();
+        for (size_t index = 0; index < trackCount; ++index) {
+            MatroskaExtractor::TrackInfo& track = mExtractor->mTracks.editItemAt(index);
+            const mkvparser::Track *pTrack = pTracks->GetTrackByNumber(track.mTrackNum);
+            if (pTrack && pTrack->GetType() == 1 && pCP->Find(pTrack)) { // VIDEO_TRACK
+                track.mCuePoints.push_back(pCP);
+            }
+        }
+
+        if (pCP->GetTime(pSegment) >= seekTimeNs) {
+            ALOGV("Parsed past relevant Cue");
+            break;
+        }
+    }
+
+    const mkvparser::CuePoint::TrackPosition *pTP = NULL;
+    const mkvparser::Track *thisTrack = pTracks->GetTrackByNumber(mTrackNum);
+    if (thisTrack->GetType() == 1) { // video
+        MatroskaExtractor::TrackInfo& track = mExtractor->mTracks.editItemAt(mIndex);
+        pTP = track.find(seekTimeNs);
+    } else {
+        // The Cue index is built around video keyframes
+        unsigned long int trackCount = pTracks->GetTracksCount();
+        for (size_t index = 0; index < trackCount; ++index) {
+            const mkvparser::Track *pTrack = pTracks->GetTrackByIndex(index);
+            if (pTrack && pTrack->GetType() == 1 && pCues->Find(seekTimeNs, pTrack, pCP, pTP)) {
+                ALOGV("Video track located at %zu", index);
+                break;
+            }
+        }
+    }
+
+
+    // Always *search* based on the video track, but finalize based on mTrackNum
+    if (!pTP) {
+        ALOGE("Did not locate the video track for seeking");
+        return;
+    }
+
+    mCluster = pSegment->FindOrPreloadCluster(pTP->m_pos);
+
+    CHECK(mCluster);
+    CHECK(!mCluster->EOS());
+
+    // mBlockEntryIndex starts at 0 but m_block starts at 1
+    CHECK_GT(pTP->m_block, 0);
+    mBlockEntryIndex = pTP->m_block - 1;
+
+    for (;;) {
+        advance_l();
+
+        if (eos()) break;
+
+        if (isAudio || block()->IsKey()) {
+            // Accept the first key frame
+            int64_t frameTimeUs = (block()->GetTime(mCluster) + 500LL) / 1000LL;
+            if (thisTrack->GetType() == 1 || frameTimeUs >= seekTimeUs) {
+                *actualFrameTimeUs = frameTimeUs;
+                ALOGV("Requested seek point: %" PRId64 " actual: %" PRId64,
+                      seekTimeUs, *actualFrameTimeUs);
+                break;
+            }
+        }
+    }
+}
+
+const mkvparser::Block *BlockIterator::block() const {
+    CHECK(!eos());
+
+    return mBlockEntry->GetBlock();
+}
+
+int64_t BlockIterator::blockTimeUs() const {
+    if (mCluster == NULL || mBlockEntry == NULL) {
+        return -1;
+    }
+    return (mBlockEntry->GetBlock()->GetTime(mCluster) + 500ll) / 1000ll;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+static unsigned U24_AT(const uint8_t *ptr) {
+    return ptr[0] << 16 | ptr[1] << 8 | ptr[2];
+}
+
+static AString uriDebugString(const char *uri) {
+    // find scheme
+    AString scheme;
+    for (size_t i = 0; i < strlen(uri); i++) {
+        const char c = uri[i];
+        if (!isascii(c)) {
+            break;
+        } else if (isalpha(c)) {
+            continue;
+        } else if (i == 0) {
+            // first character must be a letter
+            break;
+        } else if (isdigit(c) || c == '+' || c == '.' || c =='-') {
+            continue;
+        } else if (c != ':') {
+            break;
+        }
+        scheme = AString(uri, 0, i);
+        scheme.append("://<suppressed>");
+        return scheme;
+    }
+    return AString("<no-scheme URI suppressed>");
+}
+
+void MatroskaSource::clearPendingFrames() {
+    while (!mPendingFrames.empty()) {
+        MediaBufferBase *frame = *mPendingFrames.begin();
+        mPendingFrames.erase(mPendingFrames.begin());
+
+        frame->release();
+        frame = NULL;
+    }
+}
+
+status_t MatroskaSource::setWebmBlockCryptoInfo(MediaBufferBase *mbuf) {
+    if (mbuf->range_length() < 1 || mbuf->range_length() - 1 > INT32_MAX) {
+        // 1-byte signal
+        return ERROR_MALFORMED;
+    }
+
+    const uint8_t *data = (const uint8_t *)mbuf->data() + mbuf->range_offset();
+    bool encrypted = data[0] & 0x1;
+    bool partitioned = data[0] & 0x2;
+    if (encrypted && mbuf->range_length() < 9) {
+        // 1-byte signal + 8-byte IV
+        return ERROR_MALFORMED;
+    }
+
+    MetaDataBase &meta = mbuf->meta_data();
+    if (encrypted) {
+        uint8_t ctrCounter[16] = { 0 };
+        uint32_t type;
+        const uint8_t *keyId;
+        size_t keyIdSize;
+        const MetaDataBase &trackMeta = mExtractor->mTracks.itemAt(mTrackIndex).mMeta;
+        CHECK(trackMeta.findData(kKeyCryptoKey, &type, (const void **)&keyId, &keyIdSize));
+        meta.setData(kKeyCryptoKey, 0, keyId, keyIdSize);
+        memcpy(ctrCounter, data + 1, 8);
+        meta.setData(kKeyCryptoIV, 0, ctrCounter, 16);
+        if (partitioned) {
+            /*  0                   1                   2                   3
+             *  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+             * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+             * |  Signal Byte  |                                               |
+             * +-+-+-+-+-+-+-+-+             IV                                |
+             * |                                                               |
+             * |               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+             * |               | num_partition |     Partition 0 offset ->     |
+             * |-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-|
+             * |     -> Partition 0 offset     |              ...              |
+             * |-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-|
+             * |             ...               |     Partition n-1 offset ->   |
+             * |-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-|
+             * |     -> Partition n-1 offset   |                               |
+             * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               |
+             * |                    Clear/encrypted sample data                |
+             * |                                                               |
+             * |                                                               |
+             * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+             */
+            if (mbuf->range_length() < 10) {
+                return ERROR_MALFORMED;
+            }
+            uint8_t numPartitions = data[9];
+            if (mbuf->range_length() - 10 < numPartitions * sizeof(uint32_t)) {
+                return ERROR_MALFORMED;
+            }
+            std::vector<uint32_t> plainSizes, encryptedSizes;
+            uint32_t prev = 0;
+            uint32_t frameOffset = 10 + numPartitions * sizeof(uint32_t);
+            const uint32_t *partitions = reinterpret_cast<const uint32_t*>(data + 10);
+            for (uint32_t i = 0; i <= numPartitions; ++i) {
+                uint32_t p_i = i < numPartitions
+                        ? ntohl(partitions[i])
+                        : (mbuf->range_length() - frameOffset);
+                if (p_i < prev) {
+                    return ERROR_MALFORMED;
+                }
+                uint32_t size = p_i - prev;
+                prev = p_i;
+                if (i % 2) {
+                    encryptedSizes.push_back(size);
+                } else {
+                    plainSizes.push_back(size);
+                }
+            }
+            if (plainSizes.size() > encryptedSizes.size()) {
+                encryptedSizes.push_back(0);
+            }
+            uint32_t sizeofPlainSizes = sizeof(uint32_t) * plainSizes.size();
+            uint32_t sizeofEncryptedSizes = sizeof(uint32_t) * encryptedSizes.size();
+            meta.setData(kKeyPlainSizes, 0, plainSizes.data(), sizeofPlainSizes);
+            meta.setData(kKeyEncryptedSizes, 0, encryptedSizes.data(), sizeofEncryptedSizes);
+            mbuf->set_range(frameOffset, mbuf->range_length() - frameOffset);
+        } else {
+            /*
+             *  0                   1                   2                   3
+             *  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+             *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+             *  |  Signal Byte  |                                               |
+             *  +-+-+-+-+-+-+-+-+             IV                                |
+             *  |                                                               |
+             *  |               +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+             *  |               |                                               |
+             *  |-+-+-+-+-+-+-+-+                                               |
+             *  :               Bytes 1..N of encrypted frame                   :
+             *  |                                                               |
+             *  |                                                               |
+             *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+             */
+            int32_t plainSizes[] = { 0 };
+            int32_t encryptedSizes[] = { static_cast<int32_t>(mbuf->range_length() - 9) };
+            meta.setData(kKeyPlainSizes, 0, plainSizes, sizeof(plainSizes));
+            meta.setData(kKeyEncryptedSizes, 0, encryptedSizes, sizeof(encryptedSizes));
+            mbuf->set_range(9, mbuf->range_length() - 9);
+        }
+    } else {
+        /*
+         *  0                   1                   2                   3
+         *  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+         *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+         *  |  Signal Byte  |                                               |
+         *  +-+-+-+-+-+-+-+-+                                               |
+         *  :               Bytes 1..N of unencrypted frame                 :
+         *  |                                                               |
+         *  |                                                               |
+         *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+         */
+        int32_t plainSizes[] = { static_cast<int32_t>(mbuf->range_length() - 1) };
+        int32_t encryptedSizes[] = { 0 };
+        meta.setData(kKeyPlainSizes, 0, plainSizes, sizeof(plainSizes));
+        meta.setData(kKeyEncryptedSizes, 0, encryptedSizes, sizeof(encryptedSizes));
+        mbuf->set_range(1, mbuf->range_length() - 1);
+    }
+
+    return OK;
+}
+
+status_t MatroskaSource::readBlock() {
+    CHECK(mPendingFrames.empty());
+
+    if (mBlockIter.eos()) {
+        return ERROR_END_OF_STREAM;
+    }
+
+    const mkvparser::Block *block = mBlockIter.block();
+
+    int64_t timeUs = mBlockIter.blockTimeUs();
+
+    for (int i = 0; i < block->GetFrameCount(); ++i) {
+        MatroskaExtractor::TrackInfo *trackInfo = &mExtractor->mTracks.editItemAt(mTrackIndex);
+        const mkvparser::Block::Frame &frame = block->GetFrame(i);
+        size_t len = frame.len;
+        if (SIZE_MAX - len < trackInfo->mHeaderLen) {
+            return ERROR_MALFORMED;
+        }
+
+        len += trackInfo->mHeaderLen;
+        MediaBufferBase *mbuf = MediaBufferBase::Create(len);
+        uint8_t *data = static_cast<uint8_t *>(mbuf->data());
+        if (trackInfo->mHeader) {
+            memcpy(data, trackInfo->mHeader, trackInfo->mHeaderLen);
+        }
+
+        mbuf->meta_data().setInt64(kKeyTime, timeUs);
+        mbuf->meta_data().setInt32(kKeyIsSyncFrame, block->IsKey());
+
+        status_t err = frame.Read(mExtractor->mReader, data + trackInfo->mHeaderLen);
+        if (err == OK
+                && mExtractor->mIsWebm
+                && trackInfo->mEncrypted) {
+            err = setWebmBlockCryptoInfo(mbuf);
+        }
+
+        if (err != OK) {
+            mPendingFrames.clear();
+
+            mBlockIter.advance();
+            mbuf->release();
+            return err;
+        }
+
+        mPendingFrames.push_back(mbuf);
+    }
+
+    mBlockIter.advance();
+
+    return OK;
+}
+
+status_t MatroskaSource::read(
+        MediaBufferBase **out, const ReadOptions *options) {
+    *out = NULL;
+
+    int64_t targetSampleTimeUs = -1ll;
+
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode mode;
+    if (options && options->getSeekTo(&seekTimeUs, &mode)) {
+        if (mode == ReadOptions::SEEK_FRAME_INDEX) {
+            return ERROR_UNSUPPORTED;
+        }
+
+        if (!mExtractor->isLiveStreaming()) {
+            clearPendingFrames();
+
+            // The audio we want is located by using the Cues to seek the video
+            // stream to find the target Cluster then iterating to finalize for
+            // audio.
+            int64_t actualFrameTimeUs;
+            mBlockIter.seek(seekTimeUs, mIsAudio, &actualFrameTimeUs);
+            if (mode == ReadOptions::SEEK_CLOSEST) {
+                targetSampleTimeUs = actualFrameTimeUs;
+            }
+        }
+    }
+
+    while (mPendingFrames.empty()) {
+        status_t err = readBlock();
+
+        if (err != OK) {
+            clearPendingFrames();
+
+            return err;
+        }
+    }
+
+    MediaBufferBase *frame = *mPendingFrames.begin();
+    mPendingFrames.erase(mPendingFrames.begin());
+
+    if ((mType != AVC && mType != HEVC) || mNALSizeLen == 0) {
+        if (targetSampleTimeUs >= 0ll) {
+            frame->meta_data().setInt64(
+                    kKeyTargetTime, targetSampleTimeUs);
+        }
+
+        *out = frame;
+
+        return OK;
+    }
+
+    // Each input frame contains one or more NAL fragments, each fragment
+    // is prefixed by mNALSizeLen bytes giving the fragment length,
+    // followed by a corresponding number of bytes containing the fragment.
+    // We output all these fragments into a single large buffer separated
+    // by startcodes (0x00 0x00 0x00 0x01).
+    //
+    // When mNALSizeLen is 0, we assume the data is already in the format
+    // desired.
+
+    const uint8_t *srcPtr =
+        (const uint8_t *)frame->data() + frame->range_offset();
+
+    size_t srcSize = frame->range_length();
+
+    size_t dstSize = 0;
+    MediaBufferBase *buffer = NULL;
+    uint8_t *dstPtr = NULL;
+
+    for (int32_t pass = 0; pass < 2; ++pass) {
+        size_t srcOffset = 0;
+        size_t dstOffset = 0;
+        while (srcOffset + mNALSizeLen <= srcSize) {
+            size_t NALsize;
+            switch (mNALSizeLen) {
+                case 1: NALsize = srcPtr[srcOffset]; break;
+                case 2: NALsize = U16_AT(srcPtr + srcOffset); break;
+                case 3: NALsize = U24_AT(srcPtr + srcOffset); break;
+                case 4: NALsize = U32_AT(srcPtr + srcOffset); break;
+                default:
+                    TRESPASS();
+            }
+
+            if (srcOffset + mNALSizeLen + NALsize <= srcOffset + mNALSizeLen) {
+                frame->release();
+                frame = NULL;
+
+                return ERROR_MALFORMED;
+            } else if (srcOffset + mNALSizeLen + NALsize > srcSize) {
+                break;
+            }
+
+            if (pass == 1) {
+                memcpy(&dstPtr[dstOffset], "\x00\x00\x00\x01", 4);
+
+                if (frame != buffer) {
+                    memcpy(&dstPtr[dstOffset + 4],
+                           &srcPtr[srcOffset + mNALSizeLen],
+                           NALsize);
+                }
+            }
+
+            dstOffset += 4;  // 0x00 00 00 01
+            dstOffset += NALsize;
+
+            srcOffset += mNALSizeLen + NALsize;
+        }
+
+        if (srcOffset < srcSize) {
+            // There were trailing bytes or not enough data to complete
+            // a fragment.
+
+            frame->release();
+            frame = NULL;
+
+            return ERROR_MALFORMED;
+        }
+
+        if (pass == 0) {
+            dstSize = dstOffset;
+
+            if (dstSize == srcSize && mNALSizeLen == 4) {
+                // In this special case we can re-use the input buffer by substituting
+                // each 4-byte nal size with a 4-byte start code
+                buffer = frame;
+            } else {
+                buffer = MediaBufferBase::Create(dstSize);
+            }
+
+            int64_t timeUs;
+            CHECK(frame->meta_data().findInt64(kKeyTime, &timeUs));
+            int32_t isSync;
+            CHECK(frame->meta_data().findInt32(kKeyIsSyncFrame, &isSync));
+
+            buffer->meta_data().setInt64(kKeyTime, timeUs);
+            buffer->meta_data().setInt32(kKeyIsSyncFrame, isSync);
+
+            dstPtr = (uint8_t *)buffer->data();
+        }
+    }
+
+    if (frame != buffer) {
+        frame->release();
+        frame = NULL;
+    }
+
+    if (targetSampleTimeUs >= 0ll) {
+        buffer->meta_data().setInt64(
+                kKeyTargetTime, targetSampleTimeUs);
+    }
+
+    *out = buffer;
+
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+MatroskaExtractor::MatroskaExtractor(DataSourceBase *source)
+    : mDataSource(source),
+      mReader(new DataSourceBaseReader(mDataSource)),
+      mSegment(NULL),
+      mExtractedThumbnails(false),
+      mIsWebm(false),
+      mSeekPreRollNs(0) {
+    off64_t size;
+    mIsLiveStreaming =
+        (mDataSource->flags()
+            & (DataSourceBase::kWantsPrefetching
+                | DataSourceBase::kIsCachingDataSource))
+        && mDataSource->getSize(&size) != OK;
+
+    mkvparser::EBMLHeader ebmlHeader;
+    long long pos;
+    if (ebmlHeader.Parse(mReader, pos) < 0) {
+        return;
+    }
+
+    if (ebmlHeader.m_docType && !strcmp("webm", ebmlHeader.m_docType)) {
+        mIsWebm = true;
+    }
+
+    long long ret =
+        mkvparser::Segment::CreateInstance(mReader, pos, mSegment);
+
+    if (ret) {
+        CHECK(mSegment == NULL);
+        return;
+    }
+
+    // from mkvparser::Segment::Load(), but stop at first cluster
+    ret = mSegment->ParseHeaders();
+    if (ret == 0) {
+        long len;
+        ret = mSegment->LoadCluster(pos, len);
+        if (ret >= 1) {
+            // no more clusters
+            ret = 0;
+        }
+    } else if (ret > 0) {
+        ret = mkvparser::E_BUFFER_NOT_FULL;
+    }
+
+    if (ret < 0) {
+        char uri[1024];
+        if(!mDataSource->getUri(uri, sizeof(uri))) {
+            uri[0] = '\0';
+        }
+        ALOGW("Corrupt %s source: %s", mIsWebm ? "webm" : "matroska",
+                uriDebugString(uri).c_str());
+        delete mSegment;
+        mSegment = NULL;
+        return;
+    }
+
+#if 0
+    const mkvparser::SegmentInfo *info = mSegment->GetInfo();
+    ALOGI("muxing app: %s, writing app: %s",
+         info->GetMuxingAppAsUTF8(),
+         info->GetWritingAppAsUTF8());
+#endif
+
+    addTracks();
+}
+
+MatroskaExtractor::~MatroskaExtractor() {
+    delete mSegment;
+    mSegment = NULL;
+
+    delete mReader;
+    mReader = NULL;
+}
+
+size_t MatroskaExtractor::countTracks() {
+    return mTracks.size();
+}
+
+MediaTrack *MatroskaExtractor::getTrack(size_t index) {
+    if (index >= mTracks.size()) {
+        return NULL;
+    }
+
+    return new MatroskaSource(this, index);
+}
+
+status_t MatroskaExtractor::getTrackMetaData(
+        MetaDataBase &meta,
+        size_t index, uint32_t flags) {
+    if (index >= mTracks.size()) {
+        return UNKNOWN_ERROR;
+    }
+
+    if ((flags & kIncludeExtensiveMetaData) && !mExtractedThumbnails
+            && !isLiveStreaming()) {
+        findThumbnails();
+        mExtractedThumbnails = true;
+    }
+
+    meta = mTracks.itemAt(index).mMeta;
+    return OK;
+}
+
+bool MatroskaExtractor::isLiveStreaming() const {
+    return mIsLiveStreaming;
+}
+
+static int bytesForSize(size_t size) {
+    // use at most 28 bits (4 times 7)
+    CHECK(size <= 0xfffffff);
+
+    if (size > 0x1fffff) {
+        return 4;
+    } else if (size > 0x3fff) {
+        return 3;
+    } else if (size > 0x7f) {
+        return 2;
+    }
+    return 1;
+}
+
+static void storeSize(uint8_t *data, size_t &idx, size_t size) {
+    int numBytes = bytesForSize(size);
+    idx += numBytes;
+
+    data += idx;
+    size_t next = 0;
+    while (numBytes--) {
+        *--data = (size & 0x7f) | next;
+        size >>= 7;
+        next = 0x80;
+    }
+}
+
+static void addESDSFromCodecPrivate(
+        MetaDataBase &meta,
+        bool isAudio, const void *priv, size_t privSize) {
+
+    int privSizeBytesRequired = bytesForSize(privSize);
+    int esdsSize2 = 14 + privSizeBytesRequired + privSize;
+    int esdsSize2BytesRequired = bytesForSize(esdsSize2);
+    int esdsSize1 = 4 + esdsSize2BytesRequired + esdsSize2;
+    int esdsSize1BytesRequired = bytesForSize(esdsSize1);
+    size_t esdsSize = 1 + esdsSize1BytesRequired + esdsSize1;
+    uint8_t *esds = new uint8_t[esdsSize];
+
+    size_t idx = 0;
+    esds[idx++] = 0x03;
+    storeSize(esds, idx, esdsSize1);
+    esds[idx++] = 0x00; // ES_ID
+    esds[idx++] = 0x00; // ES_ID
+    esds[idx++] = 0x00; // streamDependenceFlag, URL_Flag, OCRstreamFlag
+    esds[idx++] = 0x04;
+    storeSize(esds, idx, esdsSize2);
+    esds[idx++] = isAudio ? 0x40   // Audio ISO/IEC 14496-3
+                          : 0x20;  // Visual ISO/IEC 14496-2
+    for (int i = 0; i < 12; i++) {
+        esds[idx++] = 0x00;
+    }
+    esds[idx++] = 0x05;
+    storeSize(esds, idx, privSize);
+    memcpy(esds + idx, priv, privSize);
+
+    meta.setData(kKeyESDS, 0, esds, esdsSize);
+
+    delete[] esds;
+    esds = NULL;
+}
+
+status_t addVorbisCodecInfo(
+        MetaDataBase &meta,
+        const void *_codecPrivate, size_t codecPrivateSize) {
+    // hexdump(_codecPrivate, codecPrivateSize);
+
+    if (codecPrivateSize < 1) {
+        return ERROR_MALFORMED;
+    }
+
+    const uint8_t *codecPrivate = (const uint8_t *)_codecPrivate;
+
+    if (codecPrivate[0] != 0x02) {
+        return ERROR_MALFORMED;
+    }
+
+    // codecInfo starts with two lengths, len1 and len2, that are
+    // "Xiph-style-lacing encoded"...
+
+    size_t offset = 1;
+    size_t len1 = 0;
+    while (offset < codecPrivateSize && codecPrivate[offset] == 0xff) {
+        if (len1 > (SIZE_MAX - 0xff)) {
+            return ERROR_MALFORMED; // would overflow
+        }
+        len1 += 0xff;
+        ++offset;
+    }
+    if (offset >= codecPrivateSize) {
+        return ERROR_MALFORMED;
+    }
+    if (len1 > (SIZE_MAX - codecPrivate[offset])) {
+        return ERROR_MALFORMED; // would overflow
+    }
+    len1 += codecPrivate[offset++];
+
+    size_t len2 = 0;
+    while (offset < codecPrivateSize && codecPrivate[offset] == 0xff) {
+        if (len2 > (SIZE_MAX - 0xff)) {
+            return ERROR_MALFORMED; // would overflow
+        }
+        len2 += 0xff;
+        ++offset;
+    }
+    if (offset >= codecPrivateSize) {
+        return ERROR_MALFORMED;
+    }
+    if (len2 > (SIZE_MAX - codecPrivate[offset])) {
+        return ERROR_MALFORMED; // would overflow
+    }
+    len2 += codecPrivate[offset++];
+
+    if (len1 > SIZE_MAX - len2 || offset > SIZE_MAX - (len1 + len2) ||
+            codecPrivateSize < offset + len1 + len2) {
+        return ERROR_MALFORMED;
+    }
+
+    if (codecPrivate[offset] != 0x01) {
+        return ERROR_MALFORMED;
+    }
+    meta.setData(kKeyVorbisInfo, 0, &codecPrivate[offset], len1);
+
+    offset += len1;
+    if (codecPrivate[offset] != 0x03) {
+        return ERROR_MALFORMED;
+    }
+
+    offset += len2;
+    if (codecPrivate[offset] != 0x05) {
+        return ERROR_MALFORMED;
+    }
+
+    meta.setData(
+            kKeyVorbisBooks, 0, &codecPrivate[offset],
+            codecPrivateSize - offset);
+
+    return OK;
+}
+
+static status_t addFlacMetadata(
+        MetaDataBase &meta,
+        const void *codecPrivate, size_t codecPrivateSize) {
+    // hexdump(codecPrivate, codecPrivateSize);
+
+    meta.setData(kKeyFlacMetadata, 0, codecPrivate, codecPrivateSize);
+
+    int32_t maxInputSize = 64 << 10;
+    FLACDecoder *flacDecoder = FLACDecoder::Create();
+    if (flacDecoder != NULL
+            && flacDecoder->parseMetadata((const uint8_t*)codecPrivate, codecPrivateSize) == OK) {
+        FLAC__StreamMetadata_StreamInfo streamInfo = flacDecoder->getStreamInfo();
+        maxInputSize = streamInfo.max_framesize;
+        if (maxInputSize == 0) {
+            // In case max framesize is not available, use raw data size as max framesize,
+            // assuming there is no expansion.
+            if (streamInfo.max_blocksize != 0
+                    && streamInfo.channels != 0
+                    && ((streamInfo.bits_per_sample + 7) / 8) >
+                        INT32_MAX / streamInfo.max_blocksize / streamInfo.channels) {
+                delete flacDecoder;
+                return ERROR_MALFORMED;
+            }
+            maxInputSize = ((streamInfo.bits_per_sample + 7) / 8)
+                * streamInfo.max_blocksize * streamInfo.channels;
+        }
+    }
+    meta.setInt32(kKeyMaxInputSize, maxInputSize);
+
+    delete flacDecoder;
+    return OK;
+}
+
+status_t MatroskaExtractor::synthesizeAVCC(TrackInfo *trackInfo, size_t index) {
+    BlockIterator iter(this, trackInfo->mTrackNum, index);
+    if (iter.eos()) {
+        return ERROR_MALFORMED;
+    }
+
+    const mkvparser::Block *block = iter.block();
+    if (block->GetFrameCount() <= 0) {
+        return ERROR_MALFORMED;
+    }
+
+    const mkvparser::Block::Frame &frame = block->GetFrame(0);
+    auto tmpData = heapbuffer<unsigned char>(frame.len);
+    long n = frame.Read(mReader, tmpData.get());
+    if (n != 0) {
+        return ERROR_MALFORMED;
+    }
+
+    if (!MakeAVCCodecSpecificData(trackInfo->mMeta, tmpData.get(), frame.len)) {
+        return ERROR_MALFORMED;
+    }
+
+    // Override the synthesized nal length size, which is arbitrary
+    trackInfo->mMeta.setInt32(kKeyNalLengthSize, 0);
+    return OK;
+}
+
+static inline bool isValidInt32ColourValue(long long value) {
+    return value != mkvparser::Colour::kValueNotPresent
+            && value >= INT32_MIN
+            && value <= INT32_MAX;
+}
+
+static inline bool isValidUint16ColourValue(long long value) {
+    return value != mkvparser::Colour::kValueNotPresent
+            && value >= 0
+            && value <= UINT16_MAX;
+}
+
+static inline bool isValidPrimary(const mkvparser::PrimaryChromaticity *primary) {
+    return primary != NULL && primary->x >= 0 && primary->x <= 1
+             && primary->y >= 0 && primary->y <= 1;
+}
+
+void MatroskaExtractor::getColorInformation(
+        const mkvparser::VideoTrack *vtrack, MetaDataBase &meta) {
+    const mkvparser::Colour *color = vtrack->GetColour();
+    if (color == NULL) {
+        return;
+    }
+
+    // Color Aspects
+    {
+        int32_t primaries = 2; // ISO unspecified
+        int32_t transfer = 2; // ISO unspecified
+        int32_t coeffs = 2; // ISO unspecified
+        bool fullRange = false; // default
+        bool rangeSpecified = false;
+
+        if (isValidInt32ColourValue(color->primaries)) {
+            primaries = color->primaries;
+        }
+        if (isValidInt32ColourValue(color->transfer_characteristics)) {
+            transfer = color->transfer_characteristics;
+        }
+        if (isValidInt32ColourValue(color->matrix_coefficients)) {
+            coeffs = color->matrix_coefficients;
+        }
+        if (color->range != mkvparser::Colour::kValueNotPresent
+                && color->range != 0 /* MKV unspecified */) {
+            // We only support MKV broadcast range (== limited) and full range.
+            // We treat all other value as the default limited range.
+            fullRange = color->range == 2 /* MKV fullRange */;
+            rangeSpecified = true;
+        }
+
+        ColorAspects aspects;
+        ColorUtils::convertIsoColorAspectsToCodecAspects(
+                primaries, transfer, coeffs, fullRange, aspects);
+        meta.setInt32(kKeyColorPrimaries, aspects.mPrimaries);
+        meta.setInt32(kKeyTransferFunction, aspects.mTransfer);
+        meta.setInt32(kKeyColorMatrix, aspects.mMatrixCoeffs);
+        meta.setInt32(
+                kKeyColorRange, rangeSpecified ? aspects.mRange : ColorAspects::RangeUnspecified);
+    }
+
+    // HDR Static Info
+    {
+        HDRStaticInfo info, nullInfo; // nullInfo is a fully unspecified static info
+        memset(&info, 0, sizeof(info));
+        memset(&nullInfo, 0, sizeof(nullInfo));
+        if (isValidUint16ColourValue(color->max_cll)) {
+            info.sType1.mMaxContentLightLevel = color->max_cll;
+        }
+        if (isValidUint16ColourValue(color->max_fall)) {
+            info.sType1.mMaxFrameAverageLightLevel = color->max_fall;
+        }
+        const mkvparser::MasteringMetadata *mastering = color->mastering_metadata;
+        if (mastering != NULL) {
+            // Convert matroska values to HDRStaticInfo equivalent values for each fully specified
+            // group. See CTA-681.3 section 3.2.1 for more info.
+            if (mastering->luminance_max >= 0.5 && mastering->luminance_max < 65535.5) {
+                info.sType1.mMaxDisplayLuminance = (uint16_t)(mastering->luminance_max + 0.5);
+            }
+            if (mastering->luminance_min >= 0.00005 && mastering->luminance_min < 6.55355) {
+                // HDRStaticInfo Type1 stores min luminance scaled 10000:1
+                info.sType1.mMinDisplayLuminance =
+                    (uint16_t)(10000 * mastering->luminance_min + 0.5);
+            }
+            // HDRStaticInfo Type1 stores primaries scaled 50000:1
+            if (isValidPrimary(mastering->white_point)) {
+                info.sType1.mW.x = (uint16_t)(50000 * mastering->white_point->x + 0.5);
+                info.sType1.mW.y = (uint16_t)(50000 * mastering->white_point->y + 0.5);
+            }
+            if (isValidPrimary(mastering->r) && isValidPrimary(mastering->g)
+                    && isValidPrimary(mastering->b)) {
+                info.sType1.mR.x = (uint16_t)(50000 * mastering->r->x + 0.5);
+                info.sType1.mR.y = (uint16_t)(50000 * mastering->r->y + 0.5);
+                info.sType1.mG.x = (uint16_t)(50000 * mastering->g->x + 0.5);
+                info.sType1.mG.y = (uint16_t)(50000 * mastering->g->y + 0.5);
+                info.sType1.mB.x = (uint16_t)(50000 * mastering->b->x + 0.5);
+                info.sType1.mB.y = (uint16_t)(50000 * mastering->b->y + 0.5);
+            }
+        }
+        // Only advertise static info if at least one of the groups have been specified.
+        if (memcmp(&info, &nullInfo, sizeof(info)) != 0) {
+            info.mID = HDRStaticInfo::kType1;
+            meta.setData(kKeyHdrStaticInfo, 'hdrS', &info, sizeof(info));
+        }
+    }
+}
+
+status_t MatroskaExtractor::initTrackInfo(
+        const mkvparser::Track *track, MetaDataBase &meta, TrackInfo *trackInfo) {
+    trackInfo->mTrackNum = track->GetNumber();
+    trackInfo->mMeta = meta;
+    trackInfo->mExtractor = this;
+    trackInfo->mEncrypted = false;
+    trackInfo->mHeader = NULL;
+    trackInfo->mHeaderLen = 0;
+
+    for(size_t i = 0; i < track->GetContentEncodingCount(); i++) {
+        const mkvparser::ContentEncoding *encoding = track->GetContentEncodingByIndex(i);
+        for(size_t j = 0; j < encoding->GetEncryptionCount(); j++) {
+            const mkvparser::ContentEncoding::ContentEncryption *encryption;
+            encryption = encoding->GetEncryptionByIndex(j);
+            trackInfo->mMeta.setData(kKeyCryptoKey, 0, encryption->key_id, encryption->key_id_len);
+            trackInfo->mEncrypted = true;
+            break;
+        }
+
+        for(size_t j = 0; j < encoding->GetCompressionCount(); j++) {
+            const mkvparser::ContentEncoding::ContentCompression *compression;
+            compression = encoding->GetCompressionByIndex(j);
+            ALOGV("compression algo %llu settings_len %lld",
+                compression->algo, compression->settings_len);
+            if (compression->algo == 3
+                    && compression->settings
+                    && compression->settings_len > 0) {
+                trackInfo->mHeader = compression->settings;
+                trackInfo->mHeaderLen = compression->settings_len;
+            }
+        }
+    }
+
+    return OK;
+}
+
+void MatroskaExtractor::addTracks() {
+    const mkvparser::Tracks *tracks = mSegment->GetTracks();
+
+    for (size_t index = 0; index < tracks->GetTracksCount(); ++index) {
+        const mkvparser::Track *track = tracks->GetTrackByIndex(index);
+
+        if (track == NULL) {
+            // Apparently this is currently valid (if unexpected) behaviour
+            // of the mkv parser lib.
+            continue;
+        }
+
+        const char *const codecID = track->GetCodecId();
+        ALOGV("codec id = %s", codecID);
+        ALOGV("codec name = %s", track->GetCodecNameAsUTF8());
+
+        if (codecID == NULL) {
+            ALOGW("unknown codecID is not supported.");
+            continue;
+        }
+
+        size_t codecPrivateSize;
+        const unsigned char *codecPrivate =
+            track->GetCodecPrivate(codecPrivateSize);
+
+        enum { VIDEO_TRACK = 1, AUDIO_TRACK = 2 };
+
+        MetaDataBase meta;
+
+        status_t err = OK;
+
+        switch (track->GetType()) {
+            case VIDEO_TRACK:
+            {
+                const mkvparser::VideoTrack *vtrack =
+                    static_cast<const mkvparser::VideoTrack *>(track);
+
+                if (!strcmp("V_MPEG4/ISO/AVC", codecID)) {
+                    meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
+                    meta.setData(kKeyAVCC, 0, codecPrivate, codecPrivateSize);
+                } else if (!strcmp("V_MPEGH/ISO/HEVC", codecID)) {
+                    meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_HEVC);
+                    if (codecPrivateSize > 0) {
+                        meta.setData(kKeyHVCC, kTypeHVCC, codecPrivate, codecPrivateSize);
+                    } else {
+                        ALOGW("HEVC is detected, but does not have configuration.");
+                        continue;
+                    }
+                } else if (!strcmp("V_MPEG4/ISO/ASP", codecID)) {
+                    if (codecPrivateSize > 0) {
+                        meta.setCString(
+                                kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG4);
+                        addESDSFromCodecPrivate(
+                                meta, false, codecPrivate, codecPrivateSize);
+                    } else {
+                        ALOGW("%s is detected, but does not have configuration.",
+                                codecID);
+                        continue;
+                    }
+                } else if (!strcmp("V_VP8", codecID)) {
+                    meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_VP8);
+                } else if (!strcmp("V_VP9", codecID)) {
+                    meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_VP9);
+                    if (codecPrivateSize > 0) {
+                      // 'csd-0' for VP9 is the Blob of Codec Private data as
+                      // specified in http://www.webmproject.org/vp9/profiles/.
+                        meta.setData(
+                              kKeyVp9CodecPrivate, 0, codecPrivate,
+                              codecPrivateSize);
+                    }
+                } else {
+                    ALOGW("%s is not supported.", codecID);
+                    continue;
+                }
+
+                const long long width = vtrack->GetWidth();
+                const long long height = vtrack->GetHeight();
+                if (width <= 0 || width > INT32_MAX) {
+                    ALOGW("track width exceeds int32_t, %lld", width);
+                    continue;
+                }
+                if (height <= 0 || height > INT32_MAX) {
+                    ALOGW("track height exceeds int32_t, %lld", height);
+                    continue;
+                }
+                meta.setInt32(kKeyWidth, (int32_t)width);
+                meta.setInt32(kKeyHeight, (int32_t)height);
+
+                // setting display width/height is optional
+                const long long displayUnit = vtrack->GetDisplayUnit();
+                const long long displayWidth = vtrack->GetDisplayWidth();
+                const long long displayHeight = vtrack->GetDisplayHeight();
+                if (displayWidth > 0 && displayWidth <= INT32_MAX
+                        && displayHeight > 0 && displayHeight <= INT32_MAX) {
+                    switch (displayUnit) {
+                    case 0: // pixels
+                        meta.setInt32(kKeyDisplayWidth, (int32_t)displayWidth);
+                        meta.setInt32(kKeyDisplayHeight, (int32_t)displayHeight);
+                        break;
+                    case 1: // centimeters
+                    case 2: // inches
+                    case 3: // aspect ratio
+                    {
+                        // Physical layout size is treated the same as aspect ratio.
+                        // Note: displayWidth and displayHeight are never zero as they are
+                        // checked in the if above.
+                        const long long computedWidth =
+                                std::max(width, height * displayWidth / displayHeight);
+                        const long long computedHeight =
+                                std::max(height, width * displayHeight / displayWidth);
+                        if (computedWidth <= INT32_MAX && computedHeight <= INT32_MAX) {
+                            meta.setInt32(kKeyDisplayWidth, (int32_t)computedWidth);
+                            meta.setInt32(kKeyDisplayHeight, (int32_t)computedHeight);
+                        }
+                        break;
+                    }
+                    default: // unknown display units, perhaps future version of spec.
+                        break;
+                    }
+                }
+
+                getColorInformation(vtrack, meta);
+
+                break;
+            }
+
+            case AUDIO_TRACK:
+            {
+                const mkvparser::AudioTrack *atrack =
+                    static_cast<const mkvparser::AudioTrack *>(track);
+
+                if (!strcmp("A_AAC", codecID)) {
+                    meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AAC);
+                    CHECK(codecPrivateSize >= 2);
+
+                    addESDSFromCodecPrivate(
+                            meta, true, codecPrivate, codecPrivateSize);
+                } else if (!strcmp("A_VORBIS", codecID)) {
+                    meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_VORBIS);
+
+                    err = addVorbisCodecInfo(
+                            meta, codecPrivate, codecPrivateSize);
+                } else if (!strcmp("A_OPUS", codecID)) {
+                    meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_OPUS);
+                    meta.setData(kKeyOpusHeader, 0, codecPrivate, codecPrivateSize);
+                    meta.setInt64(kKeyOpusCodecDelay, track->GetCodecDelay());
+                    meta.setInt64(kKeyOpusSeekPreRoll, track->GetSeekPreRoll());
+                    mSeekPreRollNs = track->GetSeekPreRoll();
+                } else if (!strcmp("A_MPEG/L3", codecID)) {
+                    meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
+                } else if (!strcmp("A_FLAC", codecID)) {
+                    meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_FLAC);
+                    err = addFlacMetadata(meta, codecPrivate, codecPrivateSize);
+                } else {
+                    ALOGW("%s is not supported.", codecID);
+                    continue;
+                }
+
+                meta.setInt32(kKeySampleRate, atrack->GetSamplingRate());
+                meta.setInt32(kKeyChannelCount, atrack->GetChannels());
+                break;
+            }
+
+            default:
+                continue;
+        }
+
+        const char *language = track->GetLanguage();
+        if (language != NULL) {
+           char lang[4];
+           strncpy(lang, language, 3);
+           lang[3] = '\0';
+           meta.setCString(kKeyMediaLanguage, lang);
+        }
+
+        if (err != OK) {
+            ALOGE("skipping track, codec specific data was malformed.");
+            continue;
+        }
+
+        long long durationNs = mSegment->GetDuration();
+        meta.setInt64(kKeyDuration, (durationNs + 500) / 1000);
+
+        mTracks.push();
+        size_t n = mTracks.size() - 1;
+        TrackInfo *trackInfo = &mTracks.editItemAt(n);
+        initTrackInfo(track, meta, trackInfo);
+
+        if (!strcmp("V_MPEG4/ISO/AVC", codecID) && codecPrivateSize == 0) {
+            // Attempt to recover from AVC track without codec private data
+            err = synthesizeAVCC(trackInfo, n);
+            if (err != OK) {
+                mTracks.pop();
+            }
+        }
+    }
+}
+
+void MatroskaExtractor::findThumbnails() {
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        TrackInfo *info = &mTracks.editItemAt(i);
+
+        const char *mime;
+        CHECK(info->mMeta.findCString(kKeyMIMEType, &mime));
+
+        if (strncasecmp(mime, "video/", 6)) {
+            continue;
+        }
+
+        BlockIterator iter(this, info->mTrackNum, i);
+        int32_t j = 0;
+        int64_t thumbnailTimeUs = 0;
+        size_t maxBlockSize = 0;
+        while (!iter.eos() && j < 20) {
+            if (iter.block()->IsKey()) {
+                ++j;
+
+                size_t blockSize = 0;
+                for (int k = 0; k < iter.block()->GetFrameCount(); ++k) {
+                    blockSize += iter.block()->GetFrame(k).len;
+                }
+
+                if (blockSize > maxBlockSize) {
+                    maxBlockSize = blockSize;
+                    thumbnailTimeUs = iter.blockTimeUs();
+                }
+            }
+            iter.advance();
+        }
+        info->mMeta.setInt64(kKeyThumbnailTime, thumbnailTimeUs);
+    }
+}
+
+status_t MatroskaExtractor::getMetaData(MetaDataBase &meta) {
+    meta.setCString(
+            kKeyMIMEType,
+            mIsWebm ? "video/webm" : MEDIA_MIMETYPE_CONTAINER_MATROSKA);
+
+    return OK;
+}
+
+uint32_t MatroskaExtractor::flags() const {
+    uint32_t x = CAN_PAUSE;
+    if (!isLiveStreaming()) {
+        x |= CAN_SEEK_BACKWARD | CAN_SEEK_FORWARD | CAN_SEEK;
+    }
+
+    return x;
+}
+
+bool SniffMatroska(
+        DataSourceBase *source, float *confidence) {
+    DataSourceBaseReader reader(source);
+    mkvparser::EBMLHeader ebmlHeader;
+    long long pos;
+    if (ebmlHeader.Parse(&reader, pos) < 0) {
+        return false;
+    }
+
+    *confidence = 0.6;
+
+    return true;
+}
+
+
+extern "C" {
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
+MediaExtractor::ExtractorDef GETEXTRACTORDEF() {
+    return {
+        MediaExtractor::EXTRACTORDEF_VERSION,
+        UUID("abbedd92-38c4-4904-a4c1-b3f45f899980"),
+        1,
+        "Matroska Extractor",
+        [](
+                DataSourceBase *source,
+                float *confidence,
+                void **,
+                MediaExtractor::FreeMetaFunc *) -> MediaExtractor::CreatorFunc {
+            if (SniffMatroska(source, confidence)) {
+                return [](
+                        DataSourceBase *source,
+                        void *) -> MediaExtractor* {
+                    return new MatroskaExtractor(source);};
+            }
+            return NULL;
+        }
+    };
+}
+
+} // extern "C"
+
+}  // namespace android

diff --git a/media/extractors/mkv/MatroskaExtractor.h b/media/extractors/mkv/MatroskaExtractor.h
new file mode 100644
index 0000000..3568ea1
--- /dev/null
+++ b/media/extractors/mkv/MatroskaExtractor.h

@@ -0,0 +1,105 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MATROSKA_EXTRACTOR_H_
+
+#define MATROSKA_EXTRACTOR_H_
+
+#include "mkvparser/mkvparser.h"
+
+#include <media/MediaExtractor.h>
+#include <media/stagefright/MetaDataBase.h>
+#include <utils/Vector.h>
+#include <utils/threads.h>
+
+namespace android {
+
+struct AMessage;
+class String8;
+
+class MetaData;
+struct DataSourceBaseReader;
+struct MatroskaSource;
+
+struct MatroskaExtractor : public MediaExtractor {
+    explicit MatroskaExtractor(DataSourceBase *source);
+
+    virtual size_t countTracks();
+
+    virtual MediaTrack *getTrack(size_t index);
+
+    virtual status_t getTrackMetaData(MetaDataBase& meta, size_t index, uint32_t flags);
+
+    virtual status_t getMetaData(MetaDataBase& meta);
+
+    virtual uint32_t flags() const;
+
+    virtual const char * name() { return "MatroskaExtractor"; }
+
+protected:
+    virtual ~MatroskaExtractor();
+
+private:
+    friend struct MatroskaSource;
+    friend struct BlockIterator;
+
+    struct TrackInfo {
+        unsigned long mTrackNum;
+        bool mEncrypted;
+        MetaDataBase mMeta;
+        const MatroskaExtractor *mExtractor;
+        Vector<const mkvparser::CuePoint*> mCuePoints;
+
+        // mHeader points to memory managed by mkvparser;
+        // mHeader would be deleted when mSegment is deleted
+        // in ~MatroskaExtractor.
+        unsigned char *mHeader;
+        size_t mHeaderLen;
+
+        const mkvparser::Track* getTrack() const;
+        const mkvparser::CuePoint::TrackPosition *find(long long timeNs) const;
+    };
+
+    Mutex mLock;
+    Vector<TrackInfo> mTracks;
+
+    DataSourceBase *mDataSource;
+    DataSourceBaseReader *mReader;
+    mkvparser::Segment *mSegment;
+    bool mExtractedThumbnails;
+    bool mIsLiveStreaming;
+    bool mIsWebm;
+    int64_t mSeekPreRollNs;
+
+    status_t synthesizeAVCC(TrackInfo *trackInfo, size_t index);
+    status_t initTrackInfo(
+            const mkvparser::Track *track,
+            MetaDataBase &meta,
+            TrackInfo *trackInfo);
+    void addTracks();
+    void findThumbnails();
+    void getColorInformation(
+            const mkvparser::VideoTrack *vtrack,
+            MetaDataBase &meta);
+    bool isLiveStreaming() const;
+
+    MatroskaExtractor(const MatroskaExtractor &);
+    MatroskaExtractor &operator=(const MatroskaExtractor &);
+};
+
+}  // namespace android
+
+#endif  // MATROSKA_EXTRACTOR_H_

diff --git a/media/libstagefright/matroska/NOTICE b/media/extractors/mkv/NOTICE
similarity index 100%
copy from media/libstagefright/matroska/NOTICE
copy to media/extractors/mkv/NOTICE


diff --git a/media/extractors/mkv/exports.lds b/media/extractors/mkv/exports.lds
new file mode 100644
index 0000000..b1309ee
--- /dev/null
+++ b/media/extractors/mkv/exports.lds

@@ -0,0 +1 @@
+{ global: GETEXTRACTORDEF; local: *; };

diff --git a/media/extractors/mp3/Android.bp b/media/extractors/mp3/Android.bp
new file mode 100644
index 0000000..a3aeaca
--- /dev/null
+++ b/media/extractors/mp3/Android.bp

@@ -0,0 +1,47 @@
+cc_library_shared {
+
+    srcs: [
+            "MP3Extractor.cpp",
+            "VBRISeeker.cpp",
+            "XINGSeeker.cpp",
+    ],
+
+    include_dirs: [
+        "frameworks/av/media/libstagefright/include",
+    ],
+
+    shared_libs: [
+        "liblog",
+        "libmediaextractor",
+        "libstagefright_foundation",
+    ],
+
+    static_libs: [
+        "libutils",
+        "libstagefright_id3",
+    ],
+
+    name: "libmp3extractor",
+    relative_install_path: "extractors",
+
+    compile_multilib: "first",
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+        "-fvisibility=hidden",
+    ],
+    version_script: "exports.lds",
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+        diag: {
+            cfi: true,
+        },
+    },
+
+}

diff --git a/media/extractors/mp3/MP3Extractor.cpp b/media/extractors/mp3/MP3Extractor.cpp
new file mode 100644
index 0000000..33cff96
--- /dev/null
+++ b/media/extractors/mp3/MP3Extractor.cpp

@@ -0,0 +1,725 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MP3Extractor"
+#include <utils/Log.h>
+
+#include "MP3Extractor.h"
+
+#include "ID3.h"
+#include "VBRISeeker.h"
+#include "XINGSeeker.h"
+
+#include <media/DataSourceBase.h>
+#include <media/MediaTrack.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/avc_utils.h>
+#include <media/stagefright/foundation/ByteUtils.h>
+#include <media/stagefright/MediaBufferBase.h>
+#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <utils/String8.h>
+
+namespace android {
+
+// Everything must match except for
+// protection, bitrate, padding, private bits, mode, mode extension,
+// copyright bit, original bit and emphasis.
+// Yes ... there are things that must indeed match...
+static const uint32_t kMask = 0xfffe0c00;
+
+static bool Resync(
+        DataSourceBase *source, uint32_t match_header,
+        off64_t *inout_pos, off64_t *post_id3_pos, uint32_t *out_header) {
+    if (post_id3_pos != NULL) {
+        *post_id3_pos = 0;
+    }
+
+    if (*inout_pos == 0) {
+        // Skip an optional ID3 header if syncing at the very beginning
+        // of the datasource.
+
+        for (;;) {
+            uint8_t id3header[10];
+            if (source->readAt(*inout_pos, id3header, sizeof(id3header))
+                    < (ssize_t)sizeof(id3header)) {
+                // If we can't even read these 10 bytes, we might as well bail
+                // out, even if there _were_ 10 bytes of valid mp3 audio data...
+                return false;
+            }
+
+            if (memcmp("ID3", id3header, 3)) {
+                break;
+            }
+
+            // Skip the ID3v2 header.
+
+            size_t len =
+                ((id3header[6] & 0x7f) << 21)
+                | ((id3header[7] & 0x7f) << 14)
+                | ((id3header[8] & 0x7f) << 7)
+                | (id3header[9] & 0x7f);
+
+            len += 10;
+
+            *inout_pos += len;
+
+            ALOGV("skipped ID3 tag, new starting offset is %lld (0x%016llx)",
+                    (long long)*inout_pos, (long long)*inout_pos);
+        }
+
+        if (post_id3_pos != NULL) {
+            *post_id3_pos = *inout_pos;
+        }
+    }
+
+    off64_t pos = *inout_pos;
+    bool valid = false;
+
+    const size_t kMaxReadBytes = 1024;
+    const size_t kMaxBytesChecked = 128 * 1024;
+    uint8_t buf[kMaxReadBytes];
+    ssize_t bytesToRead = kMaxReadBytes;
+    ssize_t totalBytesRead = 0;
+    ssize_t remainingBytes = 0;
+    bool reachEOS = false;
+    uint8_t *tmp = buf;
+
+    do {
+        if (pos >= (off64_t)(*inout_pos + kMaxBytesChecked)) {
+            // Don't scan forever.
+            ALOGV("giving up at offset %lld", (long long)pos);
+            break;
+        }
+
+        if (remainingBytes < 4) {
+            if (reachEOS) {
+                break;
+            } else {
+                memcpy(buf, tmp, remainingBytes);
+                bytesToRead = kMaxReadBytes - remainingBytes;
+
+                /*
+                 * The next read position should start from the end of
+                 * the last buffer, and thus should include the remaining
+                 * bytes in the buffer.
+                 */
+                totalBytesRead = source->readAt(pos + remainingBytes,
+                                                buf + remainingBytes,
+                                                bytesToRead);
+                if (totalBytesRead <= 0) {
+                    break;
+                }
+                reachEOS = (totalBytesRead != bytesToRead);
+                totalBytesRead += remainingBytes;
+                remainingBytes = totalBytesRead;
+                tmp = buf;
+                continue;
+            }
+        }
+
+        uint32_t header = U32_AT(tmp);
+
+        if (match_header != 0 && (header & kMask) != (match_header & kMask)) {
+            ++pos;
+            ++tmp;
+            --remainingBytes;
+            continue;
+        }
+
+        size_t frame_size;
+        int sample_rate, num_channels, bitrate;
+        if (!GetMPEGAudioFrameSize(
+                    header, &frame_size,
+                    &sample_rate, &num_channels, &bitrate)) {
+            ++pos;
+            ++tmp;
+            --remainingBytes;
+            continue;
+        }
+
+        ALOGV("found possible 1st frame at %lld (header = 0x%08x)", (long long)pos, header);
+
+        // We found what looks like a valid frame,
+        // now find its successors.
+
+        off64_t test_pos = pos + frame_size;
+
+        valid = true;
+        for (int j = 0; j < 3; ++j) {
+            uint8_t tmp[4];
+            if (source->readAt(test_pos, tmp, 4) < 4) {
+                valid = false;
+                break;
+            }
+
+            uint32_t test_header = U32_AT(tmp);
+
+            ALOGV("subsequent header is %08x", test_header);
+
+            if ((test_header & kMask) != (header & kMask)) {
+                valid = false;
+                break;
+            }
+
+            size_t test_frame_size;
+            if (!GetMPEGAudioFrameSize(
+                        test_header, &test_frame_size)) {
+                valid = false;
+                break;
+            }
+
+            ALOGV("found subsequent frame #%d at %lld", j + 2, (long long)test_pos);
+
+            test_pos += test_frame_size;
+        }
+
+        if (valid) {
+            *inout_pos = pos;
+
+            if (out_header != NULL) {
+                *out_header = header;
+            }
+        } else {
+            ALOGV("no dice, no valid sequence of frames found.");
+        }
+
+        ++pos;
+        ++tmp;
+        --remainingBytes;
+    } while (!valid);
+
+    return valid;
+}
+
+class MP3Source : public MediaTrack {
+public:
+    MP3Source(
+            MetaDataBase &meta, DataSourceBase *source,
+            off64_t first_frame_pos, uint32_t fixed_header,
+            MP3Seeker *seeker);
+
+    virtual status_t start(MetaDataBase *params = NULL);
+    virtual status_t stop();
+
+    virtual status_t getFormat(MetaDataBase &meta);
+
+    virtual status_t read(
+            MediaBufferBase **buffer, const ReadOptions *options = NULL);
+
+protected:
+    virtual ~MP3Source();
+
+private:
+    static const size_t kMaxFrameSize;
+    MetaDataBase &mMeta;
+    DataSourceBase *mDataSource;
+    off64_t mFirstFramePos;
+    uint32_t mFixedHeader;
+    off64_t mCurrentPos;
+    int64_t mCurrentTimeUs;
+    bool mStarted;
+    MP3Seeker *mSeeker;
+    MediaBufferGroup *mGroup;
+
+    int64_t mBasisTimeUs;
+    int64_t mSamplesRead;
+
+    MP3Source(const MP3Source &);
+    MP3Source &operator=(const MP3Source &);
+};
+
+struct Mp3Meta {
+    off64_t pos;
+    off64_t post_id3_pos;
+    uint32_t header;
+};
+
+MP3Extractor::MP3Extractor(
+        DataSourceBase *source, Mp3Meta *meta)
+    : mInitCheck(NO_INIT),
+      mDataSource(source),
+      mFirstFramePos(-1),
+      mFixedHeader(0),
+      mSeeker(NULL) {
+
+    off64_t pos = 0;
+    off64_t post_id3_pos;
+    uint32_t header;
+    bool success;
+
+    if (meta != NULL) {
+        // The sniffer has already done all the hard work for us, simply
+        // accept its judgement.
+        pos = meta->pos;
+        header = meta->header;
+        post_id3_pos = meta->post_id3_pos;
+        success = true;
+    } else {
+        success = Resync(mDataSource, 0, &pos, &post_id3_pos, &header);
+    }
+
+    if (!success) {
+        // mInitCheck will remain NO_INIT
+        return;
+    }
+
+    mFirstFramePos = pos;
+    mFixedHeader = header;
+    XINGSeeker *seeker = XINGSeeker::CreateFromSource(mDataSource, mFirstFramePos);
+
+    if (seeker == NULL) {
+        mSeeker = VBRISeeker::CreateFromSource(mDataSource, post_id3_pos);
+    } else {
+        mSeeker = seeker;
+        int encd = seeker->getEncoderDelay();
+        int encp = seeker->getEncoderPadding();
+        if (encd != 0 || encp != 0) {
+            mMeta.setInt32(kKeyEncoderDelay, encd);
+            mMeta.setInt32(kKeyEncoderPadding, encp);
+        }
+    }
+
+    if (mSeeker != NULL) {
+        // While it is safe to send the XING/VBRI frame to the decoder, this will
+        // result in an extra 1152 samples being output. In addition, the bitrate
+        // of the Xing header might not match the rest of the file, which could
+        // lead to problems when seeking. The real first frame to decode is after
+        // the XING/VBRI frame, so skip there.
+        size_t frame_size;
+        int sample_rate;
+        int num_channels;
+        int bitrate;
+        GetMPEGAudioFrameSize(
+                header, &frame_size, &sample_rate, &num_channels, &bitrate);
+        pos += frame_size;
+        if (!Resync(mDataSource, 0, &pos, &post_id3_pos, &header)) {
+            // mInitCheck will remain NO_INIT
+            return;
+        }
+        mFirstFramePos = pos;
+        mFixedHeader = header;
+    }
+
+    size_t frame_size;
+    int sample_rate;
+    int num_channels;
+    int bitrate;
+    GetMPEGAudioFrameSize(
+            header, &frame_size, &sample_rate, &num_channels, &bitrate);
+
+    unsigned layer = 4 - ((header >> 17) & 3);
+
+    switch (layer) {
+        case 1:
+            mMeta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG_LAYER_I);
+            break;
+        case 2:
+            mMeta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG_LAYER_II);
+            break;
+        case 3:
+            mMeta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
+            break;
+        default:
+            TRESPASS();
+    }
+
+    mMeta.setInt32(kKeySampleRate, sample_rate);
+    mMeta.setInt32(kKeyBitRate, bitrate * 1000);
+    mMeta.setInt32(kKeyChannelCount, num_channels);
+
+    int64_t durationUs;
+
+    if (mSeeker == NULL || !mSeeker->getDuration(&durationUs)) {
+        off64_t fileSize;
+        if (mDataSource->getSize(&fileSize) == OK) {
+            off64_t dataLength = fileSize - mFirstFramePos;
+            if (dataLength > INT64_MAX / 8000LL) {
+                // duration would overflow
+                durationUs = INT64_MAX;
+            } else {
+                durationUs = 8000LL * dataLength / bitrate;
+            }
+        } else {
+            durationUs = -1;
+        }
+    }
+
+    if (durationUs >= 0) {
+        mMeta.setInt64(kKeyDuration, durationUs);
+    }
+
+    mInitCheck = OK;
+
+    // Get iTunes-style gapless info if present.
+    // When getting the id3 tag, skip the V1 tags to prevent the source cache
+    // from being iterated to the end of the file.
+    ID3 id3(mDataSource, true);
+    if (id3.isValid()) {
+        ID3::Iterator *com = new ID3::Iterator(id3, "COM");
+        if (com->done()) {
+            delete com;
+            com = new ID3::Iterator(id3, "COMM");
+        }
+        while(!com->done()) {
+            String8 commentdesc;
+            String8 commentvalue;
+            com->getString(&commentdesc, &commentvalue);
+            const char * desc = commentdesc.string();
+            const char * value = commentvalue.string();
+
+            // first 3 characters are the language, which we don't care about
+            if(strlen(desc) > 3 && strcmp(desc + 3, "iTunSMPB") == 0) {
+
+                int32_t delay, padding;
+                if (sscanf(value, " %*x %x %x %*x", &delay, &padding) == 2) {
+                    mMeta.setInt32(kKeyEncoderDelay, delay);
+                    mMeta.setInt32(kKeyEncoderPadding, padding);
+                }
+                break;
+            }
+            com->next();
+        }
+        delete com;
+        com = NULL;
+    }
+}
+
+MP3Extractor::~MP3Extractor() {
+    delete mSeeker;
+}
+
+size_t MP3Extractor::countTracks() {
+    return mInitCheck != OK ? 0 : 1;
+}
+
+MediaTrack *MP3Extractor::getTrack(size_t index) {
+    if (mInitCheck != OK || index != 0) {
+        return NULL;
+    }
+
+    return new MP3Source(
+            mMeta, mDataSource, mFirstFramePos, mFixedHeader,
+            mSeeker);
+}
+
+status_t MP3Extractor::getTrackMetaData(
+        MetaDataBase &meta,
+        size_t index, uint32_t /* flags */) {
+    if (mInitCheck != OK || index != 0) {
+        return UNKNOWN_ERROR;
+    }
+    meta = mMeta;
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+// The theoretical maximum frame size for an MPEG audio stream should occur
+// while playing a Layer 2, MPEGv2.5 audio stream at 160kbps (with padding).
+// The size of this frame should be...
+// ((1152 samples/frame * 160000 bits/sec) /
+//  (8000 samples/sec * 8 bits/byte)) + 1 padding byte/frame = 2881 bytes/frame.
+// Set our max frame size to the nearest power of 2 above this size (aka, 4kB)
+const size_t MP3Source::kMaxFrameSize = (1 << 12); /* 4096 bytes */
+MP3Source::MP3Source(
+        MetaDataBase &meta, DataSourceBase *source,
+        off64_t first_frame_pos, uint32_t fixed_header,
+        MP3Seeker *seeker)
+    : mMeta(meta),
+      mDataSource(source),
+      mFirstFramePos(first_frame_pos),
+      mFixedHeader(fixed_header),
+      mCurrentPos(0),
+      mCurrentTimeUs(0),
+      mStarted(false),
+      mSeeker(seeker),
+      mGroup(NULL),
+      mBasisTimeUs(0),
+      mSamplesRead(0) {
+}
+
+MP3Source::~MP3Source() {
+    if (mStarted) {
+        stop();
+    }
+}
+
+status_t MP3Source::start(MetaDataBase *) {
+    CHECK(!mStarted);
+
+    mGroup = new MediaBufferGroup;
+
+    mGroup->add_buffer(MediaBufferBase::Create(kMaxFrameSize));
+
+    mCurrentPos = mFirstFramePos;
+    mCurrentTimeUs = 0;
+
+    mBasisTimeUs = mCurrentTimeUs;
+    mSamplesRead = 0;
+
+    mStarted = true;
+
+    return OK;
+}
+
+status_t MP3Source::stop() {
+    CHECK(mStarted);
+
+    delete mGroup;
+    mGroup = NULL;
+
+    mStarted = false;
+
+    return OK;
+}
+
+status_t MP3Source::getFormat(MetaDataBase &meta) {
+    meta = mMeta;
+    return OK;
+}
+
+status_t MP3Source::read(
+        MediaBufferBase **out, const ReadOptions *options) {
+    *out = NULL;
+
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode mode;
+    bool seekCBR = false;
+
+    if (options != NULL && options->getSeekTo(&seekTimeUs, &mode)) {
+        int64_t actualSeekTimeUs = seekTimeUs;
+        if (mSeeker == NULL
+                || !mSeeker->getOffsetForTime(&actualSeekTimeUs, &mCurrentPos)) {
+            int32_t bitrate;
+            if (!mMeta.findInt32(kKeyBitRate, &bitrate)) {
+                // bitrate is in bits/sec.
+                ALOGI("no bitrate");
+
+                return ERROR_UNSUPPORTED;
+            }
+
+            mCurrentTimeUs = seekTimeUs;
+            mCurrentPos = mFirstFramePos + seekTimeUs * bitrate / 8000000;
+            seekCBR = true;
+        } else {
+            mCurrentTimeUs = actualSeekTimeUs;
+        }
+
+        mBasisTimeUs = mCurrentTimeUs;
+        mSamplesRead = 0;
+    }
+
+    MediaBufferBase *buffer;
+    status_t err = mGroup->acquire_buffer(&buffer);
+    if (err != OK) {
+        return err;
+    }
+
+    size_t frame_size;
+    int bitrate;
+    int num_samples;
+    int sample_rate;
+    for (;;) {
+        ssize_t n = mDataSource->readAt(mCurrentPos, buffer->data(), 4);
+        if (n < 4) {
+            buffer->release();
+            buffer = NULL;
+
+            return (n < 0 ? n : ERROR_END_OF_STREAM);
+        }
+
+        uint32_t header = U32_AT((const uint8_t *)buffer->data());
+
+        if ((header & kMask) == (mFixedHeader & kMask)
+            && GetMPEGAudioFrameSize(
+                header, &frame_size, &sample_rate, NULL,
+                &bitrate, &num_samples)) {
+
+            // re-calculate mCurrentTimeUs because we might have called Resync()
+            if (seekCBR) {
+                mCurrentTimeUs = (mCurrentPos - mFirstFramePos) * 8000 / bitrate;
+                mBasisTimeUs = mCurrentTimeUs;
+            }
+
+            break;
+        }
+
+        // Lost sync.
+        ALOGV("lost sync! header = 0x%08x, old header = 0x%08x\n", header, mFixedHeader);
+
+        off64_t pos = mCurrentPos;
+        if (!Resync(mDataSource, mFixedHeader, &pos, NULL, NULL)) {
+            ALOGE("Unable to resync. Signalling end of stream.");
+
+            buffer->release();
+            buffer = NULL;
+
+            return ERROR_END_OF_STREAM;
+        }
+
+        mCurrentPos = pos;
+
+        // Try again with the new position.
+    }
+
+    CHECK(frame_size <= buffer->size());
+
+    ssize_t n = mDataSource->readAt(mCurrentPos, buffer->data(), frame_size);
+    if (n < (ssize_t)frame_size) {
+        buffer->release();
+        buffer = NULL;
+
+        return (n < 0 ? n : ERROR_END_OF_STREAM);
+    }
+
+    buffer->set_range(0, frame_size);
+
+    buffer->meta_data().setInt64(kKeyTime, mCurrentTimeUs);
+    buffer->meta_data().setInt32(kKeyIsSyncFrame, 1);
+
+    mCurrentPos += frame_size;
+
+    mSamplesRead += num_samples;
+    mCurrentTimeUs = mBasisTimeUs + ((mSamplesRead * 1000000) / sample_rate);
+
+    *out = buffer;
+
+    return OK;
+}
+
+status_t MP3Extractor::getMetaData(MetaDataBase &meta) {
+    meta.clear();
+    if (mInitCheck != OK) {
+        return UNKNOWN_ERROR;
+    }
+    meta.setCString(kKeyMIMEType, "audio/mpeg");
+
+    ID3 id3(mDataSource);
+
+    if (!id3.isValid()) {
+        return OK;
+    }
+
+    struct Map {
+        int key;
+        const char *tag1;
+        const char *tag2;
+    };
+    static const Map kMap[] = {
+        { kKeyAlbum, "TALB", "TAL" },
+        { kKeyArtist, "TPE1", "TP1" },
+        { kKeyAlbumArtist, "TPE2", "TP2" },
+        { kKeyComposer, "TCOM", "TCM" },
+        { kKeyGenre, "TCON", "TCO" },
+        { kKeyTitle, "TIT2", "TT2" },
+        { kKeyYear, "TYE", "TYER" },
+        { kKeyAuthor, "TXT", "TEXT" },
+        { kKeyCDTrackNumber, "TRK", "TRCK" },
+        { kKeyDiscNumber, "TPA", "TPOS" },
+        { kKeyCompilation, "TCP", "TCMP" },
+    };
+    static const size_t kNumMapEntries = sizeof(kMap) / sizeof(kMap[0]);
+
+    for (size_t i = 0; i < kNumMapEntries; ++i) {
+        ID3::Iterator *it = new ID3::Iterator(id3, kMap[i].tag1);
+        if (it->done()) {
+            delete it;
+            it = new ID3::Iterator(id3, kMap[i].tag2);
+        }
+
+        if (it->done()) {
+            delete it;
+            continue;
+        }
+
+        String8 s;
+        it->getString(&s);
+        delete it;
+
+        meta.setCString(kMap[i].key, s);
+    }
+
+    size_t dataSize;
+    String8 mime;
+    const void *data = id3.getAlbumArt(&dataSize, &mime);
+
+    if (data) {
+        meta.setData(kKeyAlbumArt, MetaData::TYPE_NONE, data, dataSize);
+        meta.setCString(kKeyAlbumArtMIME, mime.string());
+    }
+
+    return OK;
+}
+
+static MediaExtractor* CreateExtractor(
+        DataSourceBase *source,
+        void *meta) {
+    Mp3Meta *metaData = static_cast<Mp3Meta *>(meta);
+    return new MP3Extractor(source, metaData);
+}
+
+static MediaExtractor::CreatorFunc Sniff(
+        DataSourceBase *source, float *confidence, void **meta,
+        MediaExtractor::FreeMetaFunc *freeMeta) {
+    off64_t pos = 0;
+    off64_t post_id3_pos;
+    uint32_t header;
+    uint8_t mpeg_header[5];
+    if (source->readAt(0, mpeg_header, sizeof(mpeg_header)) < (ssize_t)sizeof(mpeg_header)) {
+        return NULL;
+    }
+
+    if (!memcmp("\x00\x00\x01\xba", mpeg_header, 4) && (mpeg_header[4] >> 4) == 2) {
+        ALOGV("MPEG1PS container is not supported!");
+        return NULL;
+    }
+    if (!Resync(source, 0, &pos, &post_id3_pos, &header)) {
+        return NULL;
+    }
+
+    Mp3Meta *mp3Meta = new Mp3Meta;
+    mp3Meta->pos = pos;
+    mp3Meta->header = header;
+    mp3Meta->post_id3_pos = post_id3_pos;
+    *meta = mp3Meta;
+    *freeMeta = ::free;
+
+    *confidence = 0.2f;
+
+    return CreateExtractor;
+}
+
+extern "C" {
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
+MediaExtractor::ExtractorDef GETEXTRACTORDEF() {
+    return {
+        MediaExtractor::EXTRACTORDEF_VERSION,
+        UUID("812a3f6c-c8cf-46de-b529-3774b14103d4"),
+        1, // version
+        "MP3 Extractor",
+        Sniff
+    };
+}
+
+} // extern "C"
+
+}  // namespace android

diff --git a/media/extractors/mp3/MP3Extractor.h b/media/extractors/mp3/MP3Extractor.h
new file mode 100644
index 0000000..485b0ca
--- /dev/null
+++ b/media/extractors/mp3/MP3Extractor.h

@@ -0,0 +1,60 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MP3_EXTRACTOR_H_
+
+#define MP3_EXTRACTOR_H_
+
+#include <utils/Errors.h>
+#include <media/MediaExtractor.h>
+#include <media/stagefright/MetaDataBase.h>
+
+namespace android {
+
+struct AMessage;
+class DataSourceBase;
+struct MP3Seeker;
+class String8;
+struct Mp3Meta;
+
+class MP3Extractor : public MediaExtractor {
+public:
+    MP3Extractor(DataSourceBase *source, Mp3Meta *meta);
+    ~MP3Extractor();
+
+    virtual size_t countTracks();
+    virtual MediaTrack *getTrack(size_t index);
+    virtual status_t getTrackMetaData(MetaDataBase& meta, size_t index, uint32_t flags);
+
+    virtual status_t getMetaData(MetaDataBase& meta);
+    virtual const char * name() { return "MP3Extractor"; }
+
+private:
+    status_t mInitCheck;
+
+    DataSourceBase *mDataSource;
+    off64_t mFirstFramePos;
+    MetaDataBase mMeta;
+    uint32_t mFixedHeader;
+    MP3Seeker *mSeeker;
+
+    MP3Extractor(const MP3Extractor &);
+    MP3Extractor &operator=(const MP3Extractor &);
+};
+
+}  // namespace android
+
+#endif  // MP3_EXTRACTOR_H_

diff --git a/media/extractors/mp3/MP3Seeker.h b/media/extractors/mp3/MP3Seeker.h
new file mode 100644
index 0000000..0e3af25
--- /dev/null
+++ b/media/extractors/mp3/MP3Seeker.h

@@ -0,0 +1,45 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MP3_SEEKER_H_
+
+#define MP3_SEEKER_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+struct MP3Seeker {
+    MP3Seeker() {}
+
+    virtual bool getDuration(int64_t *durationUs) = 0;
+
+    // Given a request seek time in "*timeUs", find the byte offset closest
+    // to that position and return it in "*pos". Update "*timeUs" to reflect
+    // the actual time that seekpoint represents.
+    virtual bool getOffsetForTime(int64_t *timeUs, off64_t *pos) = 0;
+
+    virtual ~MP3Seeker() {}
+
+private:
+    DISALLOW_EVIL_CONSTRUCTORS(MP3Seeker);
+};
+
+}  // namespace android
+
+#endif  // MP3_SEEKER_H_
+

diff --git a/media/extractors/mp3/VBRISeeker.cpp b/media/extractors/mp3/VBRISeeker.cpp
new file mode 100644
index 0000000..523f14c
--- /dev/null
+++ b/media/extractors/mp3/VBRISeeker.cpp

@@ -0,0 +1,188 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "VBRISeeker"
+
+#include <inttypes.h>
+
+#include <utils/Log.h>
+
+#include "VBRISeeker.h"
+
+#include <media/stagefright/foundation/avc_utils.h>
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ByteUtils.h>
+#include <media/DataSourceBase.h>
+
+namespace android {
+
+static uint32_t U24_AT(const uint8_t *ptr) {
+    return ptr[0] << 16 | ptr[1] << 8 | ptr[2];
+}
+
+// static
+VBRISeeker *VBRISeeker::CreateFromSource(
+        DataSourceBase *source, off64_t post_id3_pos) {
+    off64_t pos = post_id3_pos;
+
+    uint8_t header[4];
+    ssize_t n = source->readAt(pos, header, sizeof(header));
+    if (n < (ssize_t)sizeof(header)) {
+        return NULL;
+    }
+
+    uint32_t tmp = U32_AT(&header[0]);
+    size_t frameSize;
+    int sampleRate;
+    if (!GetMPEGAudioFrameSize(tmp, &frameSize, &sampleRate)) {
+        return NULL;
+    }
+
+    // VBRI header follows 32 bytes after the header _ends_.
+    pos += sizeof(header) + 32;
+
+    uint8_t vbriHeader[26];
+    n = source->readAt(pos, vbriHeader, sizeof(vbriHeader));
+    if (n < (ssize_t)sizeof(vbriHeader)) {
+        return NULL;
+    }
+
+    if (memcmp(vbriHeader, "VBRI", 4)) {
+        return NULL;
+    }
+
+    size_t numFrames = U32_AT(&vbriHeader[14]);
+
+    int64_t durationUs =
+        numFrames * 1000000ll * (sampleRate >= 32000 ? 1152 : 576) / sampleRate;
+
+    ALOGV("duration = %.2f secs", durationUs / 1E6);
+
+    size_t numEntries = U16_AT(&vbriHeader[18]);
+    size_t entrySize = U16_AT(&vbriHeader[22]);
+    size_t scale = U16_AT(&vbriHeader[20]);
+
+    ALOGV("%zu entries, scale=%zu, size_per_entry=%zu",
+         numEntries,
+         scale,
+         entrySize);
+
+    if (entrySize > 4) {
+        ALOGE("invalid VBRI entry size: %zu", entrySize);
+        return NULL;
+    }
+
+    VBRISeeker *seeker = new (std::nothrow) VBRISeeker;
+    if (seeker == NULL) {
+        ALOGW("Couldn't allocate VBRISeeker");
+        return NULL;
+    }
+
+    size_t totalEntrySize = numEntries * entrySize;
+    uint8_t *buffer = new (std::nothrow) uint8_t[totalEntrySize];
+    if (!buffer) {
+        ALOGW("Couldn't allocate %zu bytes", totalEntrySize);
+        delete seeker;
+        return NULL;
+    }
+
+    n = source->readAt(pos + sizeof(vbriHeader), buffer, totalEntrySize);
+    if (n < (ssize_t)totalEntrySize) {
+        delete[] buffer;
+        buffer = NULL;
+        delete seeker;
+        return NULL;
+    }
+
+    seeker->mBasePos = post_id3_pos + frameSize;
+    // only update mDurationUs if the calculated duration is valid (non zero)
+    // otherwise, leave duration at -1 so that getDuration() and getOffsetForTime()
+    // return false when called, to indicate that this vbri tag does not have the
+    // requested information
+    if (durationUs) {
+        seeker->mDurationUs = durationUs;
+    }
+
+    off64_t offset = post_id3_pos;
+    for (size_t i = 0; i < numEntries; ++i) {
+        uint32_t numBytes;
+        switch (entrySize) {
+            case 1: numBytes = buffer[i]; break;
+            case 2: numBytes = U16_AT(buffer + 2 * i); break;
+            case 3: numBytes = U24_AT(buffer + 3 * i); break;
+            default:
+            {
+                CHECK_EQ(entrySize, 4u);
+                numBytes = U32_AT(buffer + 4 * i); break;
+            }
+        }
+
+        numBytes *= scale;
+
+        seeker->mSegments.push(numBytes);
+
+        ALOGV("entry #%zu: %u offset %#016llx", i, numBytes, (long long)offset);
+        offset += numBytes;
+    }
+
+    delete[] buffer;
+    buffer = NULL;
+
+    ALOGI("Found VBRI header.");
+
+    return seeker;
+}
+
+VBRISeeker::VBRISeeker()
+    : mDurationUs(-1) {
+}
+
+bool VBRISeeker::getDuration(int64_t *durationUs) {
+    if (mDurationUs < 0) {
+        return false;
+    }
+
+    *durationUs = mDurationUs;
+
+    return true;
+}
+
+bool VBRISeeker::getOffsetForTime(int64_t *timeUs, off64_t *pos) {
+    if (mDurationUs < 0 || mSegments.size() == 0) {
+        return false;
+    }
+
+    int64_t segmentDurationUs = mDurationUs / mSegments.size();
+
+    int64_t nowUs = 0;
+    *pos = mBasePos;
+    size_t segmentIndex = 0;
+    while (segmentIndex < mSegments.size() && nowUs < *timeUs) {
+        nowUs += segmentDurationUs;
+        *pos += mSegments.itemAt(segmentIndex++);
+    }
+
+    ALOGV("getOffsetForTime %lld us => 0x%016llx", (long long)*timeUs, (long long)*pos);
+
+    *timeUs = nowUs;
+
+    return true;
+}
+
+}  // namespace android
+

diff --git a/media/extractors/mp3/VBRISeeker.h b/media/extractors/mp3/VBRISeeker.h
new file mode 100644
index 0000000..9213f6e
--- /dev/null
+++ b/media/extractors/mp3/VBRISeeker.h

@@ -0,0 +1,50 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef VBRI_SEEKER_H_
+
+#define VBRI_SEEKER_H_
+
+#include "MP3Seeker.h"
+
+#include <utils/Vector.h>
+
+namespace android {
+
+class DataSourceBase;
+
+struct VBRISeeker : public MP3Seeker {
+    static VBRISeeker *CreateFromSource(
+            DataSourceBase *source, off64_t post_id3_pos);
+
+    virtual bool getDuration(int64_t *durationUs);
+    virtual bool getOffsetForTime(int64_t *timeUs, off64_t *pos);
+
+private:
+    off64_t mBasePos;
+    int64_t mDurationUs;
+    Vector<uint32_t> mSegments;
+
+    VBRISeeker();
+
+    DISALLOW_EVIL_CONSTRUCTORS(VBRISeeker);
+};
+
+}  // namespace android
+
+#endif  // VBRI_SEEKER_H_
+
+

diff --git a/media/extractors/mp3/XINGSeeker.cpp b/media/extractors/mp3/XINGSeeker.cpp
new file mode 100644
index 0000000..95ca556
--- /dev/null
+++ b/media/extractors/mp3/XINGSeeker.cpp

@@ -0,0 +1,202 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "XINGSEEKER"
+#include <utils/Log.h>
+
+#include "XINGSeeker.h"
+#include <media/stagefright/foundation/avc_utils.h>
+
+#include <media/stagefright/foundation/ByteUtils.h>
+#include <media/DataSourceBase.h>
+
+namespace android {
+
+XINGSeeker::XINGSeeker()
+    : mDurationUs(-1),
+      mSizeBytes(0),
+      mEncoderDelay(0),
+      mEncoderPadding(0),
+      mTOCValid(false) {
+}
+
+bool XINGSeeker::getDuration(int64_t *durationUs) {
+    if (mDurationUs < 0) {
+        return false;
+    }
+
+    *durationUs = mDurationUs;
+
+    return true;
+}
+
+bool XINGSeeker::getOffsetForTime(int64_t *timeUs, off64_t *pos) {
+    if (mSizeBytes == 0 || !mTOCValid || mDurationUs < 0) {
+        return false;
+    }
+
+    float percent = (float)(*timeUs) * 100 / mDurationUs;
+    float fx;
+    if( percent <= 0.0f ) {
+        fx = 0.0f;
+    } else if( percent >= 100.0f ) {
+        fx = 256.0f;
+    } else {
+        int a = (int)percent;
+        float fa, fb;
+        if ( a == 0 ) {
+            fa = 0.0f;
+        } else {
+            fa = (float)mTOC[a-1];
+        }
+        if ( a < 99 ) {
+            fb = (float)mTOC[a];
+        } else {
+            fb = 256.0f;
+        }
+        fx = fa + (fb-fa)*(percent-a);
+    }
+
+    *pos = (int)((1.0f/256.0f)*fx*mSizeBytes) + mFirstFramePos;
+
+    return true;
+}
+
+// static
+XINGSeeker *XINGSeeker::CreateFromSource(
+        DataSourceBase *source, off64_t first_frame_pos) {
+
+    uint8_t buffer[4];
+    int offset = first_frame_pos;
+    if (source->readAt(offset, &buffer, 4) < 4) { // get header
+        return NULL;
+    }
+    offset += 4;
+
+    int header = U32_AT(buffer);;
+    size_t xingframesize = 0;
+    int sampling_rate = 0;
+    int num_channels;
+    int samples_per_frame = 0;
+    if (!GetMPEGAudioFrameSize(header, &xingframesize, &sampling_rate, &num_channels,
+                               NULL, &samples_per_frame)) {
+        return NULL;
+    }
+    uint8_t version = (buffer[1] >> 3) & 3;
+
+    // determine offset of XING header
+    if(version & 1) { // mpeg1
+        if (num_channels != 1) offset += 32;
+        else offset += 17;
+    } else { // mpeg 2 or 2.5
+        if (num_channels != 1) offset += 17;
+        else offset += 9;
+    }
+
+    int xingbase = offset;
+
+    if (source->readAt(offset, &buffer, 4) < 4) { // XING header ID
+        return NULL;
+    }
+    offset += 4;
+    // Check XING ID
+    if ((buffer[0] != 'X') || (buffer[1] != 'i')
+                || (buffer[2] != 'n') || (buffer[3] != 'g')) {
+        if ((buffer[0] != 'I') || (buffer[1] != 'n')
+                    || (buffer[2] != 'f') || (buffer[3] != 'o')) {
+            return NULL;
+        }
+    }
+
+    if (source->readAt(offset, &buffer, 4) < 4) { // flags
+        return NULL;
+    }
+    offset += 4;
+    uint32_t flags = U32_AT(buffer);
+
+    XINGSeeker *seeker = new XINGSeeker;
+    seeker->mFirstFramePos = first_frame_pos + xingframesize;
+
+    if (flags & 0x0001) {  // Frames field is present
+        if (source->readAt(offset, buffer, 4) < 4) {
+            delete seeker;
+            return NULL;
+        }
+        int32_t frames = U32_AT(buffer);
+        // only update mDurationUs if the calculated duration is valid (non zero)
+        // otherwise, leave duration at -1 so that getDuration() and getOffsetForTime()
+        // return false when called, to indicate that this xing tag does not have the
+        // requested information
+        if (frames) {
+            seeker->mDurationUs = (int64_t)frames * samples_per_frame * 1000000LL / sampling_rate;
+        }
+        offset += 4;
+    }
+    if (flags & 0x0002) {  // Bytes field is present
+        if (source->readAt(offset, buffer, 4) < 4) {
+            delete seeker;
+            return NULL;
+        }
+        seeker->mSizeBytes = U32_AT(buffer);
+        offset += 4;
+    }
+    if (flags & 0x0004) {  // TOC field is present
+        if (source->readAt(offset + 1, seeker->mTOC, 99) < 99) {
+            delete seeker;
+            return NULL;
+        }
+        seeker->mTOCValid = true;
+        offset += 100;
+    }
+
+#if 0
+    if (flags & 0x0008) {  // Quality indicator field is present
+        if (source->readAt(offset, buffer, 4) < 4) {
+            delete seeker;
+            return NULL;
+        }
+        // do something with the quality indicator
+        offset += 4;
+    }
+
+    if (source->readAt(xingbase + 0xaf - 0x24, &buffer, 1) < 1) { // encoding flags
+        delete seeker;
+        return false;
+    }
+
+    ALOGV("nogap preceding: %s, nogap continued in next: %s",
+              (buffer[0] & 0x80) ? "true" : "false",
+              (buffer[0] & 0x40) ? "true" : "false");
+#endif
+
+    if (source->readAt(xingbase + 0xb1 - 0x24, &buffer, 3) == 3) {
+        seeker->mEncoderDelay = (buffer[0] << 4) + (buffer[1] >> 4);
+        seeker->mEncoderPadding = ((buffer[1] & 0xf) << 8) + buffer[2];
+    }
+
+    return seeker;
+}
+
+int32_t XINGSeeker::getEncoderDelay() {
+    return mEncoderDelay;
+}
+
+int32_t XINGSeeker::getEncoderPadding() {
+    return mEncoderPadding;
+}
+
+}  // namespace android
+

diff --git a/media/extractors/mp3/XINGSeeker.h b/media/extractors/mp3/XINGSeeker.h
new file mode 100644
index 0000000..5867eae
--- /dev/null
+++ b/media/extractors/mp3/XINGSeeker.h

@@ -0,0 +1,56 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef XING_SEEKER_H_
+
+#define XING_SEEKER_H_
+
+#include "MP3Seeker.h"
+
+namespace android {
+
+class DataSourceBase;
+
+struct XINGSeeker : public MP3Seeker {
+    static XINGSeeker *CreateFromSource(
+            DataSourceBase *source, off64_t first_frame_pos);
+
+    virtual bool getDuration(int64_t *durationUs);
+    virtual bool getOffsetForTime(int64_t *timeUs, off64_t *pos);
+
+    virtual int32_t getEncoderDelay();
+    virtual int32_t getEncoderPadding();
+
+private:
+    int64_t mFirstFramePos;
+    int64_t mDurationUs;
+    int32_t mSizeBytes;
+    int32_t mEncoderDelay;
+    int32_t mEncoderPadding;
+
+    // TOC entries in XING header. Skip the first one since it's always 0.
+    unsigned char mTOC[99];
+    bool mTOCValid;
+
+    XINGSeeker();
+
+    DISALLOW_EVIL_CONSTRUCTORS(XINGSeeker);
+};
+
+}  // namespace android
+
+#endif  // XING_SEEKER_H_
+

diff --git a/media/extractors/mp3/exports.lds b/media/extractors/mp3/exports.lds
new file mode 100644
index 0000000..b1309ee
--- /dev/null
+++ b/media/extractors/mp3/exports.lds

@@ -0,0 +1 @@
+{ global: GETEXTRACTORDEF; local: *; };

diff --git a/media/extractors/mp4/Android.bp b/media/extractors/mp4/Android.bp
new file mode 100644
index 0000000..fa739e8
--- /dev/null
+++ b/media/extractors/mp4/Android.bp

@@ -0,0 +1,60 @@
+cc_defaults {
+    name: "libmp4extractor_defaults",
+
+    srcs: [
+        "ItemTable.cpp",
+        "MPEG4Extractor.cpp",
+        "SampleIterator.cpp",
+        "SampleTable.cpp",
+    ],
+
+    include_dirs: [
+        "frameworks/av/media/libstagefright/",
+    ],
+
+    shared_libs: [
+        "liblog",
+        "libmediaextractor",
+    ],
+
+    static_libs: [
+        "libstagefright_esds",
+        "libstagefright_foundation",
+        "libstagefright_id3",
+        "libutils",
+    ],
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+        "-fvisibility=hidden",
+    ],
+    version_script: "exports.lds",
+    relative_install_path: "extractors",
+    compile_multilib: "first",
+}
+
+cc_library_shared {
+
+
+    name: "libmp4extractor",
+    defaults: ["libmp4extractor_defaults"],
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+        diag: {
+            cfi: true,
+        },
+    },
+
+}
+
+cc_library_static {
+    name: "libmp4extractor_fuzzing",
+
+    defaults: ["libmp4extractor_defaults"],
+}

diff --git a/media/extractors/mp4/ItemTable.cpp b/media/extractors/mp4/ItemTable.cpp
new file mode 100644
index 0000000..ca9deab
--- /dev/null
+++ b/media/extractors/mp4/ItemTable.cpp

@@ -0,0 +1,1681 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "ItemTable"
+
+#include <ItemTable.h>
+#include <media/DataSourceBase.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ByteUtils.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/foundation/MediaDefs.h>
+#include <utils/Log.h>
+
+namespace android {
+
+namespace heif {
+
+/////////////////////////////////////////////////////////////////////
+//
+//  struct to keep track of one image item
+//
+
+struct ImageItem {
+    friend struct ItemReference;
+    friend struct ItemProperty;
+
+    ImageItem() : ImageItem(0, 0, false) {}
+    ImageItem(uint32_t _type, uint32_t _id, bool _hidden) :
+            type(_type), itemId(_id), hidden(_hidden),
+            rows(0), columns(0), width(0), height(0), rotation(0),
+            offset(0), size(0), nextTileIndex(0) {}
+
+    bool isGrid() const {
+        return type == FOURCC('g', 'r', 'i', 'd');
+    }
+
+    status_t getNextTileItemId(uint32_t *nextTileItemId, bool reset) {
+        if (reset) {
+            nextTileIndex = 0;
+        }
+        if (nextTileIndex >= dimgRefs.size()) {
+            return ERROR_END_OF_STREAM;
+        }
+        *nextTileItemId = dimgRefs[nextTileIndex++];
+        return OK;
+    }
+
+    uint32_t type;
+    uint32_t itemId;
+    bool hidden;
+    int32_t rows;
+    int32_t columns;
+    int32_t width;
+    int32_t height;
+    int32_t rotation;
+    off64_t offset;
+    size_t size;
+    sp<ABuffer> hvcc;
+    sp<ABuffer> icc;
+
+    Vector<uint32_t> thumbnails;
+    Vector<uint32_t> dimgRefs;
+    Vector<uint32_t> cdscRefs;
+    size_t nextTileIndex;
+};
+
+struct ExifItem {
+    off64_t offset;
+    size_t size;
+};
+
+/////////////////////////////////////////////////////////////////////
+//
+//  ISO boxes
+//
+
+struct Box {
+protected:
+    Box(DataSourceBase *source, uint32_t type) :
+        mDataSource(source), mType(type) {}
+
+    virtual ~Box() {}
+
+    virtual status_t onChunkData(
+            uint32_t /*type*/, off64_t /*offset*/, size_t /*size*/) {
+        return OK;
+    }
+
+    inline uint32_t type() const { return mType; }
+
+    inline DataSourceBase *source() const { return mDataSource; }
+
+    status_t parseChunk(off64_t *offset);
+
+    status_t parseChunks(off64_t offset, size_t size);
+
+private:
+    DataSourceBase *mDataSource;
+    uint32_t mType;
+};
+
+status_t Box::parseChunk(off64_t *offset) {
+    if (*offset < 0) {
+        ALOGE("b/23540914");
+        return ERROR_MALFORMED;
+    }
+    uint32_t hdr[2];
+    if (mDataSource->readAt(*offset, hdr, 8) < 8) {
+        return ERROR_IO;
+    }
+    uint64_t chunk_size = ntohl(hdr[0]);
+    int32_t chunk_type = ntohl(hdr[1]);
+    off64_t data_offset = *offset + 8;
+
+    if (chunk_size == 1) {
+        if (mDataSource->readAt(*offset + 8, &chunk_size, 8) < 8) {
+            return ERROR_IO;
+        }
+        chunk_size = ntoh64(chunk_size);
+        data_offset += 8;
+
+        if (chunk_size < 16) {
+            // The smallest valid chunk is 16 bytes long in this case.
+            return ERROR_MALFORMED;
+        }
+    } else if (chunk_size == 0) {
+        // This shouldn't happen since we should never be top level
+        ALOGE("invalid chunk size 0 for non-top level box");
+        return ERROR_MALFORMED;
+    } else if (chunk_size < 8) {
+        // The smallest valid chunk is 8 bytes long.
+        ALOGE("invalid chunk size: %lld", (long long)chunk_size);
+        return ERROR_MALFORMED;
+    }
+
+    char chunk[5];
+    MakeFourCCString(chunk_type, chunk);
+    ALOGV("chunk: %s @ %lld", chunk, (long long)*offset);
+
+    off64_t chunk_data_size = chunk_size - (data_offset - *offset);
+    if (chunk_data_size < 0) {
+        ALOGE("b/23540914");
+        return ERROR_MALFORMED;
+    }
+
+    status_t err = onChunkData(chunk_type, data_offset, chunk_data_size);
+
+    if (err != OK) {
+        return err;
+    }
+    *offset += chunk_size;
+    return OK;
+}
+
+status_t Box::parseChunks(off64_t offset, size_t size) {
+    off64_t stopOffset = offset + size;
+    while (offset < stopOffset) {
+        status_t err = parseChunk(&offset);
+        if (err != OK) {
+            return err;
+        }
+    }
+    if (offset != stopOffset) {
+        return ERROR_MALFORMED;
+    }
+    return OK;
+}
+
+///////////////////////////////////////////////////////////////////////
+
+struct FullBox : public Box {
+protected:
+    FullBox(DataSourceBase *source, uint32_t type) :
+        Box(source, type), mVersion(0), mFlags(0) {}
+
+    inline uint8_t version() const { return mVersion; }
+
+    inline uint32_t flags() const { return mFlags; }
+
+    status_t parseFullBoxHeader(off64_t *offset, size_t *size);
+
+private:
+    uint8_t mVersion;
+    uint32_t mFlags;
+};
+
+status_t FullBox::parseFullBoxHeader(off64_t *offset, size_t *size) {
+    if (*size < 4) {
+        return ERROR_MALFORMED;
+    }
+    if (!source()->readAt(*offset, &mVersion, 1)) {
+        return ERROR_IO;
+    }
+    if (!source()->getUInt24(*offset + 1, &mFlags)) {
+        return ERROR_IO;
+    }
+    *offset += 4;
+    *size -= 4;
+    return OK;
+}
+
+/////////////////////////////////////////////////////////////////////
+//
+//  PrimaryImage box
+//
+
+struct PitmBox : public FullBox {
+    PitmBox(DataSourceBase *source) :
+        FullBox(source, FOURCC('p', 'i', 't', 'm')) {}
+
+    status_t parse(off64_t offset, size_t size, uint32_t *primaryItemId);
+};
+
+status_t PitmBox::parse(off64_t offset, size_t size, uint32_t *primaryItemId) {
+    status_t err = parseFullBoxHeader(&offset, &size);
+    if (err != OK) {
+        return err;
+    }
+
+    size_t itemIdSize = (version() == 0) ? 2 : 4;
+    if (size < itemIdSize) {
+        return ERROR_MALFORMED;
+    }
+    uint32_t itemId;
+    if (!source()->getUInt32Var(offset, &itemId, itemIdSize)) {
+        return ERROR_IO;
+    }
+
+    ALOGV("primary id %d", itemId);
+    *primaryItemId = itemId;
+
+    return OK;
+}
+
+/////////////////////////////////////////////////////////////////////
+//
+//  ItemLocation related boxes
+//
+
+struct ExtentEntry {
+    uint64_t extentIndex;
+    uint64_t extentOffset;
+    uint64_t extentLength;
+};
+
+struct ItemLoc {
+    ItemLoc() : ItemLoc(0, 0, 0, 0) {}
+    ItemLoc(uint32_t item_id, uint16_t construction_method,
+            uint16_t data_reference_index, uint64_t base_offset) :
+        itemId(item_id),
+        constructionMethod(construction_method),
+        dataReferenceIndex(data_reference_index),
+        baseOffset(base_offset) {}
+
+    void addExtent(const ExtentEntry& extent) {
+        extents.push_back(extent);
+    }
+
+    status_t getLoc(off64_t *offset, size_t *size,
+            off64_t idatOffset, size_t idatSize) const {
+        // TODO: fix extent handling, fix constructionMethod = 2
+        CHECK(extents.size() == 1);
+        if (constructionMethod == 0) {
+            *offset = baseOffset + extents[0].extentOffset;
+            *size = extents[0].extentLength;
+            return OK;
+        } else if (constructionMethod == 1) {
+            if (baseOffset + extents[0].extentOffset + extents[0].extentLength
+                    > idatSize) {
+                return ERROR_MALFORMED;
+            }
+            *offset = baseOffset + extents[0].extentOffset + idatOffset;
+            *size = extents[0].extentLength;
+            return OK;
+        }
+        return ERROR_UNSUPPORTED;
+    }
+
+    // parsed info
+    uint32_t itemId;
+    uint16_t constructionMethod;
+    uint16_t dataReferenceIndex;
+    off64_t baseOffset;
+    Vector<ExtentEntry> extents;
+};
+
+struct IlocBox : public FullBox {
+    IlocBox(DataSourceBase *source, KeyedVector<uint32_t, ItemLoc> *itemLocs) :
+        FullBox(source, FOURCC('i', 'l', 'o', 'c')),
+        mItemLocs(itemLocs), mHasConstructMethod1(false) {}
+
+    status_t parse(off64_t offset, size_t size);
+
+    bool hasConstructMethod1() { return mHasConstructMethod1; }
+
+private:
+    static bool isSizeFieldValid(uint32_t offset_size) {
+        return offset_size == 0 || offset_size == 4 || offset_size == 8;
+    }
+    KeyedVector<uint32_t, ItemLoc> *mItemLocs;
+    bool mHasConstructMethod1;
+};
+
+status_t IlocBox::parse(off64_t offset, size_t size) {
+    status_t err = parseFullBoxHeader(&offset, &size);
+    if (err != OK) {
+        return err;
+    }
+    if (version() > 2) {
+        ALOGE("%s: invalid version %d", __FUNCTION__, version());
+        return ERROR_MALFORMED;
+    }
+
+    if (size < 2) {
+        return ERROR_MALFORMED;
+    }
+    uint8_t offset_size;
+    if (!source()->readAt(offset++, &offset_size, 1)) {
+        return ERROR_IO;
+    }
+    uint8_t length_size = (offset_size & 0xF);
+    offset_size >>= 4;
+
+    uint8_t base_offset_size;
+    if (!source()->readAt(offset++, &base_offset_size, 1)) {
+        return ERROR_IO;
+    }
+    uint8_t index_size = 0;
+    if (version() == 1 || version() == 2) {
+        index_size = (base_offset_size & 0xF);
+    }
+    base_offset_size >>= 4;
+    size -= 2;
+
+    if (!isSizeFieldValid(offset_size)
+            || !isSizeFieldValid(length_size)
+            || !isSizeFieldValid(base_offset_size)
+            || !isSizeFieldValid((index_size))) {
+        ALOGE("%s: offset size not valid: %d, %d, %d, %d", __FUNCTION__,
+                offset_size, length_size, base_offset_size, index_size);
+        return ERROR_MALFORMED;
+    }
+
+    uint32_t item_count;
+    size_t itemFieldSize = version() < 2 ? 2 : 4;
+    if (size < itemFieldSize) {
+        return ERROR_MALFORMED;
+    }
+    if (!source()->getUInt32Var(offset, &item_count, itemFieldSize)) {
+        return ERROR_IO;
+    }
+
+    ALOGV("item_count %lld", (long long) item_count);
+    offset += itemFieldSize;
+    size -= itemFieldSize;
+
+    for (size_t i = 0; i < item_count; i++) {
+        uint32_t item_id;
+        if (!source()->getUInt32Var(offset, &item_id, itemFieldSize)) {
+            return ERROR_IO;
+        }
+        ALOGV("item[%zu]: id %lld", i, (long long)item_id);
+        offset += itemFieldSize;
+
+        uint8_t construction_method = 0;
+        if (version() == 1 || version() == 2) {
+            uint8_t buf[2];
+            if (!source()->readAt(offset, buf, 2)) {
+                return ERROR_IO;
+            }
+            construction_method = (buf[1] & 0xF);
+            ALOGV("construction_method %d", construction_method);
+            if (construction_method == 1) {
+                mHasConstructMethod1 = true;
+            }
+
+            offset += 2;
+        }
+
+        uint16_t data_reference_index;
+        if (!source()->getUInt16(offset, &data_reference_index)) {
+            return ERROR_IO;
+        }
+        ALOGV("data_reference_index %d", data_reference_index);
+        if (data_reference_index != 0) {
+            // we don't support reference to other files
+            return ERROR_UNSUPPORTED;
+        }
+        offset += 2;
+
+        uint64_t base_offset = 0;
+        if (base_offset_size != 0) {
+            if (!source()->getUInt64Var(offset, &base_offset, base_offset_size)) {
+                return ERROR_IO;
+            }
+            offset += base_offset_size;
+        }
+        ALOGV("base_offset %lld", (long long) base_offset);
+
+        ssize_t index = mItemLocs->add(item_id, ItemLoc(
+                item_id, construction_method, data_reference_index, base_offset));
+        ItemLoc &item = mItemLocs->editValueAt(index);
+
+        uint16_t extent_count;
+        if (!source()->getUInt16(offset, &extent_count)) {
+            return ERROR_IO;
+        }
+        ALOGV("extent_count %d", extent_count);
+
+        if (extent_count > 1 && (offset_size == 0 || length_size == 0)) {
+            // if the item is dividec into more than one extents, offset and
+            // length must be present.
+            return ERROR_MALFORMED;
+        }
+        offset += 2;
+
+        for (size_t j = 0; j < extent_count; j++) {
+            uint64_t extent_index = 1; // default=1
+            if ((version() == 1 || version() == 2) && (index_size > 0)) {
+                if (!source()->getUInt64Var(offset, &extent_index, index_size)) {
+                    return ERROR_IO;
+                }
+                // TODO: add support for this mode
+                offset += index_size;
+                ALOGV("extent_index %lld", (long long)extent_index);
+            }
+
+            uint64_t extent_offset = 0; // default=0
+            if (offset_size > 0) {
+                if (!source()->getUInt64Var(offset, &extent_offset, offset_size)) {
+                    return ERROR_IO;
+                }
+                offset += offset_size;
+            }
+            ALOGV("extent_offset %lld", (long long)extent_offset);
+
+            uint64_t extent_length = 0; // this indicates full length of file
+            if (length_size > 0) {
+                if (!source()->getUInt64Var(offset, &extent_length, length_size)) {
+                    return ERROR_IO;
+                }
+                offset += length_size;
+            }
+            ALOGV("extent_length %lld", (long long)extent_length);
+
+            item.addExtent({ extent_index, extent_offset, extent_length });
+        }
+    }
+    return OK;
+}
+
+/////////////////////////////////////////////////////////////////////
+//
+//  ItemReference related boxes
+//
+
+struct ItemReference : public Box, public RefBase {
+    ItemReference(DataSourceBase *source, uint32_t type, uint32_t itemIdSize) :
+        Box(source, type), mItemId(0), mRefIdSize(itemIdSize) {}
+
+    status_t parse(off64_t offset, size_t size);
+
+    uint32_t itemId() { return mItemId; }
+
+    void apply(
+            KeyedVector<uint32_t, ImageItem> &itemIdToItemMap,
+            KeyedVector<uint32_t, ExifItem> &itemIdToExifMap) const;
+
+private:
+    uint32_t mItemId;
+    uint32_t mRefIdSize;
+    Vector<uint32_t> mRefs;
+
+    DISALLOW_EVIL_CONSTRUCTORS(ItemReference);
+};
+
+void ItemReference::apply(
+        KeyedVector<uint32_t, ImageItem> &itemIdToItemMap,
+        KeyedVector<uint32_t, ExifItem> &itemIdToExifMap) const {
+    ALOGV("attach reference type 0x%x to item id %d)", type(), mItemId);
+
+    switch(type()) {
+    case FOURCC('d', 'i', 'm', 'g'): {
+        ssize_t itemIndex = itemIdToItemMap.indexOfKey(mItemId);
+
+        // ignore non-image items
+        if (itemIndex < 0) {
+            return;
+        }
+
+        ImageItem &derivedImage = itemIdToItemMap.editValueAt(itemIndex);
+        if (!derivedImage.dimgRefs.empty()) {
+            ALOGW("dimgRefs not clean!");
+        }
+        derivedImage.dimgRefs.appendVector(mRefs);
+
+        for (size_t i = 0; i < mRefs.size(); i++) {
+            itemIndex = itemIdToItemMap.indexOfKey(mRefs[i]);
+
+            // ignore non-image items
+            if (itemIndex < 0) {
+                continue;
+            }
+            ImageItem &sourceImage = itemIdToItemMap.editValueAt(itemIndex);
+
+            // mark the source image of the derivation as hidden
+            sourceImage.hidden = true;
+        }
+        break;
+    }
+    case FOURCC('t', 'h', 'm', 'b'): {
+        ssize_t itemIndex = itemIdToItemMap.indexOfKey(mItemId);
+
+        // ignore non-image items
+        if (itemIndex < 0) {
+            return;
+        }
+
+        // mark thumbnail image as hidden, these can be retrieved if the client
+        // request thumbnail explicitly, but won't be exposed as displayables.
+        ImageItem &thumbImage = itemIdToItemMap.editValueAt(itemIndex);
+        thumbImage.hidden = true;
+
+        for (size_t i = 0; i < mRefs.size(); i++) {
+            itemIndex = itemIdToItemMap.indexOfKey(mRefs[i]);
+
+            // ignore non-image items
+            if (itemIndex < 0) {
+                continue;
+            }
+            ALOGV("Image item id %d uses thumbnail item id %d", mRefs[i], mItemId);
+            ImageItem &masterImage = itemIdToItemMap.editValueAt(itemIndex);
+            if (!masterImage.thumbnails.empty()) {
+                ALOGW("already has thumbnails!");
+            }
+            masterImage.thumbnails.push_back(mItemId);
+        }
+        break;
+    }
+    case FOURCC('c', 'd', 's', 'c'): {
+        ssize_t itemIndex = itemIdToExifMap.indexOfKey(mItemId);
+
+        // ignore non-exif block items
+        if (itemIndex < 0) {
+            return;
+        }
+
+        for (size_t i = 0; i < mRefs.size(); i++) {
+            itemIndex = itemIdToItemMap.indexOfKey(mRefs[i]);
+
+            // ignore non-image items
+            if (itemIndex < 0) {
+                continue;
+            }
+            ALOGV("Image item id %d uses metadata item id %d", mRefs[i], mItemId);
+            ImageItem &image = itemIdToItemMap.editValueAt(itemIndex);
+            image.cdscRefs.push_back(mItemId);
+        }
+        break;
+    }
+    case FOURCC('a', 'u', 'x', 'l'): {
+        ssize_t itemIndex = itemIdToItemMap.indexOfKey(mItemId);
+
+        // ignore non-image items
+        if (itemIndex < 0) {
+            return;
+        }
+
+        // mark auxiliary image as hidden
+        ImageItem &auxImage = itemIdToItemMap.editValueAt(itemIndex);
+        auxImage.hidden = true;
+        break;
+    }
+    default:
+        ALOGW("ignoring unsupported ref type 0x%x", type());
+    }
+}
+
+status_t ItemReference::parse(off64_t offset, size_t size) {
+    if (size < mRefIdSize + 2) {
+        return ERROR_MALFORMED;
+    }
+    if (!source()->getUInt32Var(offset, &mItemId, mRefIdSize)) {
+        return ERROR_IO;
+    }
+    offset += mRefIdSize;
+
+    uint16_t count;
+    if (!source()->getUInt16(offset, &count)) {
+        return ERROR_IO;
+    }
+    offset += 2;
+    size -= (mRefIdSize + 2);
+
+    if (size < count * mRefIdSize) {
+        return ERROR_MALFORMED;
+    }
+
+    for (size_t i = 0; i < count; i++) {
+        uint32_t refItemId;
+        if (!source()->getUInt32Var(offset, &refItemId, mRefIdSize)) {
+            return ERROR_IO;
+        }
+        offset += mRefIdSize;
+        mRefs.push_back(refItemId);
+        ALOGV("item id %d: referencing item id %d", mItemId, refItemId);
+    }
+
+    return OK;
+}
+
+struct IrefBox : public FullBox {
+    IrefBox(DataSourceBase *source, Vector<sp<ItemReference> > *itemRefs) :
+        FullBox(source, FOURCC('i', 'r', 'e', 'f')), mRefIdSize(0), mItemRefs(itemRefs) {}
+
+    status_t parse(off64_t offset, size_t size);
+
+protected:
+    status_t onChunkData(uint32_t type, off64_t offset, size_t size) override;
+
+private:
+    uint32_t mRefIdSize;
+    Vector<sp<ItemReference> > *mItemRefs;
+};
+
+status_t IrefBox::parse(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+    status_t err = parseFullBoxHeader(&offset, &size);
+    if (err != OK) {
+        return err;
+    }
+
+    mRefIdSize = (version() == 0) ? 2 : 4;
+    return parseChunks(offset, size);
+}
+
+status_t IrefBox::onChunkData(uint32_t type, off64_t offset, size_t size) {
+    sp<ItemReference> itemRef = new ItemReference(source(), type, mRefIdSize);
+
+    status_t err = itemRef->parse(offset, size);
+    if (err != OK) {
+        return err;
+    }
+    mItemRefs->push_back(itemRef);
+    return OK;
+}
+
+/////////////////////////////////////////////////////////////////////
+//
+//  ItemProperty related boxes
+//
+
+struct AssociationEntry {
+    uint32_t itemId;
+    bool essential;
+    uint16_t index;
+};
+
+struct ItemProperty : public RefBase {
+    ItemProperty() {}
+
+    virtual void attachTo(ImageItem &/*image*/) const {
+        ALOGW("Unrecognized property");
+    }
+    virtual status_t parse(off64_t /*offset*/, size_t /*size*/) {
+        ALOGW("Unrecognized property");
+        return OK;
+    }
+
+private:
+    DISALLOW_EVIL_CONSTRUCTORS(ItemProperty);
+};
+
+struct IspeBox : public FullBox, public ItemProperty {
+    IspeBox(DataSourceBase *source) :
+        FullBox(source, FOURCC('i', 's', 'p', 'e')), mWidth(0), mHeight(0) {}
+
+    status_t parse(off64_t offset, size_t size) override;
+
+    void attachTo(ImageItem &image) const override {
+        image.width = mWidth;
+        image.height = mHeight;
+    }
+
+private:
+    uint32_t mWidth;
+    uint32_t mHeight;
+};
+
+status_t IspeBox::parse(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    status_t err = parseFullBoxHeader(&offset, &size);
+    if (err != OK) {
+        return err;
+    }
+
+    if (size < 8) {
+        return ERROR_MALFORMED;
+    }
+    if (!source()->getUInt32(offset, &mWidth)
+            || !source()->getUInt32(offset + 4, &mHeight)) {
+        return ERROR_IO;
+    }
+    ALOGV("property ispe: %dx%d", mWidth, mHeight);
+
+    return OK;
+}
+
+struct HvccBox : public Box, public ItemProperty {
+    HvccBox(DataSourceBase *source) :
+        Box(source, FOURCC('h', 'v', 'c', 'C')) {}
+
+    status_t parse(off64_t offset, size_t size) override;
+
+    void attachTo(ImageItem &image) const override {
+        image.hvcc = mHVCC;
+    }
+
+private:
+    sp<ABuffer> mHVCC;
+};
+
+status_t HvccBox::parse(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    mHVCC = new ABuffer(size);
+
+    if (mHVCC->data() == NULL) {
+        ALOGE("b/28471206");
+        return NO_MEMORY;
+    }
+
+    if (source()->readAt(offset, mHVCC->data(), size) < (ssize_t)size) {
+        return ERROR_IO;
+    }
+
+    ALOGV("property hvcC");
+
+    return OK;
+}
+
+struct IrotBox : public Box, public ItemProperty {
+    IrotBox(DataSourceBase *source) :
+        Box(source, FOURCC('i', 'r', 'o', 't')), mAngle(0) {}
+
+    status_t parse(off64_t offset, size_t size) override;
+
+    void attachTo(ImageItem &image) const override {
+        image.rotation = mAngle * 90;
+    }
+
+private:
+    uint8_t mAngle;
+};
+
+status_t IrotBox::parse(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    if (size < 1) {
+        return ERROR_MALFORMED;
+    }
+    if (source()->readAt(offset, &mAngle, 1) != 1) {
+        return ERROR_IO;
+    }
+    mAngle &= 0x3;
+    ALOGV("property irot: %d", mAngle);
+
+    return OK;
+}
+
+struct ColrBox : public Box, public ItemProperty {
+    ColrBox(DataSourceBase *source) :
+        Box(source, FOURCC('c', 'o', 'l', 'r')) {}
+
+    status_t parse(off64_t offset, size_t size) override;
+
+    void attachTo(ImageItem &image) const override {
+        image.icc = mICCData;
+    }
+
+private:
+    sp<ABuffer> mICCData;
+};
+
+status_t ColrBox::parse(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    if (size < 4) {
+        return ERROR_MALFORMED;
+    }
+    uint32_t colour_type;
+    if (!source()->getUInt32(offset, &colour_type)) {
+        return ERROR_IO;
+    }
+    offset += 4;
+    size -= 4;
+    if (colour_type == FOURCC('n', 'c', 'l', 'x')) {
+        return OK;
+    }
+    if ((colour_type != FOURCC('r', 'I', 'C', 'C')) &&
+        (colour_type != FOURCC('p', 'r', 'o', 'f'))) {
+        return ERROR_MALFORMED;
+    }
+
+    mICCData = new ABuffer(size);
+    if (mICCData->data() == NULL) {
+        ALOGE("b/28471206");
+        return NO_MEMORY;
+    }
+
+    if (source()->readAt(offset, mICCData->data(), size) != (ssize_t)size) {
+        return ERROR_IO;
+    }
+
+    ALOGV("property Colr: size %zd", size);
+    return OK;
+}
+
+struct IpmaBox : public FullBox {
+    IpmaBox(DataSourceBase *source, Vector<AssociationEntry> *associations) :
+        FullBox(source, FOURCC('i', 'p', 'm', 'a')), mAssociations(associations) {}
+
+    status_t parse(off64_t offset, size_t size);
+private:
+    Vector<AssociationEntry> *mAssociations;
+};
+
+status_t IpmaBox::parse(off64_t offset, size_t size) {
+    status_t err = parseFullBoxHeader(&offset, &size);
+    if (err != OK) {
+        return err;
+    }
+
+    if (size < 4) {
+        return ERROR_MALFORMED;
+    }
+    uint32_t entryCount;
+    if (!source()->getUInt32(offset, &entryCount)) {
+        return ERROR_IO;
+    }
+    offset += 4;
+    size -= 4;
+
+    for (size_t k = 0; k < entryCount; ++k) {
+        uint32_t itemId = 0;
+        size_t itemIdSize = (version() < 1) ? 2 : 4;
+
+        if (size < itemIdSize + 1) {
+            return ERROR_MALFORMED;
+        }
+
+        if (!source()->getUInt32Var(offset, &itemId, itemIdSize)) {
+            return ERROR_IO;
+        }
+        offset += itemIdSize;
+        size -= itemIdSize;
+
+        uint8_t associationCount;
+        if (!source()->readAt(offset, &associationCount, 1)) {
+            return ERROR_IO;
+        }
+        offset++;
+        size--;
+
+        for (size_t i = 0; i < associationCount; ++i) {
+            size_t propIndexSize = (flags() & 1) ? 2 : 1;
+            if (size < propIndexSize) {
+                return ERROR_MALFORMED;
+            }
+            uint16_t propIndex;
+            if (!source()->getUInt16Var(offset, &propIndex, propIndexSize)) {
+                return ERROR_IO;
+            }
+            offset += propIndexSize;
+            size -= propIndexSize;
+            uint16_t bitmask = (1 << (8 * propIndexSize - 1));
+            AssociationEntry entry = {
+                    .itemId = itemId,
+                    .essential = !!(propIndex & bitmask),
+                    .index = (uint16_t) (propIndex & ~bitmask)
+            };
+
+            ALOGV("item id %d associated to property %d (essential %d)",
+                    itemId, entry.index, entry.essential);
+
+            mAssociations->push_back(entry);
+        }
+    }
+
+    return OK;
+}
+
+struct IpcoBox : public Box {
+    IpcoBox(DataSourceBase *source, Vector<sp<ItemProperty> > *properties) :
+        Box(source, FOURCC('i', 'p', 'c', 'o')), mItemProperties(properties) {}
+
+    status_t parse(off64_t offset, size_t size);
+protected:
+    status_t onChunkData(uint32_t type, off64_t offset, size_t size) override;
+
+private:
+    Vector<sp<ItemProperty> > *mItemProperties;
+};
+
+status_t IpcoBox::parse(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+    // push dummy as the index is 1-based
+    mItemProperties->push_back(new ItemProperty());
+    return parseChunks(offset, size);
+}
+
+status_t IpcoBox::onChunkData(uint32_t type, off64_t offset, size_t size) {
+    sp<ItemProperty> itemProperty;
+    switch(type) {
+        case FOURCC('h', 'v', 'c', 'C'):
+        {
+            itemProperty = new HvccBox(source());
+            break;
+        }
+        case FOURCC('i', 's', 'p', 'e'):
+        {
+            itemProperty = new IspeBox(source());
+            break;
+        }
+        case FOURCC('i', 'r', 'o', 't'):
+        {
+            itemProperty = new IrotBox(source());
+            break;
+        }
+        case FOURCC('c', 'o', 'l', 'r'):
+        {
+            itemProperty = new ColrBox(source());
+            break;
+        }
+        default:
+        {
+            // push dummy to maintain correct item property index
+            itemProperty = new ItemProperty();
+            break;
+        }
+    }
+    status_t err = itemProperty->parse(offset, size);
+    if (err != OK) {
+        return err;
+    }
+    mItemProperties->push_back(itemProperty);
+    return OK;
+}
+
+struct IprpBox : public Box {
+    IprpBox(DataSourceBase *source,
+            Vector<sp<ItemProperty> > *properties,
+            Vector<AssociationEntry> *associations) :
+        Box(source, FOURCC('i', 'p', 'r', 'p')),
+        mProperties(properties), mAssociations(associations) {}
+
+    status_t parse(off64_t offset, size_t size);
+protected:
+    status_t onChunkData(uint32_t type, off64_t offset, size_t size) override;
+
+private:
+    Vector<sp<ItemProperty> > *mProperties;
+    Vector<AssociationEntry> *mAssociations;
+};
+
+status_t IprpBox::parse(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    status_t err = parseChunks(offset, size);
+    if (err != OK) {
+        return err;
+    }
+    return OK;
+}
+
+status_t IprpBox::onChunkData(uint32_t type, off64_t offset, size_t size) {
+    switch(type) {
+        case FOURCC('i', 'p', 'c', 'o'):
+        {
+            IpcoBox ipcoBox(source(), mProperties);
+            return ipcoBox.parse(offset, size);
+        }
+        case FOURCC('i', 'p', 'm', 'a'):
+        {
+            IpmaBox ipmaBox(source(), mAssociations);
+            return ipmaBox.parse(offset, size);
+        }
+        default:
+        {
+            ALOGW("Unrecognized box.");
+            break;
+        }
+    }
+    return OK;
+}
+
+/////////////////////////////////////////////////////////////////////
+//
+//  ItemInfo related boxes
+//
+struct ItemInfo {
+    uint32_t itemId;
+    uint32_t itemType;
+    bool hidden;
+};
+
+struct InfeBox : public FullBox {
+    InfeBox(DataSourceBase *source) :
+        FullBox(source, FOURCC('i', 'n', 'f', 'e')) {}
+
+    status_t parse(off64_t offset, size_t size, ItemInfo *itemInfo);
+
+private:
+    bool parseNullTerminatedString(off64_t *offset, size_t *size, String8 *out);
+};
+
+bool InfeBox::parseNullTerminatedString(
+        off64_t *offset, size_t *size, String8 *out) {
+    char tmp;
+    Vector<char> buf;
+    buf.setCapacity(256);
+    off64_t newOffset = *offset;
+    off64_t stopOffset = *offset + *size;
+    while (newOffset < stopOffset) {
+        if (!source()->readAt(newOffset++, &tmp, 1)) {
+            return false;
+        }
+        buf.push_back(tmp);
+        if (tmp == 0) {
+            out->setTo(buf.array());
+
+            *offset = newOffset;
+            *size = stopOffset - newOffset;
+
+            return true;
+        }
+    }
+    return false;
+}
+
+status_t InfeBox::parse(off64_t offset, size_t size, ItemInfo *itemInfo) {
+    status_t err = parseFullBoxHeader(&offset, &size);
+    if (err != OK) {
+        return err;
+    }
+
+    if (version() == 0 || version() == 1) {
+        return ERROR_UNSUPPORTED;
+    } else { // version >= 2
+        uint32_t item_id;
+        size_t itemIdSize = (version() == 2) ? 2 : 4;
+        if (size < itemIdSize + 6) {
+            return ERROR_MALFORMED;
+        }
+        if (!source()->getUInt32Var(offset, &item_id, itemIdSize)) {
+            return ERROR_IO;
+        }
+        ALOGV("item_id %d", item_id);
+        offset += itemIdSize;
+        uint16_t item_protection_index;
+        if (!source()->getUInt16(offset, &item_protection_index)) {
+            return ERROR_IO;
+        }
+        ALOGV("item_protection_index %d", item_protection_index);
+        offset += 2;
+        uint32_t item_type;
+        if (!source()->getUInt32(offset, &item_type)) {
+            return ERROR_IO;
+        }
+
+        itemInfo->itemId = item_id;
+        itemInfo->itemType = item_type;
+        // According to HEIF spec, (flags & 1) indicates the image is hidden
+        // and not supposed to be displayed.
+        itemInfo->hidden = (flags() & 1);
+
+        char itemTypeString[5];
+        MakeFourCCString(item_type, itemTypeString);
+        ALOGV("item_type %s", itemTypeString);
+        offset += 4;
+        size -= itemIdSize + 6;
+
+        String8 item_name;
+        if (!parseNullTerminatedString(&offset, &size, &item_name)) {
+            return ERROR_MALFORMED;
+        }
+        ALOGV("item_name %s", item_name.c_str());
+
+        if (item_type == FOURCC('m', 'i', 'm', 'e')) {
+            String8 content_type;
+            if (!parseNullTerminatedString(&offset, &size, &content_type)) {
+                return ERROR_MALFORMED;
+            }
+
+            // content_encoding is optional; can be omitted if would be empty
+            if (size > 0) {
+                String8 content_encoding;
+                if (!parseNullTerminatedString(&offset, &size, &content_encoding)) {
+                    return ERROR_MALFORMED;
+                }
+            }
+        } else if (item_type == FOURCC('u', 'r', 'i', ' ')) {
+            String8 item_uri_type;
+            if (!parseNullTerminatedString(&offset, &size, &item_uri_type)) {
+                return ERROR_MALFORMED;
+            }
+        }
+    }
+    return OK;
+}
+
+struct IinfBox : public FullBox {
+    IinfBox(DataSourceBase *source, Vector<ItemInfo> *itemInfos) :
+        FullBox(source, FOURCC('i', 'i', 'n', 'f')),
+        mItemInfos(itemInfos), mHasGrids(false) {}
+
+    status_t parse(off64_t offset, size_t size);
+
+    bool hasGrids() { return mHasGrids; }
+
+protected:
+    status_t onChunkData(uint32_t type, off64_t offset, size_t size) override;
+
+private:
+    Vector<ItemInfo> *mItemInfos;
+    bool mHasGrids;
+};
+
+status_t IinfBox::parse(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    status_t err = parseFullBoxHeader(&offset, &size);
+    if (err != OK) {
+        return err;
+    }
+
+    size_t entryCountSize = version() == 0 ? 2 : 4;
+    if (size < entryCountSize) {
+        return ERROR_MALFORMED;
+    }
+    uint32_t entry_count;
+    if (!source()->getUInt32Var(offset, &entry_count, entryCountSize)) {
+        return ERROR_IO;
+    }
+    ALOGV("entry_count %d", entry_count);
+
+    off64_t stopOffset = offset + size;
+    offset += entryCountSize;
+    for (size_t i = 0; i < entry_count && offset < stopOffset; i++) {
+        ALOGV("entry %zu", i);
+        status_t err = parseChunk(&offset);
+        if (err != OK) {
+            return err;
+        }
+    }
+    if (offset != stopOffset) {
+        return ERROR_MALFORMED;
+    }
+
+    return OK;
+}
+
+status_t IinfBox::onChunkData(uint32_t type, off64_t offset, size_t size) {
+    if (type != FOURCC('i', 'n', 'f', 'e')) {
+        return OK;
+    }
+
+    InfeBox infeBox(source());
+    ItemInfo itemInfo;
+    status_t err = infeBox.parse(offset, size, &itemInfo);
+    if (err == OK) {
+        mItemInfos->push_back(itemInfo);
+        mHasGrids |= (itemInfo.itemType == FOURCC('g', 'r', 'i', 'd'));
+    }
+    // InfeBox parse returns ERROR_UNSUPPORTED if the box if an unsupported
+    // version. Ignore this error as it's not fatal.
+    return (err == ERROR_UNSUPPORTED) ? OK : err;
+}
+
+//////////////////////////////////////////////////////////////////
+
+ItemTable::ItemTable(DataSourceBase *source)
+    : mDataSource(source),
+      mPrimaryItemId(0),
+      mIdatOffset(0),
+      mIdatSize(0),
+      mImageItemsValid(false),
+      mCurrentItemIndex(0) {
+    mRequiredBoxes.insert('iprp');
+    mRequiredBoxes.insert('iloc');
+    mRequiredBoxes.insert('pitm');
+    mRequiredBoxes.insert('iinf');
+}
+
+ItemTable::~ItemTable() {}
+
+status_t ItemTable::parse(uint32_t type, off64_t data_offset, size_t chunk_data_size) {
+    switch(type) {
+        case FOURCC('i', 'l', 'o', 'c'):
+        {
+            return parseIlocBox(data_offset, chunk_data_size);
+        }
+        case FOURCC('i', 'i', 'n', 'f'):
+        {
+            return parseIinfBox(data_offset, chunk_data_size);
+        }
+        case FOURCC('i', 'p', 'r', 'p'):
+        {
+            return parseIprpBox(data_offset, chunk_data_size);
+        }
+        case FOURCC('p', 'i', 't', 'm'):
+        {
+            return parsePitmBox(data_offset, chunk_data_size);
+        }
+        case FOURCC('i', 'd', 'a', 't'):
+        {
+            return parseIdatBox(data_offset, chunk_data_size);
+        }
+        case FOURCC('i', 'r', 'e', 'f'):
+        {
+            return parseIrefBox(data_offset, chunk_data_size);
+        }
+        case FOURCC('i', 'p', 'r', 'o'):
+        {
+            ALOGW("ipro box not supported!");
+            break;
+        }
+        default:
+        {
+            ALOGW("unrecognized box type: 0x%x", type);
+            break;
+        }
+    }
+    return ERROR_UNSUPPORTED;
+}
+
+status_t ItemTable::parseIlocBox(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    IlocBox ilocBox(mDataSource, &mItemLocs);
+    status_t err = ilocBox.parse(offset, size);
+    if (err != OK) {
+        return err;
+    }
+
+    if (ilocBox.hasConstructMethod1()) {
+        mRequiredBoxes.insert('idat');
+    }
+
+    return buildImageItemsIfPossible('iloc');
+}
+
+status_t ItemTable::parseIinfBox(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    IinfBox iinfBox(mDataSource, &mItemInfos);
+    status_t err = iinfBox.parse(offset, size);
+    if (err != OK) {
+        return err;
+    }
+
+    if (iinfBox.hasGrids()) {
+        mRequiredBoxes.insert('iref');
+    }
+
+    return buildImageItemsIfPossible('iinf');
+}
+
+status_t ItemTable::parsePitmBox(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    PitmBox pitmBox(mDataSource);
+    status_t err = pitmBox.parse(offset, size, &mPrimaryItemId);
+    if (err != OK) {
+        return err;
+    }
+
+    return buildImageItemsIfPossible('pitm');
+}
+
+status_t ItemTable::parseIprpBox(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    IprpBox iprpBox(mDataSource, &mItemProperties, &mAssociations);
+    status_t err = iprpBox.parse(offset, size);
+    if (err != OK) {
+        return err;
+    }
+
+    return buildImageItemsIfPossible('iprp');
+}
+
+status_t ItemTable::parseIdatBox(off64_t offset, size_t size) {
+    ALOGV("%s: idat offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    // only remember the offset and size of idat box for later use
+    mIdatOffset = offset;
+    mIdatSize = size;
+
+    return buildImageItemsIfPossible('idat');
+}
+
+status_t ItemTable::parseIrefBox(off64_t offset, size_t size) {
+    ALOGV("%s: offset %lld, size %zu", __FUNCTION__, (long long)offset, size);
+
+    IrefBox irefBox(mDataSource, &mItemReferences);
+    status_t err = irefBox.parse(offset, size);
+    if (err != OK) {
+        return err;
+    }
+
+    return buildImageItemsIfPossible('iref');
+}
+
+status_t ItemTable::buildImageItemsIfPossible(uint32_t type) {
+    if (mImageItemsValid) {
+        return OK;
+    }
+
+    mBoxesSeen.insert(type);
+
+    // need at least 'iprp', 'iloc', 'pitm', 'iinf';
+    // need 'idat' if any items used construction_method of 2;
+    // need 'iref' if there are grids.
+    if (!std::includes(
+            mBoxesSeen.begin(), mBoxesSeen.end(),
+            mRequiredBoxes.begin(), mRequiredBoxes.end())) {
+        return OK;
+    }
+
+    ALOGV("building image table...");
+
+    for (size_t i = 0; i < mItemInfos.size(); i++) {
+        const ItemInfo &info = mItemInfos[i];
+
+        // Only handle 3 types of items, all others are ignored:
+        //   'grid': derived image from tiles
+        //   'hvc1': coded image (or tile)
+        //   'Exif': EXIF metadata
+        if (info.itemType != FOURCC('g', 'r', 'i', 'd') &&
+            info.itemType != FOURCC('h', 'v', 'c', '1') &&
+            info.itemType != FOURCC('E', 'x', 'i', 'f')) {
+            continue;
+        }
+
+        ssize_t itemIndex = mItemIdToItemMap.indexOfKey(info.itemId);
+        if (itemIndex >= 0) {
+            ALOGW("ignoring duplicate image item id %d", info.itemId);
+            continue;
+        }
+
+        ssize_t ilocIndex = mItemLocs.indexOfKey(info.itemId);
+        if (ilocIndex < 0) {
+            ALOGE("iloc missing for image item id %d", info.itemId);
+            continue;
+        }
+        const ItemLoc &iloc = mItemLocs[ilocIndex];
+
+        off64_t offset;
+        size_t size;
+        if (iloc.getLoc(&offset, &size, mIdatOffset, mIdatSize) != OK) {
+            return ERROR_MALFORMED;
+        }
+
+        if (info.itemType == FOURCC('E', 'x', 'i', 'f')) {
+            // Only add if the Exif data is non-empty. The first 4 bytes contain
+            // the offset to TIFF header, which the Exif parser doesn't use.
+            if (size > 4) {
+                ExifItem exifItem = {
+                        .offset = offset,
+                        .size = size,
+                };
+                mItemIdToExifMap.add(info.itemId, exifItem);
+            }
+            continue;
+        }
+
+        ImageItem image(info.itemType, info.itemId, info.hidden);
+
+        ALOGV("adding %s: itemId %d", image.isGrid() ? "grid" : "image", info.itemId);
+
+        if (image.isGrid()) {
+            // ImageGrid struct is at least 8-byte, at most 12-byte (if flags&1)
+            if (size < 8 || size > 12) {
+                return ERROR_MALFORMED;
+            }
+            uint8_t buf[12];
+            if (!mDataSource->readAt(offset, buf, size)) {
+                return ERROR_IO;
+            }
+
+            image.rows = buf[2] + 1;
+            image.columns = buf[3] + 1;
+
+            ALOGV("rows %d, columans %d", image.rows, image.columns);
+        } else {
+            image.offset = offset;
+            image.size = size;
+        }
+        mItemIdToItemMap.add(info.itemId, image);
+    }
+
+    for (size_t i = 0; i < mAssociations.size(); i++) {
+        attachProperty(mAssociations[i]);
+    }
+
+    for (size_t i = 0; i < mItemReferences.size(); i++) {
+        mItemReferences[i]->apply(mItemIdToItemMap, mItemIdToExifMap);
+    }
+
+    bool foundPrimary = false;
+    for (size_t i = 0; i < mItemIdToItemMap.size(); i++) {
+        // add all non-hidden images, also add the primary even if it's marked
+        // hidden, in case the primary is set to a thumbnail
+        bool isPrimary = (mItemIdToItemMap[i].itemId == mPrimaryItemId);
+        if (!mItemIdToItemMap[i].hidden || isPrimary) {
+            mDisplayables.push_back(i);
+        }
+        foundPrimary |= isPrimary;
+    }
+
+    ALOGV("found %zu displayables", mDisplayables.size());
+
+    // fail if no displayables are found
+    if (mDisplayables.empty()) {
+        return ERROR_MALFORMED;
+    }
+
+    // if the primary item id is invalid, set primary to the first displayable
+    if (!foundPrimary) {
+        mPrimaryItemId = mItemIdToItemMap[mDisplayables[0]].itemId;
+    }
+
+    mImageItemsValid = true;
+    return OK;
+}
+
+void ItemTable::attachProperty(const AssociationEntry &association) {
+    ssize_t itemIndex = mItemIdToItemMap.indexOfKey(association.itemId);
+
+    // ignore non-image items
+    if (itemIndex < 0) {
+        return;
+    }
+
+    uint16_t propertyIndex = association.index;
+    if (propertyIndex >= mItemProperties.size()) {
+        ALOGW("Ignoring invalid property index %d", propertyIndex);
+        return;
+    }
+
+    ALOGV("attach property %d to item id %d)",
+            propertyIndex, association.itemId);
+
+    mItemProperties[propertyIndex]->attachTo(mItemIdToItemMap.editValueAt(itemIndex));
+}
+
+uint32_t ItemTable::countImages() const {
+    return mImageItemsValid ? mDisplayables.size() : 0;
+}
+
+sp<MetaData> ItemTable::getImageMeta(const uint32_t imageIndex) {
+    if (!mImageItemsValid) {
+        return NULL;
+    }
+
+    if (imageIndex >= mDisplayables.size()) {
+        ALOGE("%s: invalid image index %u", __FUNCTION__, imageIndex);
+        return NULL;
+    }
+    const uint32_t itemIndex = mDisplayables[imageIndex];
+    ALOGV("image[%u]: item index %u", imageIndex, itemIndex);
+
+    const ImageItem *image = &mItemIdToItemMap[itemIndex];
+
+    ssize_t tileItemIndex = -1;
+    if (image->isGrid()) {
+        if (image->dimgRefs.empty()) {
+            return NULL;
+        }
+        tileItemIndex = mItemIdToItemMap.indexOfKey(image->dimgRefs[0]);
+        if (tileItemIndex < 0) {
+            return NULL;
+        }
+    }
+
+    sp<MetaData> meta = new MetaData;
+    meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_IMAGE_ANDROID_HEIC);
+
+    if (image->itemId == mPrimaryItemId) {
+        meta->setInt32(kKeyTrackIsDefault, 1);
+    }
+
+    ALOGV("image[%u]: size %dx%d", imageIndex, image->width, image->height);
+
+    meta->setInt32(kKeyWidth, image->width);
+    meta->setInt32(kKeyHeight, image->height);
+    if (image->rotation != 0) {
+        // Rotation angle in HEIF is CCW, convert to CW here to be
+        // consistent with the other media formats.
+        switch(image->rotation) {
+            case 90: meta->setInt32(kKeyRotation, 270); break;
+            case 180: meta->setInt32(kKeyRotation, 180); break;
+            case 270: meta->setInt32(kKeyRotation, 90); break;
+            default: break; // don't set if invalid
+        }
+    }
+    meta->setInt32(kKeyMaxInputSize, image->width * image->height * 1.5);
+
+    if (!image->thumbnails.empty()) {
+        ssize_t thumbItemIndex = mItemIdToItemMap.indexOfKey(image->thumbnails[0]);
+        if (thumbItemIndex >= 0) {
+            const ImageItem &thumbnail = mItemIdToItemMap[thumbItemIndex];
+
+            meta->setInt32(kKeyThumbnailWidth, thumbnail.width);
+            meta->setInt32(kKeyThumbnailHeight, thumbnail.height);
+            meta->setData(kKeyThumbnailHVCC, kTypeHVCC,
+                    thumbnail.hvcc->data(), thumbnail.hvcc->size());
+            ALOGV("image[%u]: thumbnail: size %dx%d, item index %zd",
+                    imageIndex, thumbnail.width, thumbnail.height, thumbItemIndex);
+        } else {
+            ALOGW("%s: Referenced thumbnail does not exist!", __FUNCTION__);
+        }
+    }
+
+    if (image->isGrid()) {
+        meta->setInt32(kKeyGridRows, image->rows);
+        meta->setInt32(kKeyGridCols, image->columns);
+
+        // point image to the first tile for grid size and HVCC
+        image = &mItemIdToItemMap.editValueAt(tileItemIndex);
+        meta->setInt32(kKeyTileWidth, image->width);
+        meta->setInt32(kKeyTileHeight, image->height);
+        meta->setInt32(kKeyMaxInputSize, image->width * image->height * 1.5);
+    }
+
+    if (image->hvcc == NULL) {
+        ALOGE("%s: hvcc is missing for image[%u]!", __FUNCTION__, imageIndex);
+        return NULL;
+    }
+    meta->setData(kKeyHVCC, kTypeHVCC, image->hvcc->data(), image->hvcc->size());
+
+    if (image->icc != NULL) {
+        meta->setData(kKeyIccProfile, 0, image->icc->data(), image->icc->size());
+    }
+    return meta;
+}
+
+status_t ItemTable::findImageItem(const uint32_t imageIndex, uint32_t *itemIndex) {
+    if (!mImageItemsValid) {
+        return INVALID_OPERATION;
+    }
+
+    if (imageIndex >= mDisplayables.size()) {
+        ALOGE("%s: invalid image index %d", __FUNCTION__, imageIndex);
+        return BAD_VALUE;
+    }
+
+    *itemIndex = mDisplayables[imageIndex];
+
+    ALOGV("image[%u]: item index %u", imageIndex, *itemIndex);
+    return OK;
+}
+
+status_t ItemTable::findThumbnailItem(const uint32_t imageIndex, uint32_t *itemIndex) {
+    if (!mImageItemsValid) {
+        return INVALID_OPERATION;
+    }
+
+    if (imageIndex >= mDisplayables.size()) {
+        ALOGE("%s: invalid image index %d", __FUNCTION__, imageIndex);
+        return BAD_VALUE;
+    }
+
+    uint32_t masterItemIndex = mDisplayables[imageIndex];
+
+    const ImageItem &masterImage = mItemIdToItemMap[masterItemIndex];
+    if (masterImage.thumbnails.empty()) {
+        *itemIndex = masterItemIndex;
+        return OK;
+    }
+
+    ssize_t thumbItemIndex = mItemIdToItemMap.indexOfKey(masterImage.thumbnails[0]);
+    if (thumbItemIndex < 0) {
+        // Do not return the master image in this case, fail it so that the
+        // thumbnail extraction code knows we really don't have it.
+        return INVALID_OPERATION;
+    }
+
+    *itemIndex = thumbItemIndex;
+    return OK;
+}
+
+status_t ItemTable::getImageOffsetAndSize(
+        uint32_t *itemIndex, off64_t *offset, size_t *size) {
+    if (!mImageItemsValid) {
+        return INVALID_OPERATION;
+    }
+
+    if (itemIndex != NULL) {
+        if (*itemIndex >= mItemIdToItemMap.size()) {
+            ALOGE("%s: Bad item index!", __FUNCTION__);
+            return BAD_VALUE;
+        }
+        mCurrentItemIndex = *itemIndex;
+    }
+
+    ImageItem &image = mItemIdToItemMap.editValueAt(mCurrentItemIndex);
+    if (image.isGrid()) {
+        uint32_t tileItemId;
+        status_t err = image.getNextTileItemId(&tileItemId, itemIndex != NULL);
+        if (err != OK) {
+            return err;
+        }
+        ssize_t tileItemIndex = mItemIdToItemMap.indexOfKey(tileItemId);
+        if (tileItemIndex < 0) {
+            return ERROR_END_OF_STREAM;
+        }
+        *offset = mItemIdToItemMap[tileItemIndex].offset;
+        *size = mItemIdToItemMap[tileItemIndex].size;
+    } else {
+        if (itemIndex == NULL) {
+            // For single images, we only allow it to be read once, after that
+            // it's EOS.  New item index must be requested each time.
+            return ERROR_END_OF_STREAM;
+        }
+        *offset = mItemIdToItemMap[mCurrentItemIndex].offset;
+        *size = mItemIdToItemMap[mCurrentItemIndex].size;
+    }
+
+    return OK;
+}
+
+status_t ItemTable::getExifOffsetAndSize(off64_t *offset, size_t *size) {
+    if (!mImageItemsValid) {
+        return INVALID_OPERATION;
+    }
+
+    ssize_t itemIndex = mItemIdToItemMap.indexOfKey(mPrimaryItemId);
+
+    // this should not happen, something's seriously wrong.
+    if (itemIndex < 0) {
+        return INVALID_OPERATION;
+    }
+
+    const ImageItem &image = mItemIdToItemMap[itemIndex];
+    if (image.cdscRefs.size() == 0) {
+        return NAME_NOT_FOUND;
+    }
+
+    ssize_t exifIndex = mItemIdToExifMap.indexOfKey(image.cdscRefs[0]);
+    if (exifIndex < 0) {
+        return NAME_NOT_FOUND;
+    }
+
+    // skip the first 4-byte of the offset to TIFF header
+    *offset = mItemIdToExifMap[exifIndex].offset + 4;
+    *size = mItemIdToExifMap[exifIndex].size - 4;
+    return OK;
+}
+
+} // namespace heif
+
+}  // namespace android

diff --git a/media/extractors/mp4/ItemTable.h b/media/extractors/mp4/ItemTable.h
new file mode 100644
index 0000000..536dcb0
--- /dev/null
+++ b/media/extractors/mp4/ItemTable.h

@@ -0,0 +1,102 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ITEM_TABLE_H_
+#define ITEM_TABLE_H_
+
+#include <set>
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <utils/KeyedVector.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+class DataSourceBase;
+class MetaData;
+
+namespace heif {
+
+struct AssociationEntry;
+struct ImageItem;
+struct ExifItem;
+struct ItemLoc;
+struct ItemInfo;
+struct ItemProperty;
+struct ItemReference;
+
+/*
+ * ItemTable keeps track of all image items (including coded images, grids and
+ * tiles) inside a HEIF still image (ISO/IEC FDIS 23008-12.2:2017(E)).
+ */
+
+class ItemTable : public RefBase {
+public:
+    explicit ItemTable(DataSourceBase *source);
+
+    status_t parse(uint32_t type, off64_t offset, size_t size);
+
+    bool isValid() { return mImageItemsValid; }
+    uint32_t countImages() const;
+    sp<MetaData> getImageMeta(const uint32_t imageIndex);
+    status_t findImageItem(const uint32_t imageIndex, uint32_t *itemIndex);
+    status_t findThumbnailItem(const uint32_t imageIndex, uint32_t *itemIndex);
+    status_t getImageOffsetAndSize(
+            uint32_t *itemIndex, off64_t *offset, size_t *size);
+    status_t getExifOffsetAndSize(off64_t *offset, size_t *size);
+
+protected:
+    ~ItemTable();
+
+private:
+    DataSourceBase *mDataSource;
+
+    KeyedVector<uint32_t, ItemLoc> mItemLocs;
+    Vector<ItemInfo> mItemInfos;
+    Vector<AssociationEntry> mAssociations;
+    Vector<sp<ItemProperty> > mItemProperties;
+    Vector<sp<ItemReference> > mItemReferences;
+
+    uint32_t mPrimaryItemId;
+    off64_t mIdatOffset;
+    size_t mIdatSize;
+
+    std::set<uint32_t> mRequiredBoxes;
+    std::set<uint32_t> mBoxesSeen;
+
+    bool mImageItemsValid;
+    uint32_t mCurrentItemIndex;
+    KeyedVector<uint32_t, ImageItem> mItemIdToItemMap;
+    KeyedVector<uint32_t, ExifItem> mItemIdToExifMap;
+    Vector<uint32_t> mDisplayables;
+
+    status_t parseIlocBox(off64_t offset, size_t size);
+    status_t parseIinfBox(off64_t offset, size_t size);
+    status_t parsePitmBox(off64_t offset, size_t size);
+    status_t parseIprpBox(off64_t offset, size_t size);
+    status_t parseIdatBox(off64_t offset, size_t size);
+    status_t parseIrefBox(off64_t offset, size_t size);
+
+    void attachProperty(const AssociationEntry &association);
+    status_t buildImageItemsIfPossible(uint32_t type);
+
+    DISALLOW_EVIL_CONSTRUCTORS(ItemTable);
+};
+
+} // namespace heif
+} // namespace android
+
+#endif  // ITEM_TABLE_H_

diff --git a/media/libstagefright/matroska/MODULE_LICENSE_APACHE2 b/media/extractors/mp4/MODULE_LICENSE_APACHE2
similarity index 100%
copy from media/libstagefright/matroska/MODULE_LICENSE_APACHE2
copy to media/extractors/mp4/MODULE_LICENSE_APACHE2


diff --git a/media/extractors/mp4/MPEG4Extractor.cpp b/media/extractors/mp4/MPEG4Extractor.cpp
new file mode 100644
index 0000000..7b3b81d
--- /dev/null
+++ b/media/extractors/mp4/MPEG4Extractor.cpp

@@ -0,0 +1,5585 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MPEG4Extractor"
+
+#include <ctype.h>
+#include <inttypes.h>
+#include <memory>
+#include <stdint.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <utils/Log.h>
+
+#include "MPEG4Extractor.h"
+#include "SampleTable.h"
+#include "ItemTable.h"
+#include "include/ESDS.h"
+
+#include <media/ExtractorUtils.h>
+#include <media/MediaTrack.h>
+#include <media/stagefright/foundation/ABitReader.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/AUtils.h>
+#include <media/stagefright/foundation/ByteUtils.h>
+#include <media/stagefright/foundation/ColorUtils.h>
+#include <media/stagefright/foundation/avc_utils.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/MediaBufferBase.h>
+#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <utils/String8.h>
+
+#include <byteswap.h>
+#include "include/ID3.h"
+
+#ifndef UINT32_MAX
+#define UINT32_MAX       (4294967295U)
+#endif
+
+namespace android {
+
+enum {
+    // max track header chunk to return
+    kMaxTrackHeaderSize = 32,
+
+    // maximum size of an atom. Some atoms can be bigger according to the spec,
+    // but we only allow up to this size.
+    kMaxAtomSize = 64 * 1024 * 1024,
+};
+
+class MPEG4Source : public MediaTrack {
+public:
+    // Caller retains ownership of both "dataSource" and "sampleTable".
+    MPEG4Source(MetaDataBase &format,
+                DataSourceBase *dataSource,
+                int32_t timeScale,
+                const sp<SampleTable> &sampleTable,
+                Vector<SidxEntry> &sidx,
+                const Trex *trex,
+                off64_t firstMoofOffset,
+                const sp<ItemTable> &itemTable);
+    virtual status_t init();
+
+    virtual status_t start(MetaDataBase *params = NULL);
+    virtual status_t stop();
+
+    virtual status_t getFormat(MetaDataBase &);
+
+    virtual status_t read(MediaBufferBase **buffer, const ReadOptions *options = NULL);
+    virtual bool supportNonblockingRead() { return true; }
+    virtual status_t fragmentedRead(MediaBufferBase **buffer, const ReadOptions *options = NULL);
+
+    virtual ~MPEG4Source();
+
+private:
+    Mutex mLock;
+
+    MetaDataBase &mFormat;
+    DataSourceBase *mDataSource;
+    int32_t mTimescale;
+    sp<SampleTable> mSampleTable;
+    uint32_t mCurrentSampleIndex;
+    uint32_t mCurrentFragmentIndex;
+    Vector<SidxEntry> &mSegments;
+    const Trex *mTrex;
+    off64_t mFirstMoofOffset;
+    off64_t mCurrentMoofOffset;
+    off64_t mNextMoofOffset;
+    uint32_t mCurrentTime;
+    int32_t mLastParsedTrackId;
+    int32_t mTrackId;
+
+    int32_t mCryptoMode;    // passed in from extractor
+    int32_t mDefaultIVSize; // passed in from extractor
+    uint8_t mCryptoKey[16]; // passed in from extractor
+    int32_t mDefaultEncryptedByteBlock;
+    int32_t mDefaultSkipByteBlock;
+    uint32_t mCurrentAuxInfoType;
+    uint32_t mCurrentAuxInfoTypeParameter;
+    int32_t mCurrentDefaultSampleInfoSize;
+    uint32_t mCurrentSampleInfoCount;
+    uint32_t mCurrentSampleInfoAllocSize;
+    uint8_t* mCurrentSampleInfoSizes;
+    uint32_t mCurrentSampleInfoOffsetCount;
+    uint32_t mCurrentSampleInfoOffsetsAllocSize;
+    uint64_t* mCurrentSampleInfoOffsets;
+
+    bool mIsAVC;
+    bool mIsHEVC;
+    size_t mNALLengthSize;
+
+    bool mStarted;
+
+    MediaBufferGroup *mGroup;
+
+    MediaBufferBase *mBuffer;
+
+    bool mWantsNALFragments;
+
+    uint8_t *mSrcBuffer;
+
+    bool mIsHeif;
+    sp<ItemTable> mItemTable;
+
+    size_t parseNALSize(const uint8_t *data) const;
+    status_t parseChunk(off64_t *offset);
+    status_t parseTrackFragmentHeader(off64_t offset, off64_t size);
+    status_t parseTrackFragmentRun(off64_t offset, off64_t size);
+    status_t parseSampleAuxiliaryInformationSizes(off64_t offset, off64_t size);
+    status_t parseSampleAuxiliaryInformationOffsets(off64_t offset, off64_t size);
+    status_t parseClearEncryptedSizes(off64_t offset, bool isSubsampleEncryption, uint32_t flags);
+    status_t parseSampleEncryption(off64_t offset);
+
+    struct TrackFragmentHeaderInfo {
+        enum Flags {
+            kBaseDataOffsetPresent         = 0x01,
+            kSampleDescriptionIndexPresent = 0x02,
+            kDefaultSampleDurationPresent  = 0x08,
+            kDefaultSampleSizePresent      = 0x10,
+            kDefaultSampleFlagsPresent     = 0x20,
+            kDurationIsEmpty               = 0x10000,
+        };
+
+        uint32_t mTrackID;
+        uint32_t mFlags;
+        uint64_t mBaseDataOffset;
+        uint32_t mSampleDescriptionIndex;
+        uint32_t mDefaultSampleDuration;
+        uint32_t mDefaultSampleSize;
+        uint32_t mDefaultSampleFlags;
+
+        uint64_t mDataOffset;
+    };
+    TrackFragmentHeaderInfo mTrackFragmentHeaderInfo;
+
+    struct Sample {
+        off64_t offset;
+        size_t size;
+        uint32_t duration;
+        int32_t compositionOffset;
+        uint8_t iv[16];
+        Vector<size_t> clearsizes;
+        Vector<size_t> encryptedsizes;
+    };
+    Vector<Sample> mCurrentSamples;
+
+    MPEG4Source(const MPEG4Source &);
+    MPEG4Source &operator=(const MPEG4Source &);
+};
+
+// This custom data source wraps an existing one and satisfies requests
+// falling entirely within a cached range from the cache while forwarding
+// all remaining requests to the wrapped datasource.
+// This is used to cache the full sampletable metadata for a single track,
+// possibly wrapping multiple times to cover all tracks, i.e.
+// Each CachedRangedDataSource caches the sampletable metadata for a single track.
+
+struct CachedRangedDataSource : public DataSourceBase {
+    explicit CachedRangedDataSource(DataSourceBase *source);
+    virtual ~CachedRangedDataSource();
+
+    virtual status_t initCheck() const;
+    virtual ssize_t readAt(off64_t offset, void *data, size_t size);
+    virtual status_t getSize(off64_t *size);
+    virtual uint32_t flags();
+
+    status_t setCachedRange(off64_t offset, size_t size, bool assumeSourceOwnershipOnSuccess);
+
+
+private:
+    Mutex mLock;
+
+    DataSourceBase *mSource;
+    bool mOwnsDataSource;
+    off64_t mCachedOffset;
+    size_t mCachedSize;
+    uint8_t *mCache;
+
+    void clearCache();
+
+    CachedRangedDataSource(const CachedRangedDataSource &);
+    CachedRangedDataSource &operator=(const CachedRangedDataSource &);
+};
+
+CachedRangedDataSource::CachedRangedDataSource(DataSourceBase *source)
+    : mSource(source),
+      mOwnsDataSource(false),
+      mCachedOffset(0),
+      mCachedSize(0),
+      mCache(NULL) {
+}
+
+CachedRangedDataSource::~CachedRangedDataSource() {
+    clearCache();
+    if (mOwnsDataSource) {
+        delete (CachedRangedDataSource*)mSource;
+    }
+}
+
+void CachedRangedDataSource::clearCache() {
+    if (mCache) {
+        free(mCache);
+        mCache = NULL;
+    }
+
+    mCachedOffset = 0;
+    mCachedSize = 0;
+}
+
+status_t CachedRangedDataSource::initCheck() const {
+    return mSource->initCheck();
+}
+
+ssize_t CachedRangedDataSource::readAt(off64_t offset, void *data, size_t size) {
+    Mutex::Autolock autoLock(mLock);
+
+    if (isInRange(mCachedOffset, mCachedSize, offset, size)) {
+        memcpy(data, &mCache[offset - mCachedOffset], size);
+        return size;
+    }
+
+    return mSource->readAt(offset, data, size);
+}
+
+status_t CachedRangedDataSource::getSize(off64_t *size) {
+    return mSource->getSize(size);
+}
+
+uint32_t CachedRangedDataSource::flags() {
+    return mSource->flags();
+}
+
+status_t CachedRangedDataSource::setCachedRange(off64_t offset,
+        size_t size,
+        bool assumeSourceOwnershipOnSuccess) {
+    Mutex::Autolock autoLock(mLock);
+
+    clearCache();
+
+    mCache = (uint8_t *)malloc(size);
+
+    if (mCache == NULL) {
+        return -ENOMEM;
+    }
+
+    mCachedOffset = offset;
+    mCachedSize = size;
+
+    ssize_t err = mSource->readAt(mCachedOffset, mCache, mCachedSize);
+
+    if (err < (ssize_t)size) {
+        clearCache();
+
+        return ERROR_IO;
+    }
+    mOwnsDataSource = assumeSourceOwnershipOnSuccess;
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+static const bool kUseHexDump = false;
+
+static const char *FourCC2MIME(uint32_t fourcc) {
+    switch (fourcc) {
+        case FOURCC('m', 'p', '4', 'a'):
+            return MEDIA_MIMETYPE_AUDIO_AAC;
+
+        case FOURCC('s', 'a', 'm', 'r'):
+            return MEDIA_MIMETYPE_AUDIO_AMR_NB;
+
+        case FOURCC('s', 'a', 'w', 'b'):
+            return MEDIA_MIMETYPE_AUDIO_AMR_WB;
+
+        case FOURCC('m', 'p', '4', 'v'):
+            return MEDIA_MIMETYPE_VIDEO_MPEG4;
+
+        case FOURCC('s', '2', '6', '3'):
+        case FOURCC('h', '2', '6', '3'):
+        case FOURCC('H', '2', '6', '3'):
+            return MEDIA_MIMETYPE_VIDEO_H263;
+
+        case FOURCC('a', 'v', 'c', '1'):
+            return MEDIA_MIMETYPE_VIDEO_AVC;
+
+        case FOURCC('h', 'v', 'c', '1'):
+        case FOURCC('h', 'e', 'v', '1'):
+            return MEDIA_MIMETYPE_VIDEO_HEVC;
+        default:
+            ALOGW("Unknown fourcc: %c%c%c%c",
+                   (fourcc >> 24) & 0xff,
+                   (fourcc >> 16) & 0xff,
+                   (fourcc >> 8) & 0xff,
+                   fourcc & 0xff
+                   );
+            return "application/octet-stream";
+    }
+}
+
+static bool AdjustChannelsAndRate(uint32_t fourcc, uint32_t *channels, uint32_t *rate) {
+    if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_AMR_NB, FourCC2MIME(fourcc))) {
+        // AMR NB audio is always mono, 8kHz
+        *channels = 1;
+        *rate = 8000;
+        return true;
+    } else if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_AMR_WB, FourCC2MIME(fourcc))) {
+        // AMR WB audio is always mono, 16kHz
+        *channels = 1;
+        *rate = 16000;
+        return true;
+    }
+    return false;
+}
+
+MPEG4Extractor::MPEG4Extractor(DataSourceBase *source, const char *mime)
+    : mMoofOffset(0),
+      mMoofFound(false),
+      mMdatFound(false),
+      mDataSource(source),
+      mCachedSource(NULL),
+      mInitCheck(NO_INIT),
+      mHeaderTimescale(0),
+      mIsQT(false),
+      mIsHeif(false),
+      mHasMoovBox(false),
+      mPreferHeif(mime != NULL && !strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_HEIF)),
+      mFirstTrack(NULL),
+      mLastTrack(NULL) {
+    ALOGV("mime=%s, mPreferHeif=%d", mime, mPreferHeif);
+}
+
+MPEG4Extractor::~MPEG4Extractor() {
+    Track *track = mFirstTrack;
+    while (track) {
+        Track *next = track->next;
+
+        delete track;
+        track = next;
+    }
+    mFirstTrack = mLastTrack = NULL;
+
+    for (size_t i = 0; i < mPssh.size(); i++) {
+        delete [] mPssh[i].data;
+    }
+    mPssh.clear();
+
+    delete mCachedSource;
+}
+
+uint32_t MPEG4Extractor::flags() const {
+    return CAN_PAUSE |
+            ((mMoofOffset == 0 || mSidxEntries.size() != 0) ?
+                    (CAN_SEEK_BACKWARD | CAN_SEEK_FORWARD | CAN_SEEK) : 0);
+}
+
+status_t MPEG4Extractor::getMetaData(MetaDataBase &meta) {
+    status_t err;
+    if ((err = readMetaData()) != OK) {
+        return UNKNOWN_ERROR;
+    }
+    meta = mFileMetaData;
+    return OK;
+}
+
+size_t MPEG4Extractor::countTracks() {
+    status_t err;
+    if ((err = readMetaData()) != OK) {
+        ALOGV("MPEG4Extractor::countTracks: no tracks");
+        return 0;
+    }
+
+    size_t n = 0;
+    Track *track = mFirstTrack;
+    while (track) {
+        ++n;
+        track = track->next;
+    }
+
+    ALOGV("MPEG4Extractor::countTracks: %zu tracks", n);
+    return n;
+}
+
+status_t MPEG4Extractor::getTrackMetaData(
+        MetaDataBase &meta,
+        size_t index, uint32_t flags) {
+    status_t err;
+    if ((err = readMetaData()) != OK) {
+        return UNKNOWN_ERROR;
+    }
+
+    Track *track = mFirstTrack;
+    while (index > 0) {
+        if (track == NULL) {
+            return UNKNOWN_ERROR;
+        }
+
+        track = track->next;
+        --index;
+    }
+
+    if (track == NULL) {
+        return UNKNOWN_ERROR;
+    }
+
+    [=] {
+        int64_t duration;
+        int32_t samplerate;
+        if (track->has_elst && mHeaderTimescale != 0 &&
+                track->meta.findInt64(kKeyDuration, &duration) &&
+                track->meta.findInt32(kKeySampleRate, &samplerate)) {
+
+            track->has_elst = false;
+
+            if (track->elst_segment_duration > INT64_MAX) {
+                return;
+            }
+            int64_t segment_duration = track->elst_segment_duration;
+            int64_t media_time = track->elst_media_time;
+            int64_t halfscale = mHeaderTimescale / 2;
+            ALOGV("segment_duration = %" PRId64 ", media_time = %" PRId64
+                  ", halfscale = %" PRId64 ", timescale = %d",
+                  segment_duration,
+                  media_time,
+                  halfscale,
+                  mHeaderTimescale);
+
+            int64_t delay;
+            // delay = ((media_time * samplerate) + halfscale) / mHeaderTimescale;
+            if (__builtin_mul_overflow(media_time, samplerate, &delay) ||
+                    __builtin_add_overflow(delay, halfscale, &delay) ||
+                    (delay /= mHeaderTimescale, false) ||
+                    delay > INT32_MAX ||
+                    delay < INT32_MIN) {
+                return;
+            }
+            ALOGV("delay = %" PRId64, delay);
+            track->meta.setInt32(kKeyEncoderDelay, delay);
+
+            int64_t scaled_duration;
+            // scaled_duration = duration * mHeaderTimescale;
+            if (__builtin_mul_overflow(duration, mHeaderTimescale, &scaled_duration)) {
+                return;
+            }
+            ALOGV("scaled_duration = %" PRId64, scaled_duration);
+
+            int64_t segment_end;
+            int64_t padding;
+            // padding = scaled_duration - ((segment_duration + media_time) * 1000000);
+            if (__builtin_add_overflow(segment_duration, media_time, &segment_end) ||
+                    __builtin_mul_overflow(segment_end, 1000000, &segment_end) ||
+                    __builtin_sub_overflow(scaled_duration, segment_end, &padding)) {
+                return;
+            }
+            ALOGV("segment_end = %" PRId64 ", padding = %" PRId64, segment_end, padding);
+
+            if (padding < 0) {
+                // track duration from media header (which is what kKeyDuration is) might
+                // be slightly shorter than the segment duration, which would make the
+                // padding negative. Clamp to zero.
+                padding = 0;
+            }
+
+            int64_t paddingsamples;
+            int64_t halfscale_e6;
+            int64_t timescale_e6;
+            // paddingsamples = ((padding * samplerate) + (halfscale * 1000000))
+            //                / (mHeaderTimescale * 1000000);
+            if (__builtin_mul_overflow(padding, samplerate, &paddingsamples) ||
+                    __builtin_mul_overflow(halfscale, 1000000, &halfscale_e6) ||
+                    __builtin_mul_overflow(mHeaderTimescale, 1000000, &timescale_e6) ||
+                    __builtin_add_overflow(paddingsamples, halfscale_e6, &paddingsamples) ||
+                    (paddingsamples /= timescale_e6, false) ||
+                    paddingsamples > INT32_MAX) {
+                return;
+            }
+            ALOGV("paddingsamples = %" PRId64, paddingsamples);
+            track->meta.setInt32(kKeyEncoderPadding, paddingsamples);
+        }
+    }();
+
+    if ((flags & kIncludeExtensiveMetaData)
+            && !track->includes_expensive_metadata) {
+        track->includes_expensive_metadata = true;
+
+        const char *mime;
+        CHECK(track->meta.findCString(kKeyMIMEType, &mime));
+        if (!strncasecmp("video/", mime, 6)) {
+            // MPEG2 tracks do not provide CSD, so read the stream header
+            if (!strcmp(mime, MEDIA_MIMETYPE_VIDEO_MPEG2)) {
+                off64_t offset;
+                size_t size;
+                if (track->sampleTable->getMetaDataForSample(
+                            0 /* sampleIndex */, &offset, &size, NULL /* sampleTime */) == OK) {
+                    if (size > kMaxTrackHeaderSize) {
+                        size = kMaxTrackHeaderSize;
+                    }
+                    uint8_t header[kMaxTrackHeaderSize];
+                    if (mDataSource->readAt(offset, &header, size) == (ssize_t)size) {
+                        track->meta.setData(kKeyStreamHeader, 'mdat', header, size);
+                    }
+                }
+            }
+
+            if (mMoofOffset > 0) {
+                int64_t duration;
+                if (track->meta.findInt64(kKeyDuration, &duration)) {
+                    // nothing fancy, just pick a frame near 1/4th of the duration
+                    track->meta.setInt64(
+                            kKeyThumbnailTime, duration / 4);
+                }
+            } else {
+                uint32_t sampleIndex;
+                uint32_t sampleTime;
+                if (track->timescale != 0 &&
+                        track->sampleTable->findThumbnailSample(&sampleIndex) == OK
+                        && track->sampleTable->getMetaDataForSample(
+                            sampleIndex, NULL /* offset */, NULL /* size */,
+                            &sampleTime) == OK) {
+                    track->meta.setInt64(
+                            kKeyThumbnailTime,
+                            ((int64_t)sampleTime * 1000000) / track->timescale);
+                }
+            }
+        }
+    }
+
+    meta = track->meta;
+    return OK;
+}
+
+status_t MPEG4Extractor::readMetaData() {
+    if (mInitCheck != NO_INIT) {
+        return mInitCheck;
+    }
+
+    off64_t offset = 0;
+    status_t err;
+    bool sawMoovOrSidx = false;
+
+    while (!((mHasMoovBox && sawMoovOrSidx && (mMdatFound || mMoofFound)) ||
+             (mIsHeif && (mPreferHeif || !mHasMoovBox) &&
+                     (mItemTable != NULL) && mItemTable->isValid()))) {
+        off64_t orig_offset = offset;
+        err = parseChunk(&offset, 0);
+
+        if (err != OK && err != UNKNOWN_ERROR) {
+            break;
+        } else if (offset <= orig_offset) {
+            // only continue parsing if the offset was advanced,
+            // otherwise we might end up in an infinite loop
+            ALOGE("did not advance: %lld->%lld", (long long)orig_offset, (long long)offset);
+            err = ERROR_MALFORMED;
+            break;
+        } else if (err == UNKNOWN_ERROR) {
+            sawMoovOrSidx = true;
+        }
+    }
+
+    if (mIsHeif && (mItemTable != NULL) && (mItemTable->countImages() > 0)) {
+        off64_t exifOffset;
+        size_t exifSize;
+        if (mItemTable->getExifOffsetAndSize(&exifOffset, &exifSize) == OK) {
+            mFileMetaData.setInt64(kKeyExifOffset, (int64_t)exifOffset);
+            mFileMetaData.setInt64(kKeyExifSize, (int64_t)exifSize);
+        }
+        for (uint32_t imageIndex = 0;
+                imageIndex < mItemTable->countImages(); imageIndex++) {
+            sp<MetaData> meta = mItemTable->getImageMeta(imageIndex);
+            if (meta == NULL) {
+                ALOGE("heif image %u has no meta!", imageIndex);
+                continue;
+            }
+            // Some heif files advertise image sequence brands (eg. 'hevc') in
+            // ftyp box, but don't have any valid tracks in them. Instead of
+            // reporting the entire file as malformed, we override the error
+            // to allow still images to be extracted.
+            if (err != OK) {
+                ALOGW("Extracting still images only");
+                err = OK;
+            }
+            mInitCheck = OK;
+
+            ALOGV("adding HEIF image track %u", imageIndex);
+            Track *track = new Track;
+            track->next = NULL;
+            if (mLastTrack != NULL) {
+                mLastTrack->next = track;
+            } else {
+                mFirstTrack = track;
+            }
+            mLastTrack = track;
+
+            track->meta = *(meta.get());
+            track->meta.setInt32(kKeyTrackID, imageIndex);
+            track->includes_expensive_metadata = false;
+            track->skipTrack = false;
+            track->timescale = 1000000;
+        }
+    }
+
+    if (mInitCheck == OK) {
+        if (findTrackByMimePrefix("video/") != NULL) {
+            mFileMetaData.setCString(
+                    kKeyMIMEType, MEDIA_MIMETYPE_CONTAINER_MPEG4);
+        } else if (findTrackByMimePrefix("audio/") != NULL) {
+            mFileMetaData.setCString(kKeyMIMEType, "audio/mp4");
+        } else if (findTrackByMimePrefix(
+                MEDIA_MIMETYPE_IMAGE_ANDROID_HEIC) != NULL) {
+            mFileMetaData.setCString(
+                    kKeyMIMEType, MEDIA_MIMETYPE_CONTAINER_HEIF);
+        } else {
+            mFileMetaData.setCString(kKeyMIMEType, "application/octet-stream");
+        }
+    } else {
+        mInitCheck = err;
+    }
+
+    CHECK_NE(err, (status_t)NO_INIT);
+
+    // copy pssh data into file metadata
+    uint64_t psshsize = 0;
+    for (size_t i = 0; i < mPssh.size(); i++) {
+        psshsize += 20 + mPssh[i].datalen;
+    }
+    if (psshsize > 0 && psshsize <= UINT32_MAX) {
+        char *buf = (char*)malloc(psshsize);
+        if (!buf) {
+            ALOGE("b/28471206");
+            return NO_MEMORY;
+        }
+        char *ptr = buf;
+        for (size_t i = 0; i < mPssh.size(); i++) {
+            memcpy(ptr, mPssh[i].uuid, 20); // uuid + length
+            memcpy(ptr + 20, mPssh[i].data, mPssh[i].datalen);
+            ptr += (20 + mPssh[i].datalen);
+        }
+        mFileMetaData.setData(kKeyPssh, 'pssh', buf, psshsize);
+        free(buf);
+    }
+
+    return mInitCheck;
+}
+
+struct PathAdder {
+    PathAdder(Vector<uint32_t> *path, uint32_t chunkType)
+        : mPath(path) {
+        mPath->push(chunkType);
+    }
+
+    ~PathAdder() {
+        mPath->pop();
+    }
+
+private:
+    Vector<uint32_t> *mPath;
+
+    PathAdder(const PathAdder &);
+    PathAdder &operator=(const PathAdder &);
+};
+
+static bool underMetaDataPath(const Vector<uint32_t> &path) {
+    return path.size() >= 5
+        && path[0] == FOURCC('m', 'o', 'o', 'v')
+        && path[1] == FOURCC('u', 'd', 't', 'a')
+        && path[2] == FOURCC('m', 'e', 't', 'a')
+        && path[3] == FOURCC('i', 'l', 's', 't');
+}
+
+static bool underQTMetaPath(const Vector<uint32_t> &path, int32_t depth) {
+    return path.size() >= 2
+            && path[0] == FOURCC('m', 'o', 'o', 'v')
+            && path[1] == FOURCC('m', 'e', 't', 'a')
+            && (depth == 2
+            || (depth == 3
+                    && (path[2] == FOURCC('h', 'd', 'l', 'r')
+                    ||  path[2] == FOURCC('i', 'l', 's', 't')
+                    ||  path[2] == FOURCC('k', 'e', 'y', 's'))));
+}
+
+// Given a time in seconds since Jan 1 1904, produce a human-readable string.
+static bool convertTimeToDate(int64_t time_1904, String8 *s) {
+    // delta between mpeg4 time and unix epoch time
+    static const int64_t delta = (((66 * 365 + 17) * 24) * 3600);
+    if (time_1904 < INT64_MIN + delta) {
+        return false;
+    }
+    time_t time_1970 = time_1904 - delta;
+
+    char tmp[32];
+    struct tm* tm = gmtime(&time_1970);
+    if (tm != NULL &&
+            strftime(tmp, sizeof(tmp), "%Y%m%dT%H%M%S.000Z", tm) > 0) {
+        s->setTo(tmp);
+        return true;
+    }
+    return false;
+}
+
+status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) {
+    ALOGV("entering parseChunk %lld/%d", (long long)*offset, depth);
+
+    if (*offset < 0) {
+        ALOGE("b/23540914");
+        return ERROR_MALFORMED;
+    }
+    if (depth > 100) {
+        ALOGE("b/27456299");
+        return ERROR_MALFORMED;
+    }
+    uint32_t hdr[2];
+    if (mDataSource->readAt(*offset, hdr, 8) < 8) {
+        return ERROR_IO;
+    }
+    uint64_t chunk_size = ntohl(hdr[0]);
+    int32_t chunk_type = ntohl(hdr[1]);
+    off64_t data_offset = *offset + 8;
+
+    if (chunk_size == 1) {
+        if (mDataSource->readAt(*offset + 8, &chunk_size, 8) < 8) {
+            return ERROR_IO;
+        }
+        chunk_size = ntoh64(chunk_size);
+        data_offset += 8;
+
+        if (chunk_size < 16) {
+            // The smallest valid chunk is 16 bytes long in this case.
+            return ERROR_MALFORMED;
+        }
+    } else if (chunk_size == 0) {
+        if (depth == 0) {
+            // atom extends to end of file
+            off64_t sourceSize;
+            if (mDataSource->getSize(&sourceSize) == OK) {
+                chunk_size = (sourceSize - *offset);
+            } else {
+                // XXX could we just pick a "sufficiently large" value here?
+                ALOGE("atom size is 0, and data source has no size");
+                return ERROR_MALFORMED;
+            }
+        } else {
+            // not allowed for non-toplevel atoms, skip it
+            *offset += 4;
+            return OK;
+        }
+    } else if (chunk_size < 8) {
+        // The smallest valid chunk is 8 bytes long.
+        ALOGE("invalid chunk size: %" PRIu64, chunk_size);
+        return ERROR_MALFORMED;
+    }
+
+    char chunk[5];
+    MakeFourCCString(chunk_type, chunk);
+    ALOGV("chunk: %s @ %lld, %d", chunk, (long long)*offset, depth);
+
+    if (kUseHexDump) {
+        static const char kWhitespace[] = "                                        ";
+        const char *indent = &kWhitespace[sizeof(kWhitespace) - 1 - 2 * depth];
+        printf("%sfound chunk '%s' of size %" PRIu64 "\n", indent, chunk, chunk_size);
+
+        char buffer[256];
+        size_t n = chunk_size;
+        if (n > sizeof(buffer)) {
+            n = sizeof(buffer);
+        }
+        if (mDataSource->readAt(*offset, buffer, n)
+                < (ssize_t)n) {
+            return ERROR_IO;
+        }
+
+        hexdump(buffer, n);
+    }
+
+    PathAdder autoAdder(&mPath, chunk_type);
+
+    // (data_offset - *offset) is either 8 or 16
+    off64_t chunk_data_size = chunk_size - (data_offset - *offset);
+    if (chunk_data_size < 0) {
+        ALOGE("b/23540914");
+        return ERROR_MALFORMED;
+    }
+    if (chunk_type != FOURCC('m', 'd', 'a', 't') && chunk_data_size > kMaxAtomSize) {
+        char errMsg[100];
+        sprintf(errMsg, "%s atom has size %" PRId64, chunk, chunk_data_size);
+        ALOGE("%s (b/28615448)", errMsg);
+        android_errorWriteWithInfoLog(0x534e4554, "28615448", -1, errMsg, strlen(errMsg));
+        return ERROR_MALFORMED;
+    }
+
+    if (chunk_type != FOURCC('c', 'p', 'r', 't')
+            && chunk_type != FOURCC('c', 'o', 'v', 'r')
+            && mPath.size() == 5 && underMetaDataPath(mPath)) {
+        off64_t stop_offset = *offset + chunk_size;
+        *offset = data_offset;
+        while (*offset < stop_offset) {
+            status_t err = parseChunk(offset, depth + 1);
+            if (err != OK) {
+                return err;
+            }
+        }
+
+        if (*offset != stop_offset) {
+            return ERROR_MALFORMED;
+        }
+
+        return OK;
+    }
+
+    switch(chunk_type) {
+        case FOURCC('m', 'o', 'o', 'v'):
+        case FOURCC('t', 'r', 'a', 'k'):
+        case FOURCC('m', 'd', 'i', 'a'):
+        case FOURCC('m', 'i', 'n', 'f'):
+        case FOURCC('d', 'i', 'n', 'f'):
+        case FOURCC('s', 't', 'b', 'l'):
+        case FOURCC('m', 'v', 'e', 'x'):
+        case FOURCC('m', 'o', 'o', 'f'):
+        case FOURCC('t', 'r', 'a', 'f'):
+        case FOURCC('m', 'f', 'r', 'a'):
+        case FOURCC('u', 'd', 't', 'a'):
+        case FOURCC('i', 'l', 's', 't'):
+        case FOURCC('s', 'i', 'n', 'f'):
+        case FOURCC('s', 'c', 'h', 'i'):
+        case FOURCC('e', 'd', 't', 's'):
+        case FOURCC('w', 'a', 'v', 'e'):
+        {
+            if (chunk_type == FOURCC('m', 'o', 'o', 'v') && depth != 0) {
+                ALOGE("moov: depth %d", depth);
+                return ERROR_MALFORMED;
+            }
+
+            if (chunk_type == FOURCC('m', 'o', 'o', 'v') && mInitCheck == OK) {
+                ALOGE("duplicate moov");
+                return ERROR_MALFORMED;
+            }
+
+            if (chunk_type == FOURCC('m', 'o', 'o', 'f') && !mMoofFound) {
+                // store the offset of the first segment
+                mMoofFound = true;
+                mMoofOffset = *offset;
+            }
+
+            if (chunk_type == FOURCC('s', 't', 'b', 'l')) {
+                ALOGV("sampleTable chunk is %" PRIu64 " bytes long.", chunk_size);
+
+                if (mDataSource->flags()
+                        & (DataSourceBase::kWantsPrefetching
+                            | DataSourceBase::kIsCachingDataSource)) {
+                    CachedRangedDataSource *cachedSource =
+                        new CachedRangedDataSource(mDataSource);
+
+                    if (cachedSource->setCachedRange(
+                            *offset, chunk_size,
+                            mCachedSource != NULL /* assume ownership on success */) == OK) {
+                        mDataSource = mCachedSource = cachedSource;
+                    } else {
+                        delete cachedSource;
+                    }
+                }
+
+                if (mLastTrack == NULL) {
+                    return ERROR_MALFORMED;
+                }
+
+                mLastTrack->sampleTable = new SampleTable(mDataSource);
+            }
+
+            bool isTrack = false;
+            if (chunk_type == FOURCC('t', 'r', 'a', 'k')) {
+                if (depth != 1) {
+                    ALOGE("trak: depth %d", depth);
+                    return ERROR_MALFORMED;
+                }
+                isTrack = true;
+
+                ALOGV("adding new track");
+                Track *track = new Track;
+                track->next = NULL;
+                if (mLastTrack) {
+                    mLastTrack->next = track;
+                } else {
+                    mFirstTrack = track;
+                }
+                mLastTrack = track;
+
+                track->includes_expensive_metadata = false;
+                track->skipTrack = false;
+                track->timescale = 0;
+                track->meta.setCString(kKeyMIMEType, "application/octet-stream");
+                track->has_elst = false;
+                track->subsample_encryption = false;
+            }
+
+            off64_t stop_offset = *offset + chunk_size;
+            *offset = data_offset;
+            while (*offset < stop_offset) {
+                status_t err = parseChunk(offset, depth + 1);
+                if (err != OK) {
+                    if (isTrack) {
+                        mLastTrack->skipTrack = true;
+                        break;
+                    }
+                    return err;
+                }
+            }
+
+            if (*offset != stop_offset) {
+                return ERROR_MALFORMED;
+            }
+
+            if (isTrack) {
+                int32_t trackId;
+                // There must be exact one track header per track.
+                if (!mLastTrack->meta.findInt32(kKeyTrackID, &trackId)) {
+                    mLastTrack->skipTrack = true;
+                }
+
+                status_t err = verifyTrack(mLastTrack);
+                if (err != OK) {
+                    mLastTrack->skipTrack = true;
+                }
+
+                if (mLastTrack->skipTrack) {
+                    ALOGV("skipping this track...");
+                    Track *cur = mFirstTrack;
+
+                    if (cur == mLastTrack) {
+                        delete cur;
+                        mFirstTrack = mLastTrack = NULL;
+                    } else {
+                        while (cur && cur->next != mLastTrack) {
+                            cur = cur->next;
+                        }
+                        if (cur) {
+                            cur->next = NULL;
+                        }
+                        delete mLastTrack;
+                        mLastTrack = cur;
+                    }
+
+                    return OK;
+                }
+            } else if (chunk_type == FOURCC('m', 'o', 'o', 'v')) {
+                mInitCheck = OK;
+
+                return UNKNOWN_ERROR;  // Return a dummy error.
+            }
+            break;
+        }
+
+        case FOURCC('s', 'c', 'h', 'm'):
+        {
+
+            *offset += chunk_size;
+            if (!mLastTrack) {
+                return ERROR_MALFORMED;
+            }
+
+            uint32_t scheme_type;
+            if (mDataSource->readAt(data_offset + 4, &scheme_type, 4) < 4) {
+                return ERROR_IO;
+            }
+            scheme_type = ntohl(scheme_type);
+            int32_t mode = kCryptoModeUnencrypted;
+            switch(scheme_type) {
+                case FOURCC('c', 'b', 'c', '1'):
+                {
+                    mode = kCryptoModeAesCbc;
+                    break;
+                }
+                case FOURCC('c', 'b', 'c', 's'):
+                {
+                    mode = kCryptoModeAesCbc;
+                    mLastTrack->subsample_encryption = true;
+                    break;
+                }
+                case FOURCC('c', 'e', 'n', 'c'):
+                {
+                    mode = kCryptoModeAesCtr;
+                    break;
+                }
+                case FOURCC('c', 'e', 'n', 's'):
+                {
+                    mode = kCryptoModeAesCtr;
+                    mLastTrack->subsample_encryption = true;
+                    break;
+                }
+            }
+            if (mode != kCryptoModeUnencrypted) {
+                mLastTrack->meta.setInt32(kKeyCryptoMode, mode);
+            }
+            break;
+        }
+
+
+        case FOURCC('e', 'l', 's', 't'):
+        {
+            *offset += chunk_size;
+
+            if (!mLastTrack) {
+                return ERROR_MALFORMED;
+            }
+
+            // See 14496-12 8.6.6
+            uint8_t version;
+            if (mDataSource->readAt(data_offset, &version, 1) < 1) {
+                return ERROR_IO;
+            }
+
+            uint32_t entry_count;
+            if (!mDataSource->getUInt32(data_offset + 4, &entry_count)) {
+                return ERROR_IO;
+            }
+
+            if (entry_count != 1) {
+                // we only support a single entry at the moment, for gapless playback
+                ALOGW("ignoring edit list with %d entries", entry_count);
+            } else {
+                off64_t entriesoffset = data_offset + 8;
+                uint64_t segment_duration;
+                int64_t media_time;
+
+                if (version == 1) {
+                    if (!mDataSource->getUInt64(entriesoffset, &segment_duration) ||
+                            !mDataSource->getUInt64(entriesoffset + 8, (uint64_t*)&media_time)) {
+                        return ERROR_IO;
+                    }
+                } else if (version == 0) {
+                    uint32_t sd;
+                    int32_t mt;
+                    if (!mDataSource->getUInt32(entriesoffset, &sd) ||
+                            !mDataSource->getUInt32(entriesoffset + 4, (uint32_t*)&mt)) {
+                        return ERROR_IO;
+                    }
+                    segment_duration = sd;
+                    media_time = mt;
+                } else {
+                    return ERROR_IO;
+                }
+
+                // save these for later, because the elst atom might precede
+                // the atoms that actually gives us the duration and sample rate
+                // needed to calculate the padding and delay values
+                mLastTrack->has_elst = true;
+                mLastTrack->elst_media_time = media_time;
+                mLastTrack->elst_segment_duration = segment_duration;
+            }
+            break;
+        }
+
+        case FOURCC('f', 'r', 'm', 'a'):
+        {
+            *offset += chunk_size;
+
+            uint32_t original_fourcc;
+            if (mDataSource->readAt(data_offset, &original_fourcc, 4) < 4) {
+                return ERROR_IO;
+            }
+            original_fourcc = ntohl(original_fourcc);
+            ALOGV("read original format: %d", original_fourcc);
+
+            if (mLastTrack == NULL) {
+                return ERROR_MALFORMED;
+            }
+
+            mLastTrack->meta.setCString(kKeyMIMEType, FourCC2MIME(original_fourcc));
+            uint32_t num_channels = 0;
+            uint32_t sample_rate = 0;
+            if (AdjustChannelsAndRate(original_fourcc, &num_channels, &sample_rate)) {
+                mLastTrack->meta.setInt32(kKeyChannelCount, num_channels);
+                mLastTrack->meta.setInt32(kKeySampleRate, sample_rate);
+            }
+            break;
+        }
+
+        case FOURCC('t', 'e', 'n', 'c'):
+        {
+            *offset += chunk_size;
+
+            if (chunk_size < 32) {
+                return ERROR_MALFORMED;
+            }
+
+            // tenc box contains 1 byte version, 3 byte flags, 3 byte default algorithm id, one byte
+            // default IV size, 16 bytes default KeyID
+            // (ISO 23001-7)
+
+            uint8_t version;
+            if (mDataSource->readAt(data_offset, &version, sizeof(version))
+                    < (ssize_t)sizeof(version)) {
+                return ERROR_IO;
+            }
+
+            uint8_t buf[4];
+            memset(buf, 0, 4);
+            if (mDataSource->readAt(data_offset + 4, buf + 1, 3) < 3) {
+                return ERROR_IO;
+            }
+
+            if (mLastTrack == NULL) {
+                return ERROR_MALFORMED;
+            }
+
+            uint8_t defaultEncryptedByteBlock = 0;
+            uint8_t defaultSkipByteBlock = 0;
+            uint32_t defaultAlgorithmId = ntohl(*((int32_t*)buf));
+            if (version == 1) {
+                uint32_t pattern = buf[2];
+                defaultEncryptedByteBlock = pattern >> 4;
+                defaultSkipByteBlock = pattern & 0xf;
+                if (defaultEncryptedByteBlock == 0 && defaultSkipByteBlock == 0) {
+                    // use (1,0) to mean "encrypt everything"
+                    defaultEncryptedByteBlock = 1;
+                }
+            } else if (mLastTrack->subsample_encryption) {
+                ALOGW("subsample_encryption should be version 1");
+            } else if (defaultAlgorithmId > 1) {
+                // only 0 (clear) and 1 (AES-128) are valid
+                ALOGW("defaultAlgorithmId: %u is a reserved value", defaultAlgorithmId);
+                defaultAlgorithmId = 1;
+            }
+
+            memset(buf, 0, 4);
+            if (mDataSource->readAt(data_offset + 7, buf + 3, 1) < 1) {
+                return ERROR_IO;
+            }
+            uint32_t defaultIVSize = ntohl(*((int32_t*)buf));
+
+            if (defaultAlgorithmId == 0 && defaultIVSize != 0) {
+                // only unencrypted data must have 0 IV size
+                return ERROR_MALFORMED;
+            } else if (defaultIVSize != 0 &&
+                    defaultIVSize != 8 &&
+                    defaultIVSize != 16) {
+                return ERROR_MALFORMED;
+            }
+
+            uint8_t defaultKeyId[16];
+
+            if (mDataSource->readAt(data_offset + 8, &defaultKeyId, 16) < 16) {
+                return ERROR_IO;
+            }
+
+            sp<ABuffer> defaultConstantIv;
+            if (defaultAlgorithmId != 0 && defaultIVSize == 0) {
+
+                uint8_t ivlength;
+                if (mDataSource->readAt(data_offset + 24, &ivlength, sizeof(ivlength))
+                        < (ssize_t)sizeof(ivlength)) {
+                    return ERROR_IO;
+                }
+
+                if (ivlength != 8 && ivlength != 16) {
+                    ALOGW("unsupported IV length: %u", ivlength);
+                    return ERROR_MALFORMED;
+                }
+
+                defaultConstantIv = new ABuffer(ivlength);
+                if (mDataSource->readAt(data_offset + 25, defaultConstantIv->data(), ivlength)
+                        < (ssize_t)ivlength) {
+                    return ERROR_IO;
+                }
+
+                defaultConstantIv->setRange(0, ivlength);
+            }
+
+            int32_t tmpAlgorithmId;
+            if (!mLastTrack->meta.findInt32(kKeyCryptoMode, &tmpAlgorithmId)) {
+                mLastTrack->meta.setInt32(kKeyCryptoMode, defaultAlgorithmId);
+            }
+
+            mLastTrack->meta.setInt32(kKeyCryptoDefaultIVSize, defaultIVSize);
+            mLastTrack->meta.setData(kKeyCryptoKey, 'tenc', defaultKeyId, 16);
+            mLastTrack->meta.setInt32(kKeyEncryptedByteBlock, defaultEncryptedByteBlock);
+            mLastTrack->meta.setInt32(kKeySkipByteBlock, defaultSkipByteBlock);
+            if (defaultConstantIv != NULL) {
+                mLastTrack->meta.setData(kKeyCryptoIV, 'dciv', defaultConstantIv->data(), defaultConstantIv->size());
+            }
+            break;
+        }
+
+        case FOURCC('t', 'k', 'h', 'd'):
+        {
+            *offset += chunk_size;
+
+            status_t err;
+            if ((err = parseTrackHeader(data_offset, chunk_data_size)) != OK) {
+                return err;
+            }
+
+            break;
+        }
+
+        case FOURCC('t', 'r', 'e', 'f'):
+        {
+            off64_t stop_offset = *offset + chunk_size;
+            *offset = data_offset;
+            while (*offset < stop_offset) {
+                status_t err = parseChunk(offset, depth + 1);
+                if (err != OK) {
+                    return err;
+                }
+            }
+            if (*offset != stop_offset) {
+                return ERROR_MALFORMED;
+            }
+            break;
+        }
+
+        case FOURCC('t', 'h', 'm', 'b'):
+        {
+            *offset += chunk_size;
+
+            if (mLastTrack != NULL) {
+                // Skip thumbnail track for now since we don't have an
+                // API to retrieve it yet.
+                // The thumbnail track can't be accessed by negative index or time,
+                // because each timed sample has its own corresponding thumbnail
+                // in the thumbnail track. We'll need a dedicated API to retrieve
+                // thumbnail at time instead.
+                mLastTrack->skipTrack = true;
+            }
+
+            break;
+        }
+
+        case FOURCC('p', 's', 's', 'h'):
+        {
+            *offset += chunk_size;
+
+            PsshInfo pssh;
+
+            if (mDataSource->readAt(data_offset + 4, &pssh.uuid, 16) < 16) {
+                return ERROR_IO;
+            }
+
+            uint32_t psshdatalen = 0;
+            if (mDataSource->readAt(data_offset + 20, &psshdatalen, 4) < 4) {
+                return ERROR_IO;
+            }
+            pssh.datalen = ntohl(psshdatalen);
+            ALOGV("pssh data size: %d", pssh.datalen);
+            if (chunk_size < 20 || pssh.datalen > chunk_size - 20) {
+                // pssh data length exceeds size of containing box
+                return ERROR_MALFORMED;
+            }
+
+            pssh.data = new (std::nothrow) uint8_t[pssh.datalen];
+            if (pssh.data == NULL) {
+                return ERROR_MALFORMED;
+            }
+            ALOGV("allocated pssh @ %p", pssh.data);
+            ssize_t requested = (ssize_t) pssh.datalen;
+            if (mDataSource->readAt(data_offset + 24, pssh.data, requested) < requested) {
+                delete[] pssh.data;
+                return ERROR_IO;
+            }
+            mPssh.push_back(pssh);
+
+            break;
+        }
+
+        case FOURCC('m', 'd', 'h', 'd'):
+        {
+            *offset += chunk_size;
+
+            if (chunk_data_size < 4 || mLastTrack == NULL) {
+                return ERROR_MALFORMED;
+            }
+
+            uint8_t version;
+            if (mDataSource->readAt(
+                        data_offset, &version, sizeof(version))
+                    < (ssize_t)sizeof(version)) {
+                return ERROR_IO;
+            }
+
+            off64_t timescale_offset;
+
+            if (version == 1) {
+                timescale_offset = data_offset + 4 + 16;
+            } else if (version == 0) {
+                timescale_offset = data_offset + 4 + 8;
+            } else {
+                return ERROR_IO;
+            }
+
+            uint32_t timescale;
+            if (mDataSource->readAt(
+                        timescale_offset, &timescale, sizeof(timescale))
+                    < (ssize_t)sizeof(timescale)) {
+                return ERROR_IO;
+            }
+
+            if (!timescale) {
+                ALOGE("timescale should not be ZERO.");
+                return ERROR_MALFORMED;
+            }
+
+            mLastTrack->timescale = ntohl(timescale);
+
+            // 14496-12 says all ones means indeterminate, but some files seem to use
+            // 0 instead. We treat both the same.
+            int64_t duration = 0;
+            if (version == 1) {
+                if (mDataSource->readAt(
+                            timescale_offset + 4, &duration, sizeof(duration))
+                        < (ssize_t)sizeof(duration)) {
+                    return ERROR_IO;
+                }
+                if (duration != -1) {
+                    duration = ntoh64(duration);
+                }
+            } else {
+                uint32_t duration32;
+                if (mDataSource->readAt(
+                            timescale_offset + 4, &duration32, sizeof(duration32))
+                        < (ssize_t)sizeof(duration32)) {
+                    return ERROR_IO;
+                }
+                if (duration32 != 0xffffffff) {
+                    duration = ntohl(duration32);
+                }
+            }
+            if (duration != 0 && mLastTrack->timescale != 0) {
+                mLastTrack->meta.setInt64(
+                        kKeyDuration, (duration * 1000000) / mLastTrack->timescale);
+            }
+
+            uint8_t lang[2];
+            off64_t lang_offset;
+            if (version == 1) {
+                lang_offset = timescale_offset + 4 + 8;
+            } else if (version == 0) {
+                lang_offset = timescale_offset + 4 + 4;
+            } else {
+                return ERROR_IO;
+            }
+
+            if (mDataSource->readAt(lang_offset, &lang, sizeof(lang))
+                    < (ssize_t)sizeof(lang)) {
+                return ERROR_IO;
+            }
+
+            // To get the ISO-639-2/T three character language code
+            // 1 bit pad followed by 3 5-bits characters. Each character
+            // is packed as the difference between its ASCII value and 0x60.
+            char lang_code[4];
+            lang_code[0] = ((lang[0] >> 2) & 0x1f) + 0x60;
+            lang_code[1] = ((lang[0] & 0x3) << 3 | (lang[1] >> 5)) + 0x60;
+            lang_code[2] = (lang[1] & 0x1f) + 0x60;
+            lang_code[3] = '\0';
+
+            mLastTrack->meta.setCString(
+                    kKeyMediaLanguage, lang_code);
+
+            break;
+        }
+
+        case FOURCC('s', 't', 's', 'd'):
+        {
+            uint8_t buffer[8];
+            if (chunk_data_size < (off64_t)sizeof(buffer)) {
+                return ERROR_MALFORMED;
+            }
+
+            if (mDataSource->readAt(
+                        data_offset, buffer, 8) < 8) {
+                return ERROR_IO;
+            }
+
+            if (U32_AT(buffer) != 0) {
+                // Should be version 0, flags 0.
+                return ERROR_MALFORMED;
+            }
+
+            uint32_t entry_count = U32_AT(&buffer[4]);
+
+            if (entry_count > 1) {
+                // For 3GPP timed text, there could be multiple tx3g boxes contain
+                // multiple text display formats. These formats will be used to
+                // display the timed text.
+                // For encrypted files, there may also be more than one entry.
+                const char *mime;
+
+                if (mLastTrack == NULL)
+                    return ERROR_MALFORMED;
+
+                CHECK(mLastTrack->meta.findCString(kKeyMIMEType, &mime));
+                if (strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP) &&
+                        strcasecmp(mime, "application/octet-stream")) {
+                    // For now we only support a single type of media per track.
+                    mLastTrack->skipTrack = true;
+                    *offset += chunk_size;
+                    break;
+                }
+            }
+            off64_t stop_offset = *offset + chunk_size;
+            *offset = data_offset + 8;
+            for (uint32_t i = 0; i < entry_count; ++i) {
+                status_t err = parseChunk(offset, depth + 1);
+                if (err != OK) {
+                    return err;
+                }
+            }
+
+            if (*offset != stop_offset) {
+                return ERROR_MALFORMED;
+            }
+            break;
+        }
+        case FOURCC('m', 'e', 't', 't'):
+        {
+            *offset += chunk_size;
+
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
+            auto buffer = heapbuffer<uint8_t>(chunk_data_size);
+            if (buffer.get() == NULL) {
+                return NO_MEMORY;
+            }
+
+            if (mDataSource->readAt(
+                        data_offset, buffer.get(), chunk_data_size) < chunk_data_size) {
+                return ERROR_IO;
+            }
+
+            String8 mimeFormat((const char *)(buffer.get()), chunk_data_size);
+            mLastTrack->meta.setCString(kKeyMIMEType, mimeFormat.string());
+
+            break;
+        }
+
+        case FOURCC('m', 'p', '4', 'a'):
+        case FOURCC('e', 'n', 'c', 'a'):
+        case FOURCC('s', 'a', 'm', 'r'):
+        case FOURCC('s', 'a', 'w', 'b'):
+        {
+            if (mIsQT && chunk_type == FOURCC('m', 'p', '4', 'a')
+                    && depth >= 1 && mPath[depth - 1] == FOURCC('w', 'a', 'v', 'e')) {
+                // Ignore mp4a embedded in QT wave atom
+                *offset += chunk_size;
+                break;
+            }
+
+            uint8_t buffer[8 + 20];
+            if (chunk_data_size < (ssize_t)sizeof(buffer)) {
+                // Basic AudioSampleEntry size.
+                return ERROR_MALFORMED;
+            }
+
+            if (mDataSource->readAt(
+                        data_offset, buffer, sizeof(buffer)) < (ssize_t)sizeof(buffer)) {
+                return ERROR_IO;
+            }
+
+            uint16_t data_ref_index __unused = U16_AT(&buffer[6]);
+            uint16_t version = U16_AT(&buffer[8]);
+            uint32_t num_channels = U16_AT(&buffer[16]);
+
+            uint16_t sample_size = U16_AT(&buffer[18]);
+            uint32_t sample_rate = U32_AT(&buffer[24]) >> 16;
+
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
+            off64_t stop_offset = *offset + chunk_size;
+            *offset = data_offset + sizeof(buffer);
+
+            if (mIsQT && chunk_type == FOURCC('m', 'p', '4', 'a')) {
+                if (version == 1) {
+                    if (mDataSource->readAt(*offset, buffer, 16) < 16) {
+                        return ERROR_IO;
+                    }
+
+#if 0
+                    U32_AT(buffer);  // samples per packet
+                    U32_AT(&buffer[4]);  // bytes per packet
+                    U32_AT(&buffer[8]);  // bytes per frame
+                    U32_AT(&buffer[12]);  // bytes per sample
+#endif
+                    *offset += 16;
+                } else if (version == 2) {
+                    uint8_t v2buffer[36];
+                    if (mDataSource->readAt(*offset, v2buffer, 36) < 36) {
+                        return ERROR_IO;
+                    }
+
+#if 0
+                    U32_AT(v2buffer);  // size of struct only
+                    sample_rate = (uint32_t)U64_AT(&v2buffer[4]);  // audio sample rate
+                    num_channels = U32_AT(&v2buffer[12]);  // num audio channels
+                    U32_AT(&v2buffer[16]);  // always 0x7f000000
+                    sample_size = (uint16_t)U32_AT(&v2buffer[20]);  // const bits per channel
+                    U32_AT(&v2buffer[24]);  // format specifc flags
+                    U32_AT(&v2buffer[28]);  // const bytes per audio packet
+                    U32_AT(&v2buffer[32]);  // const LPCM frames per audio packet
+#endif
+                    *offset += 36;
+                }
+            }
+
+            if (chunk_type != FOURCC('e', 'n', 'c', 'a')) {
+                // if the chunk type is enca, we'll get the type from the frma box later
+                mLastTrack->meta.setCString(kKeyMIMEType, FourCC2MIME(chunk_type));
+                AdjustChannelsAndRate(chunk_type, &num_channels, &sample_rate);
+            }
+            ALOGV("*** coding='%s' %d channels, size %d, rate %d\n",
+                   chunk, num_channels, sample_size, sample_rate);
+            mLastTrack->meta.setInt32(kKeyChannelCount, num_channels);
+            mLastTrack->meta.setInt32(kKeySampleRate, sample_rate);
+
+            while (*offset < stop_offset) {
+                status_t err = parseChunk(offset, depth + 1);
+                if (err != OK) {
+                    return err;
+                }
+            }
+
+            if (*offset != stop_offset) {
+                return ERROR_MALFORMED;
+            }
+            break;
+        }
+
+        case FOURCC('m', 'p', '4', 'v'):
+        case FOURCC('e', 'n', 'c', 'v'):
+        case FOURCC('s', '2', '6', '3'):
+        case FOURCC('H', '2', '6', '3'):
+        case FOURCC('h', '2', '6', '3'):
+        case FOURCC('a', 'v', 'c', '1'):
+        case FOURCC('h', 'v', 'c', '1'):
+        case FOURCC('h', 'e', 'v', '1'):
+        {
+            uint8_t buffer[78];
+            if (chunk_data_size < (ssize_t)sizeof(buffer)) {
+                // Basic VideoSampleEntry size.
+                return ERROR_MALFORMED;
+            }
+
+            if (mDataSource->readAt(
+                        data_offset, buffer, sizeof(buffer)) < (ssize_t)sizeof(buffer)) {
+                return ERROR_IO;
+            }
+
+            uint16_t data_ref_index __unused = U16_AT(&buffer[6]);
+            uint16_t width = U16_AT(&buffer[6 + 18]);
+            uint16_t height = U16_AT(&buffer[6 + 20]);
+
+            // The video sample is not standard-compliant if it has invalid dimension.
+            // Use some default width and height value, and
+            // let the decoder figure out the actual width and height (and thus
+            // be prepared for INFO_FOMRAT_CHANGED event).
+            if (width == 0)  width  = 352;
+            if (height == 0) height = 288;
+
+            // printf("*** coding='%s' width=%d height=%d\n",
+            //        chunk, width, height);
+
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
+            if (chunk_type != FOURCC('e', 'n', 'c', 'v')) {
+                // if the chunk type is encv, we'll get the type from the frma box later
+                mLastTrack->meta.setCString(kKeyMIMEType, FourCC2MIME(chunk_type));
+            }
+            mLastTrack->meta.setInt32(kKeyWidth, width);
+            mLastTrack->meta.setInt32(kKeyHeight, height);
+
+            off64_t stop_offset = *offset + chunk_size;
+            *offset = data_offset + sizeof(buffer);
+            while (*offset < stop_offset) {
+                status_t err = parseChunk(offset, depth + 1);
+                if (err != OK) {
+                    return err;
+                }
+            }
+
+            if (*offset != stop_offset) {
+                return ERROR_MALFORMED;
+            }
+            break;
+        }
+
+        case FOURCC('s', 't', 'c', 'o'):
+        case FOURCC('c', 'o', '6', '4'):
+        {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL)) {
+                return ERROR_MALFORMED;
+            }
+
+            status_t err =
+                mLastTrack->sampleTable->setChunkOffsetParams(
+                        chunk_type, data_offset, chunk_data_size);
+
+            *offset += chunk_size;
+
+            if (err != OK) {
+                return err;
+            }
+
+            break;
+        }
+
+        case FOURCC('s', 't', 's', 'c'):
+        {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
+                return ERROR_MALFORMED;
+
+            status_t err =
+                mLastTrack->sampleTable->setSampleToChunkParams(
+                        data_offset, chunk_data_size);
+
+            *offset += chunk_size;
+
+            if (err != OK) {
+                return err;
+            }
+
+            break;
+        }
+
+        case FOURCC('s', 't', 's', 'z'):
+        case FOURCC('s', 't', 'z', '2'):
+        {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL)) {
+                return ERROR_MALFORMED;
+            }
+
+            status_t err =
+                mLastTrack->sampleTable->setSampleSizeParams(
+                        chunk_type, data_offset, chunk_data_size);
+
+            *offset += chunk_size;
+
+            if (err != OK) {
+                return err;
+            }
+
+            size_t max_size;
+            err = mLastTrack->sampleTable->getMaxSampleSize(&max_size);
+
+            if (err != OK) {
+                return err;
+            }
+
+            if (max_size != 0) {
+                // Assume that a given buffer only contains at most 10 chunks,
+                // each chunk originally prefixed with a 2 byte length will
+                // have a 4 byte header (0x00 0x00 0x00 0x01) after conversion,
+                // and thus will grow by 2 bytes per chunk.
+                if (max_size > SIZE_MAX - 10 * 2) {
+                    ALOGE("max sample size too big: %zu", max_size);
+                    return ERROR_MALFORMED;
+                }
+                mLastTrack->meta.setInt32(kKeyMaxInputSize, max_size + 10 * 2);
+            } else {
+                // No size was specified. Pick a conservatively large size.
+                uint32_t width, height;
+                if (!mLastTrack->meta.findInt32(kKeyWidth, (int32_t*)&width) ||
+                    !mLastTrack->meta.findInt32(kKeyHeight,(int32_t*) &height)) {
+                    ALOGE("No width or height, assuming worst case 1080p");
+                    width = 1920;
+                    height = 1080;
+                } else {
+                    // A resolution was specified, check that it's not too big. The values below
+                    // were chosen so that the calculations below don't cause overflows, they're
+                    // not indicating that resolutions up to 32kx32k are actually supported.
+                    if (width > 32768 || height > 32768) {
+                        ALOGE("can't support %u x %u video", width, height);
+                        return ERROR_MALFORMED;
+                    }
+                }
+
+                const char *mime;
+                CHECK(mLastTrack->meta.findCString(kKeyMIMEType, &mime));
+                if (!strncmp(mime, "audio/", 6)) {
+                    // for audio, use 128KB
+                    max_size = 1024 * 128;
+                } else if (!strcmp(mime, MEDIA_MIMETYPE_VIDEO_AVC)
+                        || !strcmp(mime, MEDIA_MIMETYPE_VIDEO_HEVC)) {
+                    // AVC & HEVC requires compression ratio of at least 2, and uses
+                    // macroblocks
+                    max_size = ((width + 15) / 16) * ((height + 15) / 16) * 192;
+                } else {
+                    // For all other formats there is no minimum compression
+                    // ratio. Use compression ratio of 1.
+                    max_size = width * height * 3 / 2;
+                }
+                // HACK: allow 10% overhead
+                // TODO: read sample size from traf atom for fragmented MPEG4.
+                max_size += max_size / 10;
+                mLastTrack->meta.setInt32(kKeyMaxInputSize, max_size);
+            }
+
+            // NOTE: setting another piece of metadata invalidates any pointers (such as the
+            // mimetype) previously obtained, so don't cache them.
+            const char *mime;
+            CHECK(mLastTrack->meta.findCString(kKeyMIMEType, &mime));
+            // Calculate average frame rate.
+            if (!strncasecmp("video/", mime, 6)) {
+                size_t nSamples = mLastTrack->sampleTable->countSamples();
+                if (nSamples == 0) {
+                    int32_t trackId;
+                    if (mLastTrack->meta.findInt32(kKeyTrackID, &trackId)) {
+                        for (size_t i = 0; i < mTrex.size(); i++) {
+                            Trex *t = &mTrex.editItemAt(i);
+                            if (t->track_ID == (uint32_t) trackId) {
+                                if (t->default_sample_duration > 0) {
+                                    int32_t frameRate =
+                                            mLastTrack->timescale / t->default_sample_duration;
+                                    mLastTrack->meta.setInt32(kKeyFrameRate, frameRate);
+                                }
+                                break;
+                            }
+                        }
+                    }
+                } else {
+                    int64_t durationUs;
+                    if (mLastTrack->meta.findInt64(kKeyDuration, &durationUs)) {
+                        if (durationUs > 0) {
+                            int32_t frameRate = (nSamples * 1000000LL +
+                                        (durationUs >> 1)) / durationUs;
+                            mLastTrack->meta.setInt32(kKeyFrameRate, frameRate);
+                        }
+                    }
+                    ALOGV("setting frame count %zu", nSamples);
+                    mLastTrack->meta.setInt32(kKeyFrameCount, nSamples);
+                }
+            }
+
+            break;
+        }
+
+        case FOURCC('s', 't', 't', 's'):
+        {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
+                return ERROR_MALFORMED;
+
+            *offset += chunk_size;
+
+            status_t err =
+                mLastTrack->sampleTable->setTimeToSampleParams(
+                        data_offset, chunk_data_size);
+
+            if (err != OK) {
+                return err;
+            }
+
+            break;
+        }
+
+        case FOURCC('c', 't', 't', 's'):
+        {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
+                return ERROR_MALFORMED;
+
+            *offset += chunk_size;
+
+            status_t err =
+                mLastTrack->sampleTable->setCompositionTimeToSampleParams(
+                        data_offset, chunk_data_size);
+
+            if (err != OK) {
+                return err;
+            }
+
+            break;
+        }
+
+        case FOURCC('s', 't', 's', 's'):
+        {
+            if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
+                return ERROR_MALFORMED;
+
+            *offset += chunk_size;
+
+            status_t err =
+                mLastTrack->sampleTable->setSyncSampleParams(
+                        data_offset, chunk_data_size);
+
+            if (err != OK) {
+                return err;
+            }
+
+            break;
+        }
+
+        // \xA9xyz
+        case FOURCC(0xA9, 'x', 'y', 'z'):
+        {
+            *offset += chunk_size;
+
+            // Best case the total data length inside "\xA9xyz" box would
+            // be 9, for instance "\xA9xyz" + "\x00\x05\x15\xc7" + "+0+0/",
+            // where "\x00\x05" is the text string length with value = 5,
+            // "\0x15\xc7" is the language code = en, and "+0+0/" is a
+            // location (string) value with longitude = 0 and latitude = 0.
+            // Since some devices encountered in the wild omit the trailing
+            // slash, we'll allow that.
+            if (chunk_data_size < 8) { // 8 instead of 9 to allow for missing /
+                return ERROR_MALFORMED;
+            }
+
+            uint16_t len;
+            if (!mDataSource->getUInt16(data_offset, &len)) {
+                return ERROR_IO;
+            }
+
+            // allow "+0+0" without trailing slash
+            if (len < 4 || len > chunk_data_size - 4) {
+                return ERROR_MALFORMED;
+            }
+            // The location string following the language code is formatted
+            // according to ISO 6709:2008 (https://en.wikipedia.org/wiki/ISO_6709).
+            // Allocate 2 extra bytes, in case we need to add a trailing slash,
+            // and to add a terminating 0.
+            std::unique_ptr<char[]> buffer(new (std::nothrow) char[len+2]());
+            if (!buffer) {
+                return NO_MEMORY;
+            }
+
+            if (mDataSource->readAt(
+                        data_offset + 4, &buffer[0], len) < len) {
+                return ERROR_IO;
+            }
+
+            len = strlen(&buffer[0]);
+            if (len < 4) {
+                return ERROR_MALFORMED;
+            }
+            // Add a trailing slash if there wasn't one.
+            if (buffer[len - 1] != '/') {
+                buffer[len] = '/';
+            }
+            mFileMetaData.setCString(kKeyLocation, &buffer[0]);
+            break;
+        }
+
+        case FOURCC('e', 's', 'd', 's'):
+        {
+            *offset += chunk_size;
+
+            if (chunk_data_size < 4) {
+                return ERROR_MALFORMED;
+            }
+
+            uint8_t buffer[256];
+            if (chunk_data_size > (off64_t)sizeof(buffer)) {
+                return ERROR_BUFFER_TOO_SMALL;
+            }
+
+            if (mDataSource->readAt(
+                        data_offset, buffer, chunk_data_size) < chunk_data_size) {
+                return ERROR_IO;
+            }
+
+            if (U32_AT(buffer) != 0) {
+                // Should be version 0, flags 0.
+                return ERROR_MALFORMED;
+            }
+
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
+            mLastTrack->meta.setData(
+                    kKeyESDS, kTypeESDS, &buffer[4], chunk_data_size - 4);
+
+            if (mPath.size() >= 2
+                    && mPath[mPath.size() - 2] == FOURCC('m', 'p', '4', 'a')) {
+                // Information from the ESDS must be relied on for proper
+                // setup of sample rate and channel count for MPEG4 Audio.
+                // The generic header appears to only contain generic
+                // information...
+
+                status_t err = updateAudioTrackInfoFromESDS_MPEG4Audio(
+                        &buffer[4], chunk_data_size - 4);
+
+                if (err != OK) {
+                    return err;
+                }
+            }
+            if (mPath.size() >= 2
+                    && mPath[mPath.size() - 2] == FOURCC('m', 'p', '4', 'v')) {
+                // Check if the video is MPEG2
+                ESDS esds(&buffer[4], chunk_data_size - 4);
+
+                uint8_t objectTypeIndication;
+                if (esds.getObjectTypeIndication(&objectTypeIndication) == OK) {
+                    if (objectTypeIndication >= 0x60 && objectTypeIndication <= 0x65) {
+                        mLastTrack->meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG2);
+                    }
+                }
+            }
+            break;
+        }
+
+        case FOURCC('b', 't', 'r', 't'):
+        {
+            *offset += chunk_size;
+            if (mLastTrack == NULL) {
+                return ERROR_MALFORMED;
+            }
+
+            uint8_t buffer[12];
+            if (chunk_data_size != sizeof(buffer)) {
+                return ERROR_MALFORMED;
+            }
+
+            if (mDataSource->readAt(
+                    data_offset, buffer, chunk_data_size) < chunk_data_size) {
+                return ERROR_IO;
+            }
+
+            uint32_t maxBitrate = U32_AT(&buffer[4]);
+            uint32_t avgBitrate = U32_AT(&buffer[8]);
+            if (maxBitrate > 0 && maxBitrate < INT32_MAX) {
+                mLastTrack->meta.setInt32(kKeyMaxBitRate, (int32_t)maxBitrate);
+            }
+            if (avgBitrate > 0 && avgBitrate < INT32_MAX) {
+                mLastTrack->meta.setInt32(kKeyBitRate, (int32_t)avgBitrate);
+            }
+            break;
+        }
+
+        case FOURCC('a', 'v', 'c', 'C'):
+        {
+            *offset += chunk_size;
+
+            auto buffer = heapbuffer<uint8_t>(chunk_data_size);
+
+            if (buffer.get() == NULL) {
+                ALOGE("b/28471206");
+                return NO_MEMORY;
+            }
+
+            if (mDataSource->readAt(
+                        data_offset, buffer.get(), chunk_data_size) < chunk_data_size) {
+                return ERROR_IO;
+            }
+
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
+            mLastTrack->meta.setData(
+                    kKeyAVCC, kTypeAVCC, buffer.get(), chunk_data_size);
+
+            break;
+        }
+        case FOURCC('h', 'v', 'c', 'C'):
+        {
+            auto buffer = heapbuffer<uint8_t>(chunk_data_size);
+
+            if (buffer.get() == NULL) {
+                ALOGE("b/28471206");
+                return NO_MEMORY;
+            }
+
+            if (mDataSource->readAt(
+                        data_offset, buffer.get(), chunk_data_size) < chunk_data_size) {
+                return ERROR_IO;
+            }
+
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
+            mLastTrack->meta.setData(
+                    kKeyHVCC, kTypeHVCC, buffer.get(), chunk_data_size);
+
+            *offset += chunk_size;
+            break;
+        }
+
+        case FOURCC('d', '2', '6', '3'):
+        {
+            *offset += chunk_size;
+            /*
+             * d263 contains a fixed 7 bytes part:
+             *   vendor - 4 bytes
+             *   version - 1 byte
+             *   level - 1 byte
+             *   profile - 1 byte
+             * optionally, "d263" box itself may contain a 16-byte
+             * bit rate box (bitr)
+             *   average bit rate - 4 bytes
+             *   max bit rate - 4 bytes
+             */
+            char buffer[23];
+            if (chunk_data_size != 7 &&
+                chunk_data_size != 23) {
+                ALOGE("Incorrect D263 box size %lld", (long long)chunk_data_size);
+                return ERROR_MALFORMED;
+            }
+
+            if (mDataSource->readAt(
+                    data_offset, buffer, chunk_data_size) < chunk_data_size) {
+                return ERROR_IO;
+            }
+
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
+            mLastTrack->meta.setData(kKeyD263, kTypeD263, buffer, chunk_data_size);
+
+            break;
+        }
+
+        case FOURCC('m', 'e', 't', 'a'):
+        {
+            off64_t stop_offset = *offset + chunk_size;
+            *offset = data_offset;
+            bool isParsingMetaKeys = underQTMetaPath(mPath, 2);
+            if (!isParsingMetaKeys) {
+                uint8_t buffer[4];
+                if (chunk_data_size < (off64_t)sizeof(buffer)) {
+                    *offset = stop_offset;
+                    return ERROR_MALFORMED;
+                }
+
+                if (mDataSource->readAt(
+                            data_offset, buffer, 4) < 4) {
+                    *offset = stop_offset;
+                    return ERROR_IO;
+                }
+
+                if (U32_AT(buffer) != 0) {
+                    // Should be version 0, flags 0.
+
+                    // If it's not, let's assume this is one of those
+                    // apparently malformed chunks that don't have flags
+                    // and completely different semantics than what's
+                    // in the MPEG4 specs and skip it.
+                    *offset = stop_offset;
+                    return OK;
+                }
+                *offset +=  sizeof(buffer);
+            }
+
+            while (*offset < stop_offset) {
+                status_t err = parseChunk(offset, depth + 1);
+                if (err != OK) {
+                    return err;
+                }
+            }
+
+            if (*offset != stop_offset) {
+                return ERROR_MALFORMED;
+            }
+            break;
+        }
+
+        case FOURCC('i', 'l', 'o', 'c'):
+        case FOURCC('i', 'i', 'n', 'f'):
+        case FOURCC('i', 'p', 'r', 'p'):
+        case FOURCC('p', 'i', 't', 'm'):
+        case FOURCC('i', 'd', 'a', 't'):
+        case FOURCC('i', 'r', 'e', 'f'):
+        case FOURCC('i', 'p', 'r', 'o'):
+        {
+            if (mIsHeif) {
+                if (mItemTable == NULL) {
+                    mItemTable = new ItemTable(mDataSource);
+                }
+                status_t err = mItemTable->parse(
+                        chunk_type, data_offset, chunk_data_size);
+                if (err != OK) {
+                    return err;
+                }
+            }
+            *offset += chunk_size;
+            break;
+        }
+
+        case FOURCC('m', 'e', 'a', 'n'):
+        case FOURCC('n', 'a', 'm', 'e'):
+        case FOURCC('d', 'a', 't', 'a'):
+        {
+            *offset += chunk_size;
+
+            if (mPath.size() == 6 && underMetaDataPath(mPath)) {
+                status_t err = parseITunesMetaData(data_offset, chunk_data_size);
+
+                if (err != OK) {
+                    return err;
+                }
+            }
+
+            break;
+        }
+
+        case FOURCC('m', 'v', 'h', 'd'):
+        {
+            *offset += chunk_size;
+
+            if (depth != 1) {
+                ALOGE("mvhd: depth %d", depth);
+                return ERROR_MALFORMED;
+            }
+            if (chunk_data_size < 32) {
+                return ERROR_MALFORMED;
+            }
+
+            uint8_t header[32];
+            if (mDataSource->readAt(
+                        data_offset, header, sizeof(header))
+                    < (ssize_t)sizeof(header)) {
+                return ERROR_IO;
+            }
+
+            uint64_t creationTime;
+            uint64_t duration = 0;
+            if (header[0] == 1) {
+                creationTime = U64_AT(&header[4]);
+                mHeaderTimescale = U32_AT(&header[20]);
+                duration = U64_AT(&header[24]);
+                if (duration == 0xffffffffffffffff) {
+                    duration = 0;
+                }
+            } else if (header[0] != 0) {
+                return ERROR_MALFORMED;
+            } else {
+                creationTime = U32_AT(&header[4]);
+                mHeaderTimescale = U32_AT(&header[12]);
+                uint32_t d32 = U32_AT(&header[16]);
+                if (d32 == 0xffffffff) {
+                    d32 = 0;
+                }
+                duration = d32;
+            }
+            if (duration != 0 && mHeaderTimescale != 0 && duration < UINT64_MAX / 1000000) {
+                mFileMetaData.setInt64(kKeyDuration, duration * 1000000 / mHeaderTimescale);
+            }
+
+            String8 s;
+            if (convertTimeToDate(creationTime, &s)) {
+                mFileMetaData.setCString(kKeyDate, s.string());
+            }
+
+
+            break;
+        }
+
+        case FOURCC('m', 'e', 'h', 'd'):
+        {
+            *offset += chunk_size;
+
+            if (chunk_data_size < 8) {
+                return ERROR_MALFORMED;
+            }
+
+            uint8_t flags[4];
+            if (mDataSource->readAt(
+                        data_offset, flags, sizeof(flags))
+                    < (ssize_t)sizeof(flags)) {
+                return ERROR_IO;
+            }
+
+            uint64_t duration = 0;
+            if (flags[0] == 1) {
+                // 64 bit
+                if (chunk_data_size < 12) {
+                    return ERROR_MALFORMED;
+                }
+                mDataSource->getUInt64(data_offset + 4, &duration);
+                if (duration == 0xffffffffffffffff) {
+                    duration = 0;
+                }
+            } else if (flags[0] == 0) {
+                // 32 bit
+                uint32_t d32;
+                mDataSource->getUInt32(data_offset + 4, &d32);
+                if (d32 == 0xffffffff) {
+                    d32 = 0;
+                }
+                duration = d32;
+            } else {
+                return ERROR_MALFORMED;
+            }
+
+            if (duration != 0 && mHeaderTimescale != 0) {
+                mFileMetaData.setInt64(kKeyDuration, duration * 1000000 / mHeaderTimescale);
+            }
+
+            break;
+        }
+
+        case FOURCC('m', 'd', 'a', 't'):
+        {
+            mMdatFound = true;
+
+            *offset += chunk_size;
+            break;
+        }
+
+        case FOURCC('h', 'd', 'l', 'r'):
+        {
+            *offset += chunk_size;
+
+            if (underQTMetaPath(mPath, 3)) {
+                break;
+            }
+
+            uint32_t buffer;
+            if (mDataSource->readAt(
+                        data_offset + 8, &buffer, 4) < 4) {
+                return ERROR_IO;
+            }
+
+            uint32_t type = ntohl(buffer);
+            // For the 3GPP file format, the handler-type within the 'hdlr' box
+            // shall be 'text'. We also want to support 'sbtl' handler type
+            // for a practical reason as various MPEG4 containers use it.
+            if (type == FOURCC('t', 'e', 'x', 't') || type == FOURCC('s', 'b', 't', 'l')) {
+                if (mLastTrack != NULL) {
+                    mLastTrack->meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_TEXT_3GPP);
+                }
+            }
+
+            break;
+        }
+
+        case FOURCC('k', 'e', 'y', 's'):
+        {
+            *offset += chunk_size;
+
+            if (underQTMetaPath(mPath, 3)) {
+                status_t err = parseQTMetaKey(data_offset, chunk_data_size);
+                if (err != OK) {
+                    return err;
+                }
+            }
+            break;
+        }
+
+        case FOURCC('t', 'r', 'e', 'x'):
+        {
+            *offset += chunk_size;
+
+            if (chunk_data_size < 24) {
+                return ERROR_IO;
+            }
+            Trex trex;
+            if (!mDataSource->getUInt32(data_offset + 4, &trex.track_ID) ||
+                !mDataSource->getUInt32(data_offset + 8, &trex.default_sample_description_index) ||
+                !mDataSource->getUInt32(data_offset + 12, &trex.default_sample_duration) ||
+                !mDataSource->getUInt32(data_offset + 16, &trex.default_sample_size) ||
+                !mDataSource->getUInt32(data_offset + 20, &trex.default_sample_flags)) {
+                return ERROR_IO;
+            }
+            mTrex.add(trex);
+            break;
+        }
+
+        case FOURCC('t', 'x', '3', 'g'):
+        {
+            if (mLastTrack == NULL)
+                return ERROR_MALFORMED;
+
+            uint32_t type;
+            const void *data;
+            size_t size = 0;
+            if (!mLastTrack->meta.findData(
+                    kKeyTextFormatData, &type, &data, &size)) {
+                size = 0;
+            }
+
+            if ((chunk_size > SIZE_MAX) || (SIZE_MAX - chunk_size <= size)) {
+                return ERROR_MALFORMED;
+            }
+
+            uint8_t *buffer = new (std::nothrow) uint8_t[size + chunk_size];
+            if (buffer == NULL) {
+                return ERROR_MALFORMED;
+            }
+
+            if (size > 0) {
+                memcpy(buffer, data, size);
+            }
+
+            if ((size_t)(mDataSource->readAt(*offset, buffer + size, chunk_size))
+                    < chunk_size) {
+                delete[] buffer;
+                buffer = NULL;
+
+                // advance read pointer so we don't end up reading this again
+                *offset += chunk_size;
+                return ERROR_IO;
+            }
+
+            mLastTrack->meta.setData(
+                    kKeyTextFormatData, 0, buffer, size + chunk_size);
+
+            delete[] buffer;
+
+            *offset += chunk_size;
+            break;
+        }
+
+        case FOURCC('c', 'o', 'v', 'r'):
+        {
+            *offset += chunk_size;
+
+            ALOGV("chunk_data_size = %" PRId64 " and data_offset = %" PRId64,
+                  chunk_data_size, data_offset);
+
+            if (chunk_data_size < 0 || static_cast<uint64_t>(chunk_data_size) >= SIZE_MAX - 1) {
+                return ERROR_MALFORMED;
+            }
+            auto buffer = heapbuffer<uint8_t>(chunk_data_size);
+            if (buffer.get() == NULL) {
+                ALOGE("b/28471206");
+                return NO_MEMORY;
+            }
+            if (mDataSource->readAt(
+                data_offset, buffer.get(), chunk_data_size) != (ssize_t)chunk_data_size) {
+                return ERROR_IO;
+            }
+            const int kSkipBytesOfDataBox = 16;
+            if (chunk_data_size <= kSkipBytesOfDataBox) {
+                return ERROR_MALFORMED;
+            }
+
+            mFileMetaData.setData(
+                kKeyAlbumArt, MetaData::TYPE_NONE,
+                buffer.get() + kSkipBytesOfDataBox, chunk_data_size - kSkipBytesOfDataBox);
+
+            break;
+        }
+
+        case FOURCC('c', 'o', 'l', 'r'):
+        {
+            *offset += chunk_size;
+            // this must be in a VisualSampleEntry box under the Sample Description Box ('stsd')
+            // ignore otherwise
+            if (depth >= 2 && mPath[depth - 2] == FOURCC('s', 't', 's', 'd')) {
+                status_t err = parseColorInfo(data_offset, chunk_data_size);
+                if (err != OK) {
+                    return err;
+                }
+            }
+
+            break;
+        }
+
+        case FOURCC('t', 'i', 't', 'l'):
+        case FOURCC('p', 'e', 'r', 'f'):
+        case FOURCC('a', 'u', 't', 'h'):
+        case FOURCC('g', 'n', 'r', 'e'):
+        case FOURCC('a', 'l', 'b', 'm'):
+        case FOURCC('y', 'r', 'r', 'c'):
+        {
+            *offset += chunk_size;
+
+            status_t err = parse3GPPMetaData(data_offset, chunk_data_size, depth);
+
+            if (err != OK) {
+                return err;
+            }
+
+            break;
+        }
+
+        case FOURCC('I', 'D', '3', '2'):
+        {
+            *offset += chunk_size;
+
+            if (chunk_data_size < 6) {
+                return ERROR_MALFORMED;
+            }
+
+            parseID3v2MetaData(data_offset + 6);
+
+            break;
+        }
+
+        case FOURCC('-', '-', '-', '-'):
+        {
+            mLastCommentMean.clear();
+            mLastCommentName.clear();
+            mLastCommentData.clear();
+            *offset += chunk_size;
+            break;
+        }
+
+        case FOURCC('s', 'i', 'd', 'x'):
+        {
+            status_t err = parseSegmentIndex(data_offset, chunk_data_size);
+            if (err != OK) {
+                return err;
+            }
+            *offset += chunk_size;
+            return UNKNOWN_ERROR; // stop parsing after sidx
+        }
+
+        case FOURCC('a', 'c', '-', '3'):
+        {
+            *offset += chunk_size;
+            return parseAC3SampleEntry(data_offset);
+        }
+
+        case FOURCC('f', 't', 'y', 'p'):
+        {
+            if (chunk_data_size < 8 || depth != 0) {
+                return ERROR_MALFORMED;
+            }
+
+            off64_t stop_offset = *offset + chunk_size;
+            uint32_t numCompatibleBrands = (chunk_data_size - 8) / 4;
+            std::set<uint32_t> brandSet;
+            for (size_t i = 0; i < numCompatibleBrands + 2; ++i) {
+                if (i == 1) {
+                    // Skip this index, it refers to the minorVersion,
+                    // not a brand.
+                    continue;
+                }
+
+                uint32_t brand;
+                if (mDataSource->readAt(data_offset + 4 * i, &brand, 4) < 4) {
+                    return ERROR_MALFORMED;
+                }
+
+                brand = ntohl(brand);
+                brandSet.insert(brand);
+            }
+
+            if (brandSet.count(FOURCC('q', 't', ' ', ' ')) > 0) {
+                mIsQT = true;
+            } else {
+                if (brandSet.count(FOURCC('m', 'i', 'f', '1')) > 0
+                 && brandSet.count(FOURCC('h', 'e', 'i', 'c')) > 0) {
+                    ALOGV("identified HEIF image");
+
+                    mIsHeif = true;
+                    brandSet.erase(FOURCC('m', 'i', 'f', '1'));
+                    brandSet.erase(FOURCC('h', 'e', 'i', 'c'));
+                }
+
+                if (!brandSet.empty()) {
+                    // This means that the file should have moov box.
+                    // It could be any iso files (mp4, heifs, etc.)
+                    mHasMoovBox = true;
+                    if (mIsHeif) {
+                        ALOGV("identified HEIF image with other tracks");
+                    }
+                }
+            }
+
+            *offset = stop_offset;
+
+            break;
+        }
+
+        default:
+        {
+            // check if we're parsing 'ilst' for meta keys
+            // if so, treat type as a number (key-id).
+            if (underQTMetaPath(mPath, 3)) {
+                status_t err = parseQTMetaVal(chunk_type, data_offset, chunk_data_size);
+                if (err != OK) {
+                    return err;
+                }
+            }
+
+            *offset += chunk_size;
+            break;
+        }
+    }
+
+    return OK;
+}
+
+status_t MPEG4Extractor::parseAC3SampleEntry(off64_t offset) {
+    // skip 16 bytes:
+    //  + 6-byte reserved,
+    //  + 2-byte data reference index,
+    //  + 8-byte reserved
+    offset += 16;
+    uint16_t channelCount;
+    if (!mDataSource->getUInt16(offset, &channelCount)) {
+        return ERROR_MALFORMED;
+    }
+    // skip 8 bytes:
+    //  + 2-byte channelCount,
+    //  + 2-byte sample size,
+    //  + 4-byte reserved
+    offset += 8;
+    uint16_t sampleRate;
+    if (!mDataSource->getUInt16(offset, &sampleRate)) {
+        ALOGE("MPEG4Extractor: error while reading ac-3 block: cannot read sample rate");
+        return ERROR_MALFORMED;
+    }
+
+    // skip 4 bytes:
+    //  + 2-byte sampleRate,
+    //  + 2-byte reserved
+    offset += 4;
+    return parseAC3SpecificBox(offset, sampleRate);
+}
+
+status_t MPEG4Extractor::parseAC3SpecificBox(
+        off64_t offset, uint16_t sampleRate) {
+    uint32_t size;
+    // + 4-byte size
+    // + 4-byte type
+    // + 3-byte payload
+    const uint32_t kAC3SpecificBoxSize = 11;
+    if (!mDataSource->getUInt32(offset, &size) || size < kAC3SpecificBoxSize) {
+        ALOGE("MPEG4Extractor: error while reading ac-3 block: cannot read specific box size");
+        return ERROR_MALFORMED;
+    }
+
+    offset += 4;
+    uint32_t type;
+    if (!mDataSource->getUInt32(offset, &type) || type != FOURCC('d', 'a', 'c', '3')) {
+        ALOGE("MPEG4Extractor: error while reading ac-3 specific block: header not dac3");
+        return ERROR_MALFORMED;
+    }
+
+    offset += 4;
+    const uint32_t kAC3SpecificBoxPayloadSize = 3;
+    uint8_t chunk[kAC3SpecificBoxPayloadSize];
+    if (mDataSource->readAt(offset, chunk, sizeof(chunk)) != sizeof(chunk)) {
+        ALOGE("MPEG4Extractor: error while reading ac-3 specific block: bitstream fields");
+        return ERROR_MALFORMED;
+    }
+
+    ABitReader br(chunk, sizeof(chunk));
+    static const unsigned channelCountTable[] = {2, 1, 2, 3, 3, 4, 4, 5};
+    static const unsigned sampleRateTable[] = {48000, 44100, 32000};
+
+    unsigned fscod = br.getBits(2);
+    if (fscod == 3) {
+        ALOGE("Incorrect fscod (3) in AC3 header");
+        return ERROR_MALFORMED;
+    }
+    unsigned boxSampleRate = sampleRateTable[fscod];
+    if (boxSampleRate != sampleRate) {
+        ALOGE("sample rate mismatch: boxSampleRate = %d, sampleRate = %d",
+            boxSampleRate, sampleRate);
+        return ERROR_MALFORMED;
+    }
+
+    unsigned bsid = br.getBits(5);
+    if (bsid > 8) {
+        ALOGW("Incorrect bsid in AC3 header. Possibly E-AC-3?");
+        return ERROR_MALFORMED;
+    }
+
+    // skip
+    unsigned bsmod __unused = br.getBits(3);
+
+    unsigned acmod = br.getBits(3);
+    unsigned lfeon = br.getBits(1);
+    unsigned channelCount = channelCountTable[acmod] + lfeon;
+
+    if (mLastTrack == NULL) {
+        return ERROR_MALFORMED;
+    }
+    mLastTrack->meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC3);
+    mLastTrack->meta.setInt32(kKeyChannelCount, channelCount);
+    mLastTrack->meta.setInt32(kKeySampleRate, sampleRate);
+    return OK;
+}
+
+status_t MPEG4Extractor::parseSegmentIndex(off64_t offset, size_t size) {
+  ALOGV("MPEG4Extractor::parseSegmentIndex");
+
+    if (size < 12) {
+      return -EINVAL;
+    }
+
+    uint32_t flags;
+    if (!mDataSource->getUInt32(offset, &flags)) {
+        return ERROR_MALFORMED;
+    }
+
+    uint32_t version = flags >> 24;
+    flags &= 0xffffff;
+
+    ALOGV("sidx version %d", version);
+
+    uint32_t referenceId;
+    if (!mDataSource->getUInt32(offset + 4, &referenceId)) {
+        return ERROR_MALFORMED;
+    }
+
+    uint32_t timeScale;
+    if (!mDataSource->getUInt32(offset + 8, &timeScale)) {
+        return ERROR_MALFORMED;
+    }
+    ALOGV("sidx refid/timescale: %d/%d", referenceId, timeScale);
+    if (timeScale == 0)
+        return ERROR_MALFORMED;
+
+    uint64_t earliestPresentationTime;
+    uint64_t firstOffset;
+
+    offset += 12;
+    size -= 12;
+
+    if (version == 0) {
+        if (size < 8) {
+            return -EINVAL;
+        }
+        uint32_t tmp;
+        if (!mDataSource->getUInt32(offset, &tmp)) {
+            return ERROR_MALFORMED;
+        }
+        earliestPresentationTime = tmp;
+        if (!mDataSource->getUInt32(offset + 4, &tmp)) {
+            return ERROR_MALFORMED;
+        }
+        firstOffset = tmp;
+        offset += 8;
+        size -= 8;
+    } else {
+        if (size < 16) {
+            return -EINVAL;
+        }
+        if (!mDataSource->getUInt64(offset, &earliestPresentationTime)) {
+            return ERROR_MALFORMED;
+        }
+        if (!mDataSource->getUInt64(offset + 8, &firstOffset)) {
+            return ERROR_MALFORMED;
+        }
+        offset += 16;
+        size -= 16;
+    }
+    ALOGV("sidx pres/off: %" PRIu64 "/%" PRIu64, earliestPresentationTime, firstOffset);
+
+    if (size < 4) {
+        return -EINVAL;
+    }
+
+    uint16_t referenceCount;
+    if (!mDataSource->getUInt16(offset + 2, &referenceCount)) {
+        return ERROR_MALFORMED;
+    }
+    offset += 4;
+    size -= 4;
+    ALOGV("refcount: %d", referenceCount);
+
+    if (size < referenceCount * 12) {
+        return -EINVAL;
+    }
+
+    uint64_t total_duration = 0;
+    for (unsigned int i = 0; i < referenceCount; i++) {
+        uint32_t d1, d2, d3;
+
+        if (!mDataSource->getUInt32(offset, &d1) ||     // size
+            !mDataSource->getUInt32(offset + 4, &d2) || // duration
+            !mDataSource->getUInt32(offset + 8, &d3)) { // flags
+            return ERROR_MALFORMED;
+        }
+
+        if (d1 & 0x80000000) {
+            ALOGW("sub-sidx boxes not supported yet");
+        }
+        bool sap = d3 & 0x80000000;
+        uint32_t saptype = (d3 >> 28) & 7;
+        if (!sap || (saptype != 1 && saptype != 2)) {
+            // type 1 and 2 are sync samples
+            ALOGW("not a stream access point, or unsupported type: %08x", d3);
+        }
+        total_duration += d2;
+        offset += 12;
+        ALOGV(" item %d, %08x %08x %08x", i, d1, d2, d3);
+        SidxEntry se;
+        se.mSize = d1 & 0x7fffffff;
+        se.mDurationUs = 1000000LL * d2 / timeScale;
+        mSidxEntries.add(se);
+    }
+
+    uint64_t sidxDuration = total_duration * 1000000 / timeScale;
+
+    if (mLastTrack == NULL)
+        return ERROR_MALFORMED;
+
+    int64_t metaDuration;
+    if (!mLastTrack->meta.findInt64(kKeyDuration, &metaDuration) || metaDuration == 0) {
+        mLastTrack->meta.setInt64(kKeyDuration, sidxDuration);
+    }
+    return OK;
+}
+
+status_t MPEG4Extractor::parseQTMetaKey(off64_t offset, size_t size) {
+    if (size < 8) {
+        return ERROR_MALFORMED;
+    }
+
+    uint32_t count;
+    if (!mDataSource->getUInt32(offset + 4, &count)) {
+        return ERROR_MALFORMED;
+    }
+
+    if (mMetaKeyMap.size() > 0) {
+        ALOGW("'keys' atom seen again, discarding existing entries");
+        mMetaKeyMap.clear();
+    }
+
+    off64_t keyOffset = offset + 8;
+    off64_t stopOffset = offset + size;
+    for (size_t i = 1; i <= count; i++) {
+        if (keyOffset + 8 > stopOffset) {
+            return ERROR_MALFORMED;
+        }
+
+        uint32_t keySize;
+        if (!mDataSource->getUInt32(keyOffset, &keySize)
+                || keySize < 8
+                || keyOffset + keySize > stopOffset) {
+            return ERROR_MALFORMED;
+        }
+
+        uint32_t type;
+        if (!mDataSource->getUInt32(keyOffset + 4, &type)
+                || type != FOURCC('m', 'd', 't', 'a')) {
+            return ERROR_MALFORMED;
+        }
+
+        keySize -= 8;
+        keyOffset += 8;
+
+        auto keyData = heapbuffer<uint8_t>(keySize);
+        if (keyData.get() == NULL) {
+            return ERROR_MALFORMED;
+        }
+        if (mDataSource->readAt(
+                keyOffset, keyData.get(), keySize) < (ssize_t) keySize) {
+            return ERROR_MALFORMED;
+        }
+
+        AString key((const char *)keyData.get(), keySize);
+        mMetaKeyMap.add(i, key);
+
+        keyOffset += keySize;
+    }
+    return OK;
+}
+
+status_t MPEG4Extractor::parseQTMetaVal(
+        int32_t keyId, off64_t offset, size_t size) {
+    ssize_t index = mMetaKeyMap.indexOfKey(keyId);
+    if (index < 0) {
+        // corresponding key is not present, ignore
+        return ERROR_MALFORMED;
+    }
+
+    if (size <= 16) {
+        return ERROR_MALFORMED;
+    }
+    uint32_t dataSize;
+    if (!mDataSource->getUInt32(offset, &dataSize)
+            || dataSize > size || dataSize <= 16) {
+        return ERROR_MALFORMED;
+    }
+    uint32_t atomFourCC;
+    if (!mDataSource->getUInt32(offset + 4, &atomFourCC)
+            || atomFourCC != FOURCC('d', 'a', 't', 'a')) {
+        return ERROR_MALFORMED;
+    }
+    uint32_t dataType;
+    if (!mDataSource->getUInt32(offset + 8, &dataType)
+            || ((dataType & 0xff000000) != 0)) {
+        // not well-known type
+        return ERROR_MALFORMED;
+    }
+
+    dataSize -= 16;
+    offset += 16;
+
+    if (dataType == 23 && dataSize >= 4) {
+        // BE Float32
+        uint32_t val;
+        if (!mDataSource->getUInt32(offset, &val)) {
+            return ERROR_MALFORMED;
+        }
+        if (!strcasecmp(mMetaKeyMap[index].c_str(), "com.android.capture.fps")) {
+            mFileMetaData.setFloat(kKeyCaptureFramerate, *(float *)&val);
+        }
+    } else if (dataType == 67 && dataSize >= 4) {
+        // BE signed int32
+        uint32_t val;
+        if (!mDataSource->getUInt32(offset, &val)) {
+            return ERROR_MALFORMED;
+        }
+        if (!strcasecmp(mMetaKeyMap[index].c_str(), "com.android.video.temporal_layers_count")) {
+            mFileMetaData.setInt32(kKeyTemporalLayerCount, val);
+        }
+    } else {
+        // add more keys if needed
+        ALOGV("ignoring key: type %d, size %d", dataType, dataSize);
+    }
+
+    return OK;
+}
+
+status_t MPEG4Extractor::parseTrackHeader(
+        off64_t data_offset, off64_t data_size) {
+    if (data_size < 4) {
+        return ERROR_MALFORMED;
+    }
+
+    uint8_t version;
+    if (mDataSource->readAt(data_offset, &version, 1) < 1) {
+        return ERROR_IO;
+    }
+
+    size_t dynSize = (version == 1) ? 36 : 24;
+
+    uint8_t buffer[36 + 60];
+
+    if (data_size != (off64_t)dynSize + 60) {
+        return ERROR_MALFORMED;
+    }
+
+    if (mDataSource->readAt(
+                data_offset, buffer, data_size) < (ssize_t)data_size) {
+        return ERROR_IO;
+    }
+
+    uint64_t ctime __unused, mtime __unused, duration __unused;
+    int32_t id;
+
+    if (version == 1) {
+        ctime = U64_AT(&buffer[4]);
+        mtime = U64_AT(&buffer[12]);
+        id = U32_AT(&buffer[20]);
+        duration = U64_AT(&buffer[28]);
+    } else if (version == 0) {
+        ctime = U32_AT(&buffer[4]);
+        mtime = U32_AT(&buffer[8]);
+        id = U32_AT(&buffer[12]);
+        duration = U32_AT(&buffer[20]);
+    } else {
+        return ERROR_UNSUPPORTED;
+    }
+
+    if (mLastTrack == NULL)
+        return ERROR_MALFORMED;
+
+    mLastTrack->meta.setInt32(kKeyTrackID, id);
+
+    size_t matrixOffset = dynSize + 16;
+    int32_t a00 = U32_AT(&buffer[matrixOffset]);
+    int32_t a01 = U32_AT(&buffer[matrixOffset + 4]);
+    int32_t a10 = U32_AT(&buffer[matrixOffset + 12]);
+    int32_t a11 = U32_AT(&buffer[matrixOffset + 16]);
+
+#if 0
+    int32_t dx = U32_AT(&buffer[matrixOffset + 8]);
+    int32_t dy = U32_AT(&buffer[matrixOffset + 20]);
+
+    ALOGI("x' = %.2f * x + %.2f * y + %.2f",
+         a00 / 65536.0f, a01 / 65536.0f, dx / 65536.0f);
+    ALOGI("y' = %.2f * x + %.2f * y + %.2f",
+         a10 / 65536.0f, a11 / 65536.0f, dy / 65536.0f);
+#endif
+
+    uint32_t rotationDegrees;
+
+    static const int32_t kFixedOne = 0x10000;
+    if (a00 == kFixedOne && a01 == 0 && a10 == 0 && a11 == kFixedOne) {
+        // Identity, no rotation
+        rotationDegrees = 0;
+    } else if (a00 == 0 && a01 == kFixedOne && a10 == -kFixedOne && a11 == 0) {
+        rotationDegrees = 90;
+    } else if (a00 == 0 && a01 == -kFixedOne && a10 == kFixedOne && a11 == 0) {
+        rotationDegrees = 270;
+    } else if (a00 == -kFixedOne && a01 == 0 && a10 == 0 && a11 == -kFixedOne) {
+        rotationDegrees = 180;
+    } else {
+        ALOGW("We only support 0,90,180,270 degree rotation matrices");
+        rotationDegrees = 0;
+    }
+
+    if (rotationDegrees != 0) {
+        mLastTrack->meta.setInt32(kKeyRotation, rotationDegrees);
+    }
+
+    // Handle presentation display size, which could be different
+    // from the image size indicated by kKeyWidth and kKeyHeight.
+    uint32_t width = U32_AT(&buffer[dynSize + 52]);
+    uint32_t height = U32_AT(&buffer[dynSize + 56]);
+    mLastTrack->meta.setInt32(kKeyDisplayWidth, width >> 16);
+    mLastTrack->meta.setInt32(kKeyDisplayHeight, height >> 16);
+
+    return OK;
+}
+
+status_t MPEG4Extractor::parseITunesMetaData(off64_t offset, size_t size) {
+    if (size == 0) {
+        return OK;
+    }
+
+    if (size < 4 || size == SIZE_MAX) {
+        return ERROR_MALFORMED;
+    }
+
+    uint8_t *buffer = new (std::nothrow) uint8_t[size + 1];
+    if (buffer == NULL) {
+        return ERROR_MALFORMED;
+    }
+    if (mDataSource->readAt(
+                offset, buffer, size) != (ssize_t)size) {
+        delete[] buffer;
+        buffer = NULL;
+
+        return ERROR_IO;
+    }
+
+    uint32_t flags = U32_AT(buffer);
+
+    uint32_t metadataKey = 0;
+    char chunk[5];
+    MakeFourCCString(mPath[4], chunk);
+    ALOGV("meta: %s @ %lld", chunk, (long long)offset);
+    switch ((int32_t)mPath[4]) {
+        case FOURCC(0xa9, 'a', 'l', 'b'):
+        {
+            metadataKey = kKeyAlbum;
+            break;
+        }
+        case FOURCC(0xa9, 'A', 'R', 'T'):
+        {
+            metadataKey = kKeyArtist;
+            break;
+        }
+        case FOURCC('a', 'A', 'R', 'T'):
+        {
+            metadataKey = kKeyAlbumArtist;
+            break;
+        }
+        case FOURCC(0xa9, 'd', 'a', 'y'):
+        {
+            metadataKey = kKeyYear;
+            break;
+        }
+        case FOURCC(0xa9, 'n', 'a', 'm'):
+        {
+            metadataKey = kKeyTitle;
+            break;
+        }
+        case FOURCC(0xa9, 'w', 'r', 't'):
+        {
+            metadataKey = kKeyWriter;
+            break;
+        }
+        case FOURCC('c', 'o', 'v', 'r'):
+        {
+            metadataKey = kKeyAlbumArt;
+            break;
+        }
+        case FOURCC('g', 'n', 'r', 'e'):
+        {
+            metadataKey = kKeyGenre;
+            break;
+        }
+        case FOURCC(0xa9, 'g', 'e', 'n'):
+        {
+            metadataKey = kKeyGenre;
+            break;
+        }
+        case FOURCC('c', 'p', 'i', 'l'):
+        {
+            if (size == 9 && flags == 21) {
+                char tmp[16];
+                sprintf(tmp, "%d",
+                        (int)buffer[size - 1]);
+
+                mFileMetaData.setCString(kKeyCompilation, tmp);
+            }
+            break;
+        }
+        case FOURCC('t', 'r', 'k', 'n'):
+        {
+            if (size == 16 && flags == 0) {
+                char tmp[16];
+                uint16_t* pTrack = (uint16_t*)&buffer[10];
+                uint16_t* pTotalTracks = (uint16_t*)&buffer[12];
+                sprintf(tmp, "%d/%d", ntohs(*pTrack), ntohs(*pTotalTracks));
+
+                mFileMetaData.setCString(kKeyCDTrackNumber, tmp);
+            }
+            break;
+        }
+        case FOURCC('d', 'i', 's', 'k'):
+        {
+            if ((size == 14 || size == 16) && flags == 0) {
+                char tmp[16];
+                uint16_t* pDisc = (uint16_t*)&buffer[10];
+                uint16_t* pTotalDiscs = (uint16_t*)&buffer[12];
+                sprintf(tmp, "%d/%d", ntohs(*pDisc), ntohs(*pTotalDiscs));
+
+                mFileMetaData.setCString(kKeyDiscNumber, tmp);
+            }
+            break;
+        }
+        case FOURCC('-', '-', '-', '-'):
+        {
+            buffer[size] = '\0';
+            switch (mPath[5]) {
+                case FOURCC('m', 'e', 'a', 'n'):
+                    mLastCommentMean.setTo((const char *)buffer + 4);
+                    break;
+                case FOURCC('n', 'a', 'm', 'e'):
+                    mLastCommentName.setTo((const char *)buffer + 4);
+                    break;
+                case FOURCC('d', 'a', 't', 'a'):
+                    if (size < 8) {
+                        delete[] buffer;
+                        buffer = NULL;
+                        ALOGE("b/24346430");
+                        return ERROR_MALFORMED;
+                    }
+                    mLastCommentData.setTo((const char *)buffer + 8);
+                    break;
+            }
+
+            // Once we have a set of mean/name/data info, go ahead and process
+            // it to see if its something we are interested in.  Whether or not
+            // were are interested in the specific tag, make sure to clear out
+            // the set so we can be ready to process another tuple should one
+            // show up later in the file.
+            if ((mLastCommentMean.length() != 0) &&
+                (mLastCommentName.length() != 0) &&
+                (mLastCommentData.length() != 0)) {
+
+                if (mLastCommentMean == "com.apple.iTunes"
+                        && mLastCommentName == "iTunSMPB") {
+                    int32_t delay, padding;
+                    if (sscanf(mLastCommentData,
+                               " %*x %x %x %*x", &delay, &padding) == 2) {
+                        if (mLastTrack == NULL) {
+                            delete[] buffer;
+                            return ERROR_MALFORMED;
+                        }
+
+                        mLastTrack->meta.setInt32(kKeyEncoderDelay, delay);
+                        mLastTrack->meta.setInt32(kKeyEncoderPadding, padding);
+                    }
+                }
+
+                mLastCommentMean.clear();
+                mLastCommentName.clear();
+                mLastCommentData.clear();
+            }
+            break;
+        }
+
+        default:
+            break;
+    }
+
+    if (size >= 8 && metadataKey && !mFileMetaData.hasData(metadataKey)) {
+        if (metadataKey == kKeyAlbumArt) {
+            mFileMetaData.setData(
+                    kKeyAlbumArt, MetaData::TYPE_NONE,
+                    buffer + 8, size - 8);
+        } else if (metadataKey == kKeyGenre) {
+            if (flags == 0) {
+                // uint8_t genre code, iTunes genre codes are
+                // the standard id3 codes, except they start
+                // at 1 instead of 0 (e.g. Pop is 14, not 13)
+                // We use standard id3 numbering, so subtract 1.
+                int genrecode = (int)buffer[size - 1];
+                genrecode--;
+                if (genrecode < 0) {
+                    genrecode = 255; // reserved for 'unknown genre'
+                }
+                char genre[10];
+                sprintf(genre, "%d", genrecode);
+
+                mFileMetaData.setCString(metadataKey, genre);
+            } else if (flags == 1) {
+                // custom genre string
+                buffer[size] = '\0';
+
+                mFileMetaData.setCString(
+                        metadataKey, (const char *)buffer + 8);
+            }
+        } else {
+            buffer[size] = '\0';
+
+            mFileMetaData.setCString(
+                    metadataKey, (const char *)buffer + 8);
+        }
+    }
+
+    delete[] buffer;
+    buffer = NULL;
+
+    return OK;
+}
+
+status_t MPEG4Extractor::parseColorInfo(off64_t offset, size_t size) {
+    if (size < 4 || size == SIZE_MAX || mLastTrack == NULL) {
+        return ERROR_MALFORMED;
+    }
+
+    uint8_t *buffer = new (std::nothrow) uint8_t[size + 1];
+    if (buffer == NULL) {
+        return ERROR_MALFORMED;
+    }
+    if (mDataSource->readAt(offset, buffer, size) != (ssize_t)size) {
+        delete[] buffer;
+        buffer = NULL;
+
+        return ERROR_IO;
+    }
+
+    int32_t type = U32_AT(&buffer[0]);
+    if ((type == FOURCC('n', 'c', 'l', 'x') && size >= 11)
+            || (type == FOURCC('n', 'c', 'l', 'c') && size >= 10)) {
+        int32_t primaries = U16_AT(&buffer[4]);
+        int32_t transfer = U16_AT(&buffer[6]);
+        int32_t coeffs = U16_AT(&buffer[8]);
+        bool fullRange = (type == FOURCC('n', 'c', 'l', 'x')) && (buffer[10] & 128);
+
+        ColorAspects aspects;
+        ColorUtils::convertIsoColorAspectsToCodecAspects(
+                primaries, transfer, coeffs, fullRange, aspects);
+
+        // only store the first color specification
+        if (!mLastTrack->meta.hasData(kKeyColorPrimaries)) {
+            mLastTrack->meta.setInt32(kKeyColorPrimaries, aspects.mPrimaries);
+            mLastTrack->meta.setInt32(kKeyTransferFunction, aspects.mTransfer);
+            mLastTrack->meta.setInt32(kKeyColorMatrix, aspects.mMatrixCoeffs);
+            mLastTrack->meta.setInt32(kKeyColorRange, aspects.mRange);
+        }
+    }
+
+    delete[] buffer;
+    buffer = NULL;
+
+    return OK;
+}
+
+status_t MPEG4Extractor::parse3GPPMetaData(off64_t offset, size_t size, int depth) {
+    if (size < 4 || size == SIZE_MAX) {
+        return ERROR_MALFORMED;
+    }
+
+    uint8_t *buffer = new (std::nothrow) uint8_t[size + 1];
+    if (buffer == NULL) {
+        return ERROR_MALFORMED;
+    }
+    if (mDataSource->readAt(
+                offset, buffer, size) != (ssize_t)size) {
+        delete[] buffer;
+        buffer = NULL;
+
+        return ERROR_IO;
+    }
+
+    uint32_t metadataKey = 0;
+    switch (mPath[depth]) {
+        case FOURCC('t', 'i', 't', 'l'):
+        {
+            metadataKey = kKeyTitle;
+            break;
+        }
+        case FOURCC('p', 'e', 'r', 'f'):
+        {
+            metadataKey = kKeyArtist;
+            break;
+        }
+        case FOURCC('a', 'u', 't', 'h'):
+        {
+            metadataKey = kKeyWriter;
+            break;
+        }
+        case FOURCC('g', 'n', 'r', 'e'):
+        {
+            metadataKey = kKeyGenre;
+            break;
+        }
+        case FOURCC('a', 'l', 'b', 'm'):
+        {
+            if (buffer[size - 1] != '\0') {
+              char tmp[4];
+              sprintf(tmp, "%u", buffer[size - 1]);
+
+              mFileMetaData.setCString(kKeyCDTrackNumber, tmp);
+            }
+
+            metadataKey = kKeyAlbum;
+            break;
+        }
+        case FOURCC('y', 'r', 'r', 'c'):
+        {
+            if (size < 6) {
+                delete[] buffer;
+                buffer = NULL;
+                ALOGE("b/62133227");
+                android_errorWriteLog(0x534e4554, "62133227");
+                return ERROR_MALFORMED;
+            }
+            char tmp[5];
+            uint16_t year = U16_AT(&buffer[4]);
+
+            if (year < 10000) {
+                sprintf(tmp, "%u", year);
+
+                mFileMetaData.setCString(kKeyYear, tmp);
+            }
+            break;
+        }
+
+        default:
+            break;
+    }
+
+    if (metadataKey > 0) {
+        bool isUTF8 = true; // Common case
+        char16_t *framedata = NULL;
+        int len16 = 0; // Number of UTF-16 characters
+
+        // smallest possible valid UTF-16 string w BOM: 0xfe 0xff 0x00 0x00
+        if (size < 6) {
+            delete[] buffer;
+            buffer = NULL;
+            return ERROR_MALFORMED;
+        }
+
+        if (size - 6 >= 4) {
+            len16 = ((size - 6) / 2) - 1; // don't include 0x0000 terminator
+            framedata = (char16_t *)(buffer + 6);
+            if (0xfffe == *framedata) {
+                // endianness marker (BOM) doesn't match host endianness
+                for (int i = 0; i < len16; i++) {
+                    framedata[i] = bswap_16(framedata[i]);
+                }
+                // BOM is now swapped to 0xfeff, we will execute next block too
+            }
+
+            if (0xfeff == *framedata) {
+                // Remove the BOM
+                framedata++;
+                len16--;
+                isUTF8 = false;
+            }
+            // else normal non-zero-length UTF-8 string
+            // we can't handle UTF-16 without BOM as there is no other
+            // indication of encoding.
+        }
+
+        if (isUTF8) {
+            buffer[size] = 0;
+            mFileMetaData.setCString(metadataKey, (const char *)buffer + 6);
+        } else {
+            // Convert from UTF-16 string to UTF-8 string.
+            String8 tmpUTF8str(framedata, len16);
+            mFileMetaData.setCString(metadataKey, tmpUTF8str.string());
+        }
+    }
+
+    delete[] buffer;
+    buffer = NULL;
+
+    return OK;
+}
+
+void MPEG4Extractor::parseID3v2MetaData(off64_t offset) {
+    ID3 id3(mDataSource, true /* ignorev1 */, offset);
+
+    if (id3.isValid()) {
+        struct Map {
+            int key;
+            const char *tag1;
+            const char *tag2;
+        };
+        static const Map kMap[] = {
+            { kKeyAlbum, "TALB", "TAL" },
+            { kKeyArtist, "TPE1", "TP1" },
+            { kKeyAlbumArtist, "TPE2", "TP2" },
+            { kKeyComposer, "TCOM", "TCM" },
+            { kKeyGenre, "TCON", "TCO" },
+            { kKeyTitle, "TIT2", "TT2" },
+            { kKeyYear, "TYE", "TYER" },
+            { kKeyAuthor, "TXT", "TEXT" },
+            { kKeyCDTrackNumber, "TRK", "TRCK" },
+            { kKeyDiscNumber, "TPA", "TPOS" },
+            { kKeyCompilation, "TCP", "TCMP" },
+        };
+        static const size_t kNumMapEntries = sizeof(kMap) / sizeof(kMap[0]);
+
+        for (size_t i = 0; i < kNumMapEntries; ++i) {
+            if (!mFileMetaData.hasData(kMap[i].key)) {
+                ID3::Iterator *it = new ID3::Iterator(id3, kMap[i].tag1);
+                if (it->done()) {
+                    delete it;
+                    it = new ID3::Iterator(id3, kMap[i].tag2);
+                }
+
+                if (it->done()) {
+                    delete it;
+                    continue;
+                }
+
+                String8 s;
+                it->getString(&s);
+                delete it;
+
+                mFileMetaData.setCString(kMap[i].key, s);
+            }
+        }
+
+        size_t dataSize;
+        String8 mime;
+        const void *data = id3.getAlbumArt(&dataSize, &mime);
+
+        if (data) {
+            mFileMetaData.setData(kKeyAlbumArt, MetaData::TYPE_NONE, data, dataSize);
+            mFileMetaData.setCString(kKeyAlbumArtMIME, mime.string());
+        }
+    }
+}
+
+MediaTrack *MPEG4Extractor::getTrack(size_t index) {
+    status_t err;
+    if ((err = readMetaData()) != OK) {
+        return NULL;
+    }
+
+    Track *track = mFirstTrack;
+    while (index > 0) {
+        if (track == NULL) {
+            return NULL;
+        }
+
+        track = track->next;
+        --index;
+    }
+
+    if (track == NULL) {
+        return NULL;
+    }
+
+
+    Trex *trex = NULL;
+    int32_t trackId;
+    if (track->meta.findInt32(kKeyTrackID, &trackId)) {
+        for (size_t i = 0; i < mTrex.size(); i++) {
+            Trex *t = &mTrex.editItemAt(i);
+            if (t->track_ID == (uint32_t) trackId) {
+                trex = t;
+                break;
+            }
+        }
+    } else {
+        ALOGE("b/21657957");
+        return NULL;
+    }
+
+    ALOGV("getTrack called, pssh: %zu", mPssh.size());
+
+    const char *mime;
+    if (!track->meta.findCString(kKeyMIMEType, &mime)) {
+        return NULL;
+    }
+
+    sp<ItemTable> itemTable;
+    if (!strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC)) {
+        uint32_t type;
+        const void *data;
+        size_t size;
+        if (!track->meta.findData(kKeyAVCC, &type, &data, &size)) {
+            return NULL;
+        }
+
+        const uint8_t *ptr = (const uint8_t *)data;
+
+        if (size < 7 || ptr[0] != 1) {  // configurationVersion == 1
+            return NULL;
+        }
+    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_HEVC)
+            || !strcasecmp(mime, MEDIA_MIMETYPE_IMAGE_ANDROID_HEIC)) {
+        uint32_t type;
+        const void *data;
+        size_t size;
+        if (!track->meta.findData(kKeyHVCC, &type, &data, &size)) {
+            return NULL;
+        }
+
+        const uint8_t *ptr = (const uint8_t *)data;
+
+        if (size < 22 || ptr[0] != 1) {  // configurationVersion == 1
+            return NULL;
+        }
+        if (!strcasecmp(mime, MEDIA_MIMETYPE_IMAGE_ANDROID_HEIC)) {
+            itemTable = mItemTable;
+        }
+    }
+
+    MPEG4Source *source =  new MPEG4Source(
+            track->meta, mDataSource, track->timescale, track->sampleTable,
+            mSidxEntries, trex, mMoofOffset, itemTable);
+    if (source->init() != OK) {
+        delete source;
+        return NULL;
+    }
+    return source;
+}
+
+// static
+status_t MPEG4Extractor::verifyTrack(Track *track) {
+    const char *mime;
+    CHECK(track->meta.findCString(kKeyMIMEType, &mime));
+
+    uint32_t type;
+    const void *data;
+    size_t size;
+    if (!strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC)) {
+        if (!track->meta.findData(kKeyAVCC, &type, &data, &size)
+                || type != kTypeAVCC) {
+            return ERROR_MALFORMED;
+        }
+    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_HEVC)) {
+        if (!track->meta.findData(kKeyHVCC, &type, &data, &size)
+                    || type != kTypeHVCC) {
+            return ERROR_MALFORMED;
+        }
+    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_MPEG4)
+            || !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_MPEG2)
+            || !strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC)) {
+        if (!track->meta.findData(kKeyESDS, &type, &data, &size)
+                || type != kTypeESDS) {
+            return ERROR_MALFORMED;
+        }
+    }
+
+    if (track->sampleTable == NULL || !track->sampleTable->isValid()) {
+        // Make sure we have all the metadata we need.
+        ALOGE("stbl atom missing/invalid.");
+        return ERROR_MALFORMED;
+    }
+
+    if (track->timescale == 0) {
+        ALOGE("timescale invalid.");
+        return ERROR_MALFORMED;
+    }
+
+    return OK;
+}
+
+typedef enum {
+    //AOT_NONE             = -1,
+    //AOT_NULL_OBJECT      = 0,
+    //AOT_AAC_MAIN         = 1, /**< Main profile                              */
+    AOT_AAC_LC           = 2,   /**< Low Complexity object                     */
+    //AOT_AAC_SSR          = 3,
+    //AOT_AAC_LTP          = 4,
+    AOT_SBR              = 5,
+    //AOT_AAC_SCAL         = 6,
+    //AOT_TWIN_VQ          = 7,
+    //AOT_CELP             = 8,
+    //AOT_HVXC             = 9,
+    //AOT_RSVD_10          = 10, /**< (reserved)                                */
+    //AOT_RSVD_11          = 11, /**< (reserved)                                */
+    //AOT_TTSI             = 12, /**< TTSI Object                               */
+    //AOT_MAIN_SYNTH       = 13, /**< Main Synthetic object                     */
+    //AOT_WAV_TAB_SYNTH    = 14, /**< Wavetable Synthesis object                */
+    //AOT_GEN_MIDI         = 15, /**< General MIDI object                       */
+    //AOT_ALG_SYNTH_AUD_FX = 16, /**< Algorithmic Synthesis and Audio FX object */
+    AOT_ER_AAC_LC        = 17,   /**< Error Resilient(ER) AAC Low Complexity    */
+    //AOT_RSVD_18          = 18, /**< (reserved)                                */
+    //AOT_ER_AAC_LTP       = 19, /**< Error Resilient(ER) AAC LTP object        */
+    AOT_ER_AAC_SCAL      = 20,   /**< Error Resilient(ER) AAC Scalable object   */
+    //AOT_ER_TWIN_VQ       = 21, /**< Error Resilient(ER) TwinVQ object         */
+    AOT_ER_BSAC          = 22,   /**< Error Resilient(ER) BSAC object           */
+    AOT_ER_AAC_LD        = 23,   /**< Error Resilient(ER) AAC LowDelay object   */
+    //AOT_ER_CELP          = 24, /**< Error Resilient(ER) CELP object           */
+    //AOT_ER_HVXC          = 25, /**< Error Resilient(ER) HVXC object           */
+    //AOT_ER_HILN          = 26, /**< Error Resilient(ER) HILN object           */
+    //AOT_ER_PARA          = 27, /**< Error Resilient(ER) Parametric object     */
+    //AOT_RSVD_28          = 28, /**< might become SSC                          */
+    AOT_PS               = 29,   /**< PS, Parametric Stereo (includes SBR)      */
+    //AOT_MPEGS            = 30, /**< MPEG Surround                             */
+
+    AOT_ESCAPE           = 31,   /**< Signal AOT uses more than 5 bits          */
+
+    //AOT_MP3ONMP4_L1      = 32, /**< MPEG-Layer1 in mp4                        */
+    //AOT_MP3ONMP4_L2      = 33, /**< MPEG-Layer2 in mp4                        */
+    //AOT_MP3ONMP4_L3      = 34, /**< MPEG-Layer3 in mp4                        */
+    //AOT_RSVD_35          = 35, /**< might become DST                          */
+    //AOT_RSVD_36          = 36, /**< might become ALS                          */
+    //AOT_AAC_SLS          = 37, /**< AAC + SLS                                 */
+    //AOT_SLS              = 38, /**< SLS                                       */
+    //AOT_ER_AAC_ELD       = 39, /**< AAC Enhanced Low Delay                    */
+
+    //AOT_USAC             = 42, /**< USAC                                      */
+    //AOT_SAOC             = 43, /**< SAOC                                      */
+    //AOT_LD_MPEGS         = 44, /**< Low Delay MPEG Surround                   */
+
+    //AOT_RSVD50           = 50,  /**< Interim AOT for Rsvd50                   */
+} AUDIO_OBJECT_TYPE;
+
+status_t MPEG4Extractor::updateAudioTrackInfoFromESDS_MPEG4Audio(
+        const void *esds_data, size_t esds_size) {
+    ESDS esds(esds_data, esds_size);
+
+    uint8_t objectTypeIndication;
+    if (esds.getObjectTypeIndication(&objectTypeIndication) != OK) {
+        return ERROR_MALFORMED;
+    }
+
+    if (objectTypeIndication == 0xe1) {
+        // This isn't MPEG4 audio at all, it's QCELP 14k...
+        if (mLastTrack == NULL)
+            return ERROR_MALFORMED;
+
+        mLastTrack->meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_QCELP);
+        return OK;
+    }
+
+    if (objectTypeIndication  == 0x6b) {
+        // The media subtype is MP3 audio
+        // Our software MP3 audio decoder may not be able to handle
+        // packetized MP3 audio; for now, lets just return ERROR_UNSUPPORTED
+        ALOGE("MP3 track in MP4/3GPP file is not supported");
+        return ERROR_UNSUPPORTED;
+    }
+
+    if (mLastTrack != NULL) {
+        uint32_t maxBitrate = 0;
+        uint32_t avgBitrate = 0;
+        esds.getBitRate(&maxBitrate, &avgBitrate);
+        if (maxBitrate > 0 && maxBitrate < INT32_MAX) {
+            mLastTrack->meta.setInt32(kKeyMaxBitRate, (int32_t)maxBitrate);
+        }
+        if (avgBitrate > 0 && avgBitrate < INT32_MAX) {
+            mLastTrack->meta.setInt32(kKeyBitRate, (int32_t)avgBitrate);
+        }
+    }
+
+    const uint8_t *csd;
+    size_t csd_size;
+    if (esds.getCodecSpecificInfo(
+                (const void **)&csd, &csd_size) != OK) {
+        return ERROR_MALFORMED;
+    }
+
+    if (kUseHexDump) {
+        printf("ESD of size %zu\n", csd_size);
+        hexdump(csd, csd_size);
+    }
+
+    if (csd_size == 0) {
+        // There's no further information, i.e. no codec specific data
+        // Let's assume that the information provided in the mpeg4 headers
+        // is accurate and hope for the best.
+
+        return OK;
+    }
+
+    if (csd_size < 2) {
+        return ERROR_MALFORMED;
+    }
+
+    static uint32_t kSamplingRate[] = {
+        96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+        16000, 12000, 11025, 8000, 7350
+    };
+
+    ABitReader br(csd, csd_size);
+    uint32_t objectType = br.getBits(5);
+
+    if (objectType == 31) {  // AAC-ELD => additional 6 bits
+        objectType = 32 + br.getBits(6);
+    }
+
+    if (mLastTrack == NULL)
+        return ERROR_MALFORMED;
+
+    //keep AOT type
+    mLastTrack->meta.setInt32(kKeyAACAOT, objectType);
+
+    uint32_t freqIndex = br.getBits(4);
+
+    int32_t sampleRate = 0;
+    int32_t numChannels = 0;
+    if (freqIndex == 15) {
+        if (br.numBitsLeft() < 28) return ERROR_MALFORMED;
+        sampleRate = br.getBits(24);
+        numChannels = br.getBits(4);
+    } else {
+        if (br.numBitsLeft() < 4) return ERROR_MALFORMED;
+        numChannels = br.getBits(4);
+
+        if (freqIndex == 13 || freqIndex == 14) {
+            return ERROR_MALFORMED;
+        }
+
+        sampleRate = kSamplingRate[freqIndex];
+    }
+
+    if (objectType == AOT_SBR || objectType == AOT_PS) {//SBR specific config per 14496-3 table 1.13
+        if (br.numBitsLeft() < 4) return ERROR_MALFORMED;
+        uint32_t extFreqIndex = br.getBits(4);
+        int32_t extSampleRate __unused;
+        if (extFreqIndex == 15) {
+            if (csd_size < 8) {
+                return ERROR_MALFORMED;
+            }
+            if (br.numBitsLeft() < 24) return ERROR_MALFORMED;
+            extSampleRate = br.getBits(24);
+        } else {
+            if (extFreqIndex == 13 || extFreqIndex == 14) {
+                return ERROR_MALFORMED;
+            }
+            extSampleRate = kSamplingRate[extFreqIndex];
+        }
+        //TODO: save the extension sampling rate value in meta data =>
+        //      mLastTrack->meta.setInt32(kKeyExtSampleRate, extSampleRate);
+    }
+
+    switch (numChannels) {
+        // values defined in 14496-3_2009 amendment-4 Table 1.19 - Channel Configuration
+        case 0:
+        case 1:// FC
+        case 2:// FL FR
+        case 3:// FC, FL FR
+        case 4:// FC, FL FR, RC
+        case 5:// FC, FL FR, SL SR
+        case 6:// FC, FL FR, SL SR, LFE
+            //numChannels already contains the right value
+            break;
+        case 11:// FC, FL FR, SL SR, RC, LFE
+            numChannels = 7;
+            break;
+        case 7: // FC, FCL FCR, FL FR, SL SR, LFE
+        case 12:// FC, FL  FR,  SL SR, RL RR, LFE
+        case 14:// FC, FL  FR,  SL SR, LFE, FHL FHR
+            numChannels = 8;
+            break;
+        default:
+            return ERROR_UNSUPPORTED;
+    }
+
+    {
+        if (objectType == AOT_SBR || objectType == AOT_PS) {
+            if (br.numBitsLeft() < 5) return ERROR_MALFORMED;
+            objectType = br.getBits(5);
+
+            if (objectType == AOT_ESCAPE) {
+                if (br.numBitsLeft() < 6) return ERROR_MALFORMED;
+                objectType = 32 + br.getBits(6);
+            }
+        }
+        if (objectType == AOT_AAC_LC || objectType == AOT_ER_AAC_LC ||
+                objectType == AOT_ER_AAC_LD || objectType == AOT_ER_AAC_SCAL ||
+                objectType == AOT_ER_BSAC) {
+            if (br.numBitsLeft() < 2) return ERROR_MALFORMED;
+            const int32_t frameLengthFlag __unused = br.getBits(1);
+
+            const int32_t dependsOnCoreCoder = br.getBits(1);
+
+            if (dependsOnCoreCoder ) {
+                if (br.numBitsLeft() < 14) return ERROR_MALFORMED;
+                const int32_t coreCoderDelay __unused = br.getBits(14);
+            }
+
+            int32_t extensionFlag = -1;
+            if (br.numBitsLeft() > 0) {
+                extensionFlag = br.getBits(1);
+            } else {
+                switch (objectType) {
+                // 14496-3 4.5.1.1 extensionFlag
+                case AOT_AAC_LC:
+                    extensionFlag = 0;
+                    break;
+                case AOT_ER_AAC_LC:
+                case AOT_ER_AAC_SCAL:
+                case AOT_ER_BSAC:
+                case AOT_ER_AAC_LD:
+                    extensionFlag = 1;
+                    break;
+                default:
+                    return ERROR_MALFORMED;
+                    break;
+                }
+                ALOGW("csd missing extension flag; assuming %d for object type %u.",
+                        extensionFlag, objectType);
+            }
+
+            if (numChannels == 0) {
+                int32_t channelsEffectiveNum = 0;
+                int32_t channelsNum = 0;
+                if (br.numBitsLeft() < 32) {
+                    return ERROR_MALFORMED;
+                }
+                const int32_t ElementInstanceTag __unused = br.getBits(4);
+                const int32_t Profile __unused = br.getBits(2);
+                const int32_t SamplingFrequencyIndex __unused = br.getBits(4);
+                const int32_t NumFrontChannelElements = br.getBits(4);
+                const int32_t NumSideChannelElements = br.getBits(4);
+                const int32_t NumBackChannelElements = br.getBits(4);
+                const int32_t NumLfeChannelElements = br.getBits(2);
+                const int32_t NumAssocDataElements __unused = br.getBits(3);
+                const int32_t NumValidCcElements __unused = br.getBits(4);
+
+                const int32_t MonoMixdownPresent = br.getBits(1);
+
+                if (MonoMixdownPresent != 0) {
+                    if (br.numBitsLeft() < 4) return ERROR_MALFORMED;
+                    const int32_t MonoMixdownElementNumber __unused = br.getBits(4);
+                }
+
+                if (br.numBitsLeft() < 1) return ERROR_MALFORMED;
+                const int32_t StereoMixdownPresent = br.getBits(1);
+                if (StereoMixdownPresent != 0) {
+                    if (br.numBitsLeft() < 4) return ERROR_MALFORMED;
+                    const int32_t StereoMixdownElementNumber __unused = br.getBits(4);
+                }
+
+                if (br.numBitsLeft() < 1) return ERROR_MALFORMED;
+                const int32_t MatrixMixdownIndexPresent = br.getBits(1);
+                if (MatrixMixdownIndexPresent != 0) {
+                    if (br.numBitsLeft() < 3) return ERROR_MALFORMED;
+                    const int32_t MatrixMixdownIndex __unused = br.getBits(2);
+                    const int32_t PseudoSurroundEnable __unused = br.getBits(1);
+                }
+
+                int i;
+                for (i=0; i < NumFrontChannelElements; i++) {
+                    if (br.numBitsLeft() < 5) return ERROR_MALFORMED;
+                    const int32_t FrontElementIsCpe = br.getBits(1);
+                    const int32_t FrontElementTagSelect __unused = br.getBits(4);
+                    channelsNum += FrontElementIsCpe ? 2 : 1;
+                }
+
+                for (i=0; i < NumSideChannelElements; i++) {
+                    if (br.numBitsLeft() < 5) return ERROR_MALFORMED;
+                    const int32_t SideElementIsCpe = br.getBits(1);
+                    const int32_t SideElementTagSelect __unused = br.getBits(4);
+                    channelsNum += SideElementIsCpe ? 2 : 1;
+                }
+
+                for (i=0; i < NumBackChannelElements; i++) {
+                    if (br.numBitsLeft() < 5) return ERROR_MALFORMED;
+                    const int32_t BackElementIsCpe = br.getBits(1);
+                    const int32_t BackElementTagSelect __unused = br.getBits(4);
+                    channelsNum += BackElementIsCpe ? 2 : 1;
+                }
+                channelsEffectiveNum = channelsNum;
+
+                for (i=0; i < NumLfeChannelElements; i++) {
+                    if (br.numBitsLeft() < 4) return ERROR_MALFORMED;
+                    const int32_t LfeElementTagSelect __unused = br.getBits(4);
+                    channelsNum += 1;
+                }
+                ALOGV("mpeg4 audio channelsNum = %d", channelsNum);
+                ALOGV("mpeg4 audio channelsEffectiveNum = %d", channelsEffectiveNum);
+                numChannels = channelsNum;
+            }
+        }
+    }
+
+    if (numChannels == 0) {
+        return ERROR_UNSUPPORTED;
+    }
+
+    if (mLastTrack == NULL)
+        return ERROR_MALFORMED;
+
+    int32_t prevSampleRate;
+    CHECK(mLastTrack->meta.findInt32(kKeySampleRate, &prevSampleRate));
+
+    if (prevSampleRate != sampleRate) {
+        ALOGV("mpeg4 audio sample rate different from previous setting. "
+             "was: %d, now: %d", prevSampleRate, sampleRate);
+    }
+
+    mLastTrack->meta.setInt32(kKeySampleRate, sampleRate);
+
+    int32_t prevChannelCount;
+    CHECK(mLastTrack->meta.findInt32(kKeyChannelCount, &prevChannelCount));
+
+    if (prevChannelCount != numChannels) {
+        ALOGV("mpeg4 audio channel count different from previous setting. "
+             "was: %d, now: %d", prevChannelCount, numChannels);
+    }
+
+    mLastTrack->meta.setInt32(kKeyChannelCount, numChannels);
+
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+MPEG4Source::MPEG4Source(
+        MetaDataBase &format,
+        DataSourceBase *dataSource,
+        int32_t timeScale,
+        const sp<SampleTable> &sampleTable,
+        Vector<SidxEntry> &sidx,
+        const Trex *trex,
+        off64_t firstMoofOffset,
+        const sp<ItemTable> &itemTable)
+    : mFormat(format),
+      mDataSource(dataSource),
+      mTimescale(timeScale),
+      mSampleTable(sampleTable),
+      mCurrentSampleIndex(0),
+      mCurrentFragmentIndex(0),
+      mSegments(sidx),
+      mTrex(trex),
+      mFirstMoofOffset(firstMoofOffset),
+      mCurrentMoofOffset(firstMoofOffset),
+      mNextMoofOffset(-1),
+      mCurrentTime(0),
+      mDefaultEncryptedByteBlock(0),
+      mDefaultSkipByteBlock(0),
+      mCurrentSampleInfoAllocSize(0),
+      mCurrentSampleInfoSizes(NULL),
+      mCurrentSampleInfoOffsetsAllocSize(0),
+      mCurrentSampleInfoOffsets(NULL),
+      mIsAVC(false),
+      mIsHEVC(false),
+      mNALLengthSize(0),
+      mStarted(false),
+      mGroup(NULL),
+      mBuffer(NULL),
+      mWantsNALFragments(false),
+      mSrcBuffer(NULL),
+      mIsHeif(itemTable != NULL),
+      mItemTable(itemTable) {
+
+    memset(&mTrackFragmentHeaderInfo, 0, sizeof(mTrackFragmentHeaderInfo));
+
+    mFormat.findInt32(kKeyCryptoMode, &mCryptoMode);
+    mDefaultIVSize = 0;
+    mFormat.findInt32(kKeyCryptoDefaultIVSize, &mDefaultIVSize);
+    uint32_t keytype;
+    const void *key;
+    size_t keysize;
+    if (mFormat.findData(kKeyCryptoKey, &keytype, &key, &keysize)) {
+        CHECK(keysize <= 16);
+        memset(mCryptoKey, 0, 16);
+        memcpy(mCryptoKey, key, keysize);
+    }
+
+    mFormat.findInt32(kKeyEncryptedByteBlock, &mDefaultEncryptedByteBlock);
+    mFormat.findInt32(kKeySkipByteBlock, &mDefaultSkipByteBlock);
+
+    const char *mime;
+    bool success = mFormat.findCString(kKeyMIMEType, &mime);
+    CHECK(success);
+
+    mIsAVC = !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC);
+    mIsHEVC = !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_HEVC) ||
+              !strcasecmp(mime, MEDIA_MIMETYPE_IMAGE_ANDROID_HEIC);
+
+    if (mIsAVC) {
+        uint32_t type;
+        const void *data;
+        size_t size;
+        CHECK(format.findData(kKeyAVCC, &type, &data, &size));
+
+        const uint8_t *ptr = (const uint8_t *)data;
+
+        CHECK(size >= 7);
+        CHECK_EQ((unsigned)ptr[0], 1u);  // configurationVersion == 1
+
+        // The number of bytes used to encode the length of a NAL unit.
+        mNALLengthSize = 1 + (ptr[4] & 3);
+    } else if (mIsHEVC) {
+        uint32_t type;
+        const void *data;
+        size_t size;
+        CHECK(format.findData(kKeyHVCC, &type, &data, &size));
+
+        const uint8_t *ptr = (const uint8_t *)data;
+
+        CHECK(size >= 22);
+        CHECK_EQ((unsigned)ptr[0], 1u);  // configurationVersion == 1
+
+        mNALLengthSize = 1 + (ptr[14 + 7] & 3);
+    }
+
+    CHECK(format.findInt32(kKeyTrackID, &mTrackId));
+
+}
+
+status_t MPEG4Source::init() {
+    if (mFirstMoofOffset != 0) {
+        off64_t offset = mFirstMoofOffset;
+        return parseChunk(&offset);
+    }
+    return OK;
+}
+
+MPEG4Source::~MPEG4Source() {
+    if (mStarted) {
+        stop();
+    }
+    free(mCurrentSampleInfoSizes);
+    free(mCurrentSampleInfoOffsets);
+}
+
+status_t MPEG4Source::start(MetaDataBase *params) {
+    Mutex::Autolock autoLock(mLock);
+
+    CHECK(!mStarted);
+
+    int32_t val;
+    if (params && params->findInt32(kKeyWantsNALFragments, &val)
+        && val != 0) {
+        mWantsNALFragments = true;
+    } else {
+        mWantsNALFragments = false;
+    }
+
+    int32_t tmp;
+    CHECK(mFormat.findInt32(kKeyMaxInputSize, &tmp));
+    size_t max_size = tmp;
+
+    // A somewhat arbitrary limit that should be sufficient for 8k video frames
+    // If you see the message below for a valid input stream: increase the limit
+    const size_t kMaxBufferSize = 64 * 1024 * 1024;
+    if (max_size > kMaxBufferSize) {
+        ALOGE("bogus max input size: %zu > %zu", max_size, kMaxBufferSize);
+        return ERROR_MALFORMED;
+    }
+    if (max_size == 0) {
+        ALOGE("zero max input size");
+        return ERROR_MALFORMED;
+    }
+
+    // Allow up to kMaxBuffers, but not if the total exceeds kMaxBufferSize.
+    const size_t kInitialBuffers = 2;
+    const size_t kMaxBuffers = 8;
+    const size_t realMaxBuffers = min(kMaxBufferSize / max_size, kMaxBuffers);
+    mGroup = new MediaBufferGroup(kInitialBuffers, max_size, realMaxBuffers);
+    mSrcBuffer = new (std::nothrow) uint8_t[max_size];
+    if (mSrcBuffer == NULL) {
+        // file probably specified a bad max size
+        delete mGroup;
+        mGroup = NULL;
+        return ERROR_MALFORMED;
+    }
+
+    mStarted = true;
+
+    return OK;
+}
+
+status_t MPEG4Source::stop() {
+    Mutex::Autolock autoLock(mLock);
+
+    CHECK(mStarted);
+
+    if (mBuffer != NULL) {
+        mBuffer->release();
+        mBuffer = NULL;
+    }
+
+    delete[] mSrcBuffer;
+    mSrcBuffer = NULL;
+
+    delete mGroup;
+    mGroup = NULL;
+
+    mStarted = false;
+    mCurrentSampleIndex = 0;
+
+    return OK;
+}
+
+status_t MPEG4Source::parseChunk(off64_t *offset) {
+    uint32_t hdr[2];
+    if (mDataSource->readAt(*offset, hdr, 8) < 8) {
+        return ERROR_IO;
+    }
+    uint64_t chunk_size = ntohl(hdr[0]);
+    uint32_t chunk_type = ntohl(hdr[1]);
+    off64_t data_offset = *offset + 8;
+
+    if (chunk_size == 1) {
+        if (mDataSource->readAt(*offset + 8, &chunk_size, 8) < 8) {
+            return ERROR_IO;
+        }
+        chunk_size = ntoh64(chunk_size);
+        data_offset += 8;
+
+        if (chunk_size < 16) {
+            // The smallest valid chunk is 16 bytes long in this case.
+            return ERROR_MALFORMED;
+        }
+    } else if (chunk_size < 8) {
+        // The smallest valid chunk is 8 bytes long.
+        return ERROR_MALFORMED;
+    }
+
+    char chunk[5];
+    MakeFourCCString(chunk_type, chunk);
+    ALOGV("MPEG4Source chunk %s @ %#llx", chunk, (long long)*offset);
+
+    off64_t chunk_data_size = *offset + chunk_size - data_offset;
+
+    switch(chunk_type) {
+
+        case FOURCC('t', 'r', 'a', 'f'):
+        case FOURCC('m', 'o', 'o', 'f'): {
+            off64_t stop_offset = *offset + chunk_size;
+            *offset = data_offset;
+            while (*offset < stop_offset) {
+                status_t err = parseChunk(offset);
+                if (err != OK) {
+                    return err;
+                }
+            }
+            if (chunk_type == FOURCC('m', 'o', 'o', 'f')) {
+                // *offset points to the box following this moof. Find the next moof from there.
+
+                while (true) {
+                    if (mDataSource->readAt(*offset, hdr, 8) < 8) {
+                        // no more box to the end of file.
+                        break;
+                    }
+                    chunk_size = ntohl(hdr[0]);
+                    chunk_type = ntohl(hdr[1]);
+                    if (chunk_size == 1) {
+                        // ISO/IEC 14496-12:2012, 8.8.4 Movie Fragment Box, moof is a Box
+                        // which is defined in 4.2 Object Structure.
+                        // When chunk_size==1, 8 bytes follows as "largesize".
+                        if (mDataSource->readAt(*offset + 8, &chunk_size, 8) < 8) {
+                            return ERROR_IO;
+                        }
+                        chunk_size = ntoh64(chunk_size);
+                        if (chunk_size < 16) {
+                            // The smallest valid chunk is 16 bytes long in this case.
+                            return ERROR_MALFORMED;
+                        }
+                    } else if (chunk_size == 0) {
+                        // next box extends to end of file.
+                    } else if (chunk_size < 8) {
+                        // The smallest valid chunk is 8 bytes long in this case.
+                        return ERROR_MALFORMED;
+                    }
+
+                    if (chunk_type == FOURCC('m', 'o', 'o', 'f')) {
+                        mNextMoofOffset = *offset;
+                        break;
+                    } else if (chunk_size == 0) {
+                        break;
+                    }
+                    *offset += chunk_size;
+                }
+            }
+            break;
+        }
+
+        case FOURCC('t', 'f', 'h', 'd'): {
+                status_t err;
+                if ((err = parseTrackFragmentHeader(data_offset, chunk_data_size)) != OK) {
+                    return err;
+                }
+                *offset += chunk_size;
+                break;
+        }
+
+        case FOURCC('t', 'r', 'u', 'n'): {
+                status_t err;
+                if (mLastParsedTrackId == mTrackId) {
+                    if ((err = parseTrackFragmentRun(data_offset, chunk_data_size)) != OK) {
+                        return err;
+                    }
+                }
+
+                *offset += chunk_size;
+                break;
+        }
+
+        case FOURCC('s', 'a', 'i', 'z'): {
+            status_t err;
+            if ((err = parseSampleAuxiliaryInformationSizes(data_offset, chunk_data_size)) != OK) {
+                return err;
+            }
+            *offset += chunk_size;
+            break;
+        }
+        case FOURCC('s', 'a', 'i', 'o'): {
+            status_t err;
+            if ((err = parseSampleAuxiliaryInformationOffsets(data_offset, chunk_data_size)) != OK) {
+                return err;
+            }
+            *offset += chunk_size;
+            break;
+        }
+
+        case FOURCC('s', 'e', 'n', 'c'): {
+            status_t err;
+            if ((err = parseSampleEncryption(data_offset)) != OK) {
+                return err;
+            }
+            *offset += chunk_size;
+            break;
+        }
+
+        case FOURCC('m', 'd', 'a', 't'): {
+            // parse DRM info if present
+            ALOGV("MPEG4Source::parseChunk mdat");
+            // if saiz/saoi was previously observed, do something with the sampleinfos
+            *offset += chunk_size;
+            break;
+        }
+
+        default: {
+            *offset += chunk_size;
+            break;
+        }
+    }
+    return OK;
+}
+
+status_t MPEG4Source::parseSampleAuxiliaryInformationSizes(
+        off64_t offset, off64_t /* size */) {
+    ALOGV("parseSampleAuxiliaryInformationSizes");
+    // 14496-12 8.7.12
+    uint8_t version;
+    if (mDataSource->readAt(
+            offset, &version, sizeof(version))
+            < (ssize_t)sizeof(version)) {
+        return ERROR_IO;
+    }
+
+    if (version != 0) {
+        return ERROR_UNSUPPORTED;
+    }
+    offset++;
+
+    uint32_t flags;
+    if (!mDataSource->getUInt24(offset, &flags)) {
+        return ERROR_IO;
+    }
+    offset += 3;
+
+    if (flags & 1) {
+        uint32_t tmp;
+        if (!mDataSource->getUInt32(offset, &tmp)) {
+            return ERROR_MALFORMED;
+        }
+        mCurrentAuxInfoType = tmp;
+        offset += 4;
+        if (!mDataSource->getUInt32(offset, &tmp)) {
+            return ERROR_MALFORMED;
+        }
+        mCurrentAuxInfoTypeParameter = tmp;
+        offset += 4;
+    }
+
+    uint8_t defsize;
+    if (mDataSource->readAt(offset, &defsize, 1) != 1) {
+        return ERROR_MALFORMED;
+    }
+    mCurrentDefaultSampleInfoSize = defsize;
+    offset++;
+
+    uint32_t smplcnt;
+    if (!mDataSource->getUInt32(offset, &smplcnt)) {
+        return ERROR_MALFORMED;
+    }
+    mCurrentSampleInfoCount = smplcnt;
+    offset += 4;
+
+    if (mCurrentDefaultSampleInfoSize != 0) {
+        ALOGV("@@@@ using default sample info size of %d", mCurrentDefaultSampleInfoSize);
+        return OK;
+    }
+    if (smplcnt > mCurrentSampleInfoAllocSize) {
+        uint8_t * newPtr =  (uint8_t*) realloc(mCurrentSampleInfoSizes, smplcnt);
+        if (newPtr == NULL) {
+            ALOGE("failed to realloc %u -> %u", mCurrentSampleInfoAllocSize, smplcnt);
+            return NO_MEMORY;
+        }
+        mCurrentSampleInfoSizes = newPtr;
+        mCurrentSampleInfoAllocSize = smplcnt;
+    }
+
+    mDataSource->readAt(offset, mCurrentSampleInfoSizes, smplcnt);
+    return OK;
+}
+
+status_t MPEG4Source::parseSampleAuxiliaryInformationOffsets(
+        off64_t offset, off64_t /* size */) {
+    ALOGV("parseSampleAuxiliaryInformationOffsets");
+    // 14496-12 8.7.13
+    uint8_t version;
+    if (mDataSource->readAt(offset, &version, sizeof(version)) != 1) {
+        return ERROR_IO;
+    }
+    offset++;
+
+    uint32_t flags;
+    if (!mDataSource->getUInt24(offset, &flags)) {
+        return ERROR_IO;
+    }
+    offset += 3;
+
+    uint32_t entrycount;
+    if (!mDataSource->getUInt32(offset, &entrycount)) {
+        return ERROR_IO;
+    }
+    offset += 4;
+    if (entrycount == 0) {
+        return OK;
+    }
+    if (entrycount > UINT32_MAX / 8) {
+        return ERROR_MALFORMED;
+    }
+
+    if (entrycount > mCurrentSampleInfoOffsetsAllocSize) {
+        uint64_t *newPtr = (uint64_t *)realloc(mCurrentSampleInfoOffsets, entrycount * 8);
+        if (newPtr == NULL) {
+            ALOGE("failed to realloc %u -> %u", mCurrentSampleInfoOffsetsAllocSize, entrycount * 8);
+            return NO_MEMORY;
+        }
+        mCurrentSampleInfoOffsets = newPtr;
+        mCurrentSampleInfoOffsetsAllocSize = entrycount;
+    }
+    mCurrentSampleInfoOffsetCount = entrycount;
+
+    if (mCurrentSampleInfoOffsets == NULL) {
+        return OK;
+    }
+
+    for (size_t i = 0; i < entrycount; i++) {
+        if (version == 0) {
+            uint32_t tmp;
+            if (!mDataSource->getUInt32(offset, &tmp)) {
+                return ERROR_IO;
+            }
+            mCurrentSampleInfoOffsets[i] = tmp;
+            offset += 4;
+        } else {
+            uint64_t tmp;
+            if (!mDataSource->getUInt64(offset, &tmp)) {
+                return ERROR_IO;
+            }
+            mCurrentSampleInfoOffsets[i] = tmp;
+            offset += 8;
+        }
+    }
+
+    // parse clear/encrypted data
+
+    off64_t drmoffset = mCurrentSampleInfoOffsets[0]; // from moof
+
+    drmoffset += mCurrentMoofOffset;
+
+    return parseClearEncryptedSizes(drmoffset, false, 0);
+}
+
+status_t MPEG4Source::parseClearEncryptedSizes(off64_t offset, bool isSubsampleEncryption, uint32_t flags) {
+
+    int ivlength;
+    CHECK(mFormat.findInt32(kKeyCryptoDefaultIVSize, &ivlength));
+
+    // only 0, 8 and 16 byte initialization vectors are supported
+    if (ivlength != 0 && ivlength != 8 && ivlength != 16) {
+        ALOGW("unsupported IV length: %d", ivlength);
+        return ERROR_MALFORMED;
+    }
+
+    uint32_t sampleCount = mCurrentSampleInfoCount;
+    if (isSubsampleEncryption) {
+        if (!mDataSource->getUInt32(offset, &sampleCount)) {
+            return ERROR_IO;
+        }
+        offset += 4;
+    }
+
+    // read CencSampleAuxiliaryDataFormats
+    for (size_t i = 0; i < sampleCount; i++) {
+        if (i >= mCurrentSamples.size()) {
+            ALOGW("too few samples");
+            break;
+        }
+        Sample *smpl = &mCurrentSamples.editItemAt(i);
+        if (!smpl->clearsizes.isEmpty()) {
+            continue;
+        }
+
+        memset(smpl->iv, 0, 16);
+        if (mDataSource->readAt(offset, smpl->iv, ivlength) != ivlength) {
+            return ERROR_IO;
+        }
+
+        offset += ivlength;
+
+        bool readSubsamples;
+        if (isSubsampleEncryption) {
+            readSubsamples = flags & 2;
+        } else {
+            int32_t smplinfosize = mCurrentDefaultSampleInfoSize;
+            if (smplinfosize == 0) {
+                smplinfosize = mCurrentSampleInfoSizes[i];
+            }
+            readSubsamples = smplinfosize > ivlength;
+        }
+
+        if (readSubsamples) {
+            uint16_t numsubsamples;
+            if (!mDataSource->getUInt16(offset, &numsubsamples)) {
+                return ERROR_IO;
+            }
+            offset += 2;
+            for (size_t j = 0; j < numsubsamples; j++) {
+                uint16_t numclear;
+                uint32_t numencrypted;
+                if (!mDataSource->getUInt16(offset, &numclear)) {
+                    return ERROR_IO;
+                }
+                offset += 2;
+                if (!mDataSource->getUInt32(offset, &numencrypted)) {
+                    return ERROR_IO;
+                }
+                offset += 4;
+                smpl->clearsizes.add(numclear);
+                smpl->encryptedsizes.add(numencrypted);
+            }
+        } else {
+            smpl->clearsizes.add(0);
+            smpl->encryptedsizes.add(smpl->size);
+        }
+    }
+
+    return OK;
+}
+
+status_t MPEG4Source::parseSampleEncryption(off64_t offset) {
+    uint32_t flags;
+    if (!mDataSource->getUInt32(offset, &flags)) { // actually version + flags
+        return ERROR_MALFORMED;
+    }
+    return parseClearEncryptedSizes(offset + 4, true, flags);
+}
+
+status_t MPEG4Source::parseTrackFragmentHeader(off64_t offset, off64_t size) {
+
+    if (size < 8) {
+        return -EINVAL;
+    }
+
+    uint32_t flags;
+    if (!mDataSource->getUInt32(offset, &flags)) { // actually version + flags
+        return ERROR_MALFORMED;
+    }
+
+    if (flags & 0xff000000) {
+        return -EINVAL;
+    }
+
+    if (!mDataSource->getUInt32(offset + 4, (uint32_t*)&mLastParsedTrackId)) {
+        return ERROR_MALFORMED;
+    }
+
+    if (mLastParsedTrackId != mTrackId) {
+        // this is not the right track, skip it
+        return OK;
+    }
+
+    mTrackFragmentHeaderInfo.mFlags = flags;
+    mTrackFragmentHeaderInfo.mTrackID = mLastParsedTrackId;
+    offset += 8;
+    size -= 8;
+
+    ALOGV("fragment header: %08x %08x", flags, mTrackFragmentHeaderInfo.mTrackID);
+
+    if (flags & TrackFragmentHeaderInfo::kBaseDataOffsetPresent) {
+        if (size < 8) {
+            return -EINVAL;
+        }
+
+        if (!mDataSource->getUInt64(offset, &mTrackFragmentHeaderInfo.mBaseDataOffset)) {
+            return ERROR_MALFORMED;
+        }
+        offset += 8;
+        size -= 8;
+    }
+
+    if (flags & TrackFragmentHeaderInfo::kSampleDescriptionIndexPresent) {
+        if (size < 4) {
+            return -EINVAL;
+        }
+
+        if (!mDataSource->getUInt32(offset, &mTrackFragmentHeaderInfo.mSampleDescriptionIndex)) {
+            return ERROR_MALFORMED;
+        }
+        offset += 4;
+        size -= 4;
+    }
+
+    if (flags & TrackFragmentHeaderInfo::kDefaultSampleDurationPresent) {
+        if (size < 4) {
+            return -EINVAL;
+        }
+
+        if (!mDataSource->getUInt32(offset, &mTrackFragmentHeaderInfo.mDefaultSampleDuration)) {
+            return ERROR_MALFORMED;
+        }
+        offset += 4;
+        size -= 4;
+    }
+
+    if (flags & TrackFragmentHeaderInfo::kDefaultSampleSizePresent) {
+        if (size < 4) {
+            return -EINVAL;
+        }
+
+        if (!mDataSource->getUInt32(offset, &mTrackFragmentHeaderInfo.mDefaultSampleSize)) {
+            return ERROR_MALFORMED;
+        }
+        offset += 4;
+        size -= 4;
+    }
+
+    if (flags & TrackFragmentHeaderInfo::kDefaultSampleFlagsPresent) {
+        if (size < 4) {
+            return -EINVAL;
+        }
+
+        if (!mDataSource->getUInt32(offset, &mTrackFragmentHeaderInfo.mDefaultSampleFlags)) {
+            return ERROR_MALFORMED;
+        }
+        offset += 4;
+        size -= 4;
+    }
+
+    if (!(flags & TrackFragmentHeaderInfo::kBaseDataOffsetPresent)) {
+        mTrackFragmentHeaderInfo.mBaseDataOffset = mCurrentMoofOffset;
+    }
+
+    mTrackFragmentHeaderInfo.mDataOffset = 0;
+    return OK;
+}
+
+status_t MPEG4Source::parseTrackFragmentRun(off64_t offset, off64_t size) {
+
+    ALOGV("MPEG4Extractor::parseTrackFragmentRun");
+    if (size < 8) {
+        return -EINVAL;
+    }
+
+    enum {
+        kDataOffsetPresent                  = 0x01,
+        kFirstSampleFlagsPresent            = 0x04,
+        kSampleDurationPresent              = 0x100,
+        kSampleSizePresent                  = 0x200,
+        kSampleFlagsPresent                 = 0x400,
+        kSampleCompositionTimeOffsetPresent = 0x800,
+    };
+
+    uint32_t flags;
+    if (!mDataSource->getUInt32(offset, &flags)) {
+        return ERROR_MALFORMED;
+    }
+    // |version| only affects SampleCompositionTimeOffset field.
+    // If version == 0, SampleCompositionTimeOffset is uint32_t;
+    // Otherwise, SampleCompositionTimeOffset is int32_t.
+    // Sample.compositionOffset is defined as int32_t.
+    uint8_t version = flags >> 24;
+    flags &= 0xffffff;
+    ALOGV("fragment run version: 0x%02x, flags: 0x%06x", version, flags);
+
+    if ((flags & kFirstSampleFlagsPresent) && (flags & kSampleFlagsPresent)) {
+        // These two shall not be used together.
+        return -EINVAL;
+    }
+
+    uint32_t sampleCount;
+    if (!mDataSource->getUInt32(offset + 4, &sampleCount)) {
+        return ERROR_MALFORMED;
+    }
+    offset += 8;
+    size -= 8;
+
+    uint64_t dataOffset = mTrackFragmentHeaderInfo.mDataOffset;
+
+    uint32_t firstSampleFlags = 0;
+
+    if (flags & kDataOffsetPresent) {
+        if (size < 4) {
+            return -EINVAL;
+        }
+
+        int32_t dataOffsetDelta;
+        if (!mDataSource->getUInt32(offset, (uint32_t*)&dataOffsetDelta)) {
+            return ERROR_MALFORMED;
+        }
+
+        dataOffset = mTrackFragmentHeaderInfo.mBaseDataOffset + dataOffsetDelta;
+
+        offset += 4;
+        size -= 4;
+    }
+
+    if (flags & kFirstSampleFlagsPresent) {
+        if (size < 4) {
+            return -EINVAL;
+        }
+
+        if (!mDataSource->getUInt32(offset, &firstSampleFlags)) {
+            return ERROR_MALFORMED;
+        }
+        offset += 4;
+        size -= 4;
+    }
+
+    uint32_t sampleDuration = 0, sampleSize = 0, sampleFlags = 0,
+             sampleCtsOffset = 0;
+
+    size_t bytesPerSample = 0;
+    if (flags & kSampleDurationPresent) {
+        bytesPerSample += 4;
+    } else if (mTrackFragmentHeaderInfo.mFlags
+            & TrackFragmentHeaderInfo::kDefaultSampleDurationPresent) {
+        sampleDuration = mTrackFragmentHeaderInfo.mDefaultSampleDuration;
+    } else if (mTrex) {
+        sampleDuration = mTrex->default_sample_duration;
+    }
+
+    if (flags & kSampleSizePresent) {
+        bytesPerSample += 4;
+    } else if (mTrackFragmentHeaderInfo.mFlags
+            & TrackFragmentHeaderInfo::kDefaultSampleSizePresent) {
+        sampleSize = mTrackFragmentHeaderInfo.mDefaultSampleSize;
+    } else {
+        sampleSize = mTrackFragmentHeaderInfo.mDefaultSampleSize;
+    }
+
+    if (flags & kSampleFlagsPresent) {
+        bytesPerSample += 4;
+    } else if (mTrackFragmentHeaderInfo.mFlags
+            & TrackFragmentHeaderInfo::kDefaultSampleFlagsPresent) {
+        sampleFlags = mTrackFragmentHeaderInfo.mDefaultSampleFlags;
+    } else {
+        sampleFlags = mTrackFragmentHeaderInfo.mDefaultSampleFlags;
+    }
+
+    if (flags & kSampleCompositionTimeOffsetPresent) {
+        bytesPerSample += 4;
+    } else {
+        sampleCtsOffset = 0;
+    }
+
+    if (size < (off64_t)(sampleCount * bytesPerSample)) {
+        return -EINVAL;
+    }
+
+    Sample tmp;
+    for (uint32_t i = 0; i < sampleCount; ++i) {
+        if (flags & kSampleDurationPresent) {
+            if (!mDataSource->getUInt32(offset, &sampleDuration)) {
+                return ERROR_MALFORMED;
+            }
+            offset += 4;
+        }
+
+        if (flags & kSampleSizePresent) {
+            if (!mDataSource->getUInt32(offset, &sampleSize)) {
+                return ERROR_MALFORMED;
+            }
+            offset += 4;
+        }
+
+        if (flags & kSampleFlagsPresent) {
+            if (!mDataSource->getUInt32(offset, &sampleFlags)) {
+                return ERROR_MALFORMED;
+            }
+            offset += 4;
+        }
+
+        if (flags & kSampleCompositionTimeOffsetPresent) {
+            if (!mDataSource->getUInt32(offset, &sampleCtsOffset)) {
+                return ERROR_MALFORMED;
+            }
+            offset += 4;
+        }
+
+        ALOGV("adding sample %d at offset 0x%08" PRIx64 ", size %u, duration %u, "
+              " flags 0x%08x", i + 1,
+                dataOffset, sampleSize, sampleDuration,
+                (flags & kFirstSampleFlagsPresent) && i == 0
+                    ? firstSampleFlags : sampleFlags);
+        tmp.offset = dataOffset;
+        tmp.size = sampleSize;
+        tmp.duration = sampleDuration;
+        tmp.compositionOffset = sampleCtsOffset;
+        memset(tmp.iv, 0, sizeof(tmp.iv));
+        mCurrentSamples.add(tmp);
+
+        dataOffset += sampleSize;
+    }
+
+    mTrackFragmentHeaderInfo.mDataOffset = dataOffset;
+
+    return OK;
+}
+
+status_t MPEG4Source::getFormat(MetaDataBase &meta) {
+    Mutex::Autolock autoLock(mLock);
+    meta = mFormat;
+    return OK;
+}
+
+size_t MPEG4Source::parseNALSize(const uint8_t *data) const {
+    switch (mNALLengthSize) {
+        case 1:
+            return *data;
+        case 2:
+            return U16_AT(data);
+        case 3:
+            return ((size_t)data[0] << 16) | U16_AT(&data[1]);
+        case 4:
+            return U32_AT(data);
+    }
+
+    // This cannot happen, mNALLengthSize springs to life by adding 1 to
+    // a 2-bit integer.
+    CHECK(!"Should not be here.");
+
+    return 0;
+}
+
+status_t MPEG4Source::read(
+        MediaBufferBase **out, const ReadOptions *options) {
+    Mutex::Autolock autoLock(mLock);
+
+    CHECK(mStarted);
+
+    if (options != nullptr && options->getNonBlocking() && !mGroup->has_buffers()) {
+        *out = nullptr;
+        return WOULD_BLOCK;
+    }
+
+    if (mFirstMoofOffset > 0) {
+        return fragmentedRead(out, options);
+    }
+
+    *out = NULL;
+
+    int64_t targetSampleTimeUs = -1;
+
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode mode;
+    if (options && options->getSeekTo(&seekTimeUs, &mode)) {
+        if (mIsHeif) {
+            CHECK(mSampleTable == NULL);
+            CHECK(mItemTable != NULL);
+            int32_t imageIndex;
+            if (!mFormat.findInt32(kKeyTrackID, &imageIndex)) {
+                return ERROR_MALFORMED;
+            }
+
+            status_t err;
+            if (seekTimeUs >= 0) {
+                err = mItemTable->findImageItem(imageIndex, &mCurrentSampleIndex);
+            } else {
+                err = mItemTable->findThumbnailItem(imageIndex, &mCurrentSampleIndex);
+            }
+            if (err != OK) {
+                return err;
+            }
+        } else {
+            uint32_t findFlags = 0;
+            switch (mode) {
+                case ReadOptions::SEEK_PREVIOUS_SYNC:
+                    findFlags = SampleTable::kFlagBefore;
+                    break;
+                case ReadOptions::SEEK_NEXT_SYNC:
+                    findFlags = SampleTable::kFlagAfter;
+                    break;
+                case ReadOptions::SEEK_CLOSEST_SYNC:
+                case ReadOptions::SEEK_CLOSEST:
+                    findFlags = SampleTable::kFlagClosest;
+                    break;
+                case ReadOptions::SEEK_FRAME_INDEX:
+                    findFlags = SampleTable::kFlagFrameIndex;
+                    break;
+                default:
+                    CHECK(!"Should not be here.");
+                    break;
+            }
+
+            uint32_t sampleIndex;
+            status_t err = mSampleTable->findSampleAtTime(
+                    seekTimeUs, 1000000, mTimescale,
+                    &sampleIndex, findFlags);
+
+            if (mode == ReadOptions::SEEK_CLOSEST
+                    || mode == ReadOptions::SEEK_FRAME_INDEX) {
+                // We found the closest sample already, now we want the sync
+                // sample preceding it (or the sample itself of course), even
+                // if the subsequent sync sample is closer.
+                findFlags = SampleTable::kFlagBefore;
+            }
+
+            uint32_t syncSampleIndex;
+            if (err == OK) {
+                err = mSampleTable->findSyncSampleNear(
+                        sampleIndex, &syncSampleIndex, findFlags);
+            }
+
+            uint32_t sampleTime;
+            if (err == OK) {
+                err = mSampleTable->getMetaDataForSample(
+                        sampleIndex, NULL, NULL, &sampleTime);
+            }
+
+            if (err != OK) {
+                if (err == ERROR_OUT_OF_RANGE) {
+                    // An attempt to seek past the end of the stream would
+                    // normally cause this ERROR_OUT_OF_RANGE error. Propagating
+                    // this all the way to the MediaPlayer would cause abnormal
+                    // termination. Legacy behaviour appears to be to behave as if
+                    // we had seeked to the end of stream, ending normally.
+                    err = ERROR_END_OF_STREAM;
+                }
+                ALOGV("end of stream");
+                return err;
+            }
+
+            if (mode == ReadOptions::SEEK_CLOSEST
+                || mode == ReadOptions::SEEK_FRAME_INDEX) {
+                targetSampleTimeUs = (sampleTime * 1000000ll) / mTimescale;
+            }
+
+#if 0
+            uint32_t syncSampleTime;
+            CHECK_EQ(OK, mSampleTable->getMetaDataForSample(
+                        syncSampleIndex, NULL, NULL, &syncSampleTime));
+
+            ALOGI("seek to time %lld us => sample at time %lld us, "
+                 "sync sample at time %lld us",
+                 seekTimeUs,
+                 sampleTime * 1000000ll / mTimescale,
+                 syncSampleTime * 1000000ll / mTimescale);
+#endif
+
+            mCurrentSampleIndex = syncSampleIndex;
+        }
+
+        if (mBuffer != NULL) {
+            mBuffer->release();
+            mBuffer = NULL;
+        }
+
+        // fall through
+    }
+
+    off64_t offset = 0;
+    size_t size = 0;
+    uint32_t cts, stts;
+    bool isSyncSample;
+    bool newBuffer = false;
+    if (mBuffer == NULL) {
+        newBuffer = true;
+
+        status_t err;
+        if (!mIsHeif) {
+            err = mSampleTable->getMetaDataForSample(
+                    mCurrentSampleIndex, &offset, &size, &cts, &isSyncSample, &stts);
+        } else {
+            err = mItemTable->getImageOffsetAndSize(
+                    options && options->getSeekTo(&seekTimeUs, &mode) ?
+                            &mCurrentSampleIndex : NULL, &offset, &size);
+
+            cts = stts = 0;
+            isSyncSample = 0;
+            ALOGV("image offset %lld, size %zu", (long long)offset, size);
+        }
+
+        if (err != OK) {
+            return err;
+        }
+
+        err = mGroup->acquire_buffer(&mBuffer);
+
+        if (err != OK) {
+            CHECK(mBuffer == NULL);
+            return err;
+        }
+        if (size > mBuffer->size()) {
+            ALOGE("buffer too small: %zu > %zu", size, mBuffer->size());
+            mBuffer->release();
+            mBuffer = NULL;
+            return ERROR_BUFFER_TOO_SMALL;
+        }
+    }
+
+    if ((!mIsAVC && !mIsHEVC) || mWantsNALFragments) {
+        if (newBuffer) {
+            ssize_t num_bytes_read =
+                mDataSource->readAt(offset, (uint8_t *)mBuffer->data(), size);
+
+            if (num_bytes_read < (ssize_t)size) {
+                mBuffer->release();
+                mBuffer = NULL;
+
+                return ERROR_IO;
+            }
+
+            CHECK(mBuffer != NULL);
+            mBuffer->set_range(0, size);
+            mBuffer->meta_data().clear();
+            mBuffer->meta_data().setInt64(
+                    kKeyTime, ((int64_t)cts * 1000000) / mTimescale);
+            mBuffer->meta_data().setInt64(
+                    kKeyDuration, ((int64_t)stts * 1000000) / mTimescale);
+
+            if (targetSampleTimeUs >= 0) {
+                mBuffer->meta_data().setInt64(
+                        kKeyTargetTime, targetSampleTimeUs);
+            }
+
+            if (isSyncSample) {
+                mBuffer->meta_data().setInt32(kKeyIsSyncFrame, 1);
+            }
+
+            ++mCurrentSampleIndex;
+        }
+
+        if (!mIsAVC && !mIsHEVC) {
+            *out = mBuffer;
+            mBuffer = NULL;
+
+            return OK;
+        }
+
+        // Each NAL unit is split up into its constituent fragments and
+        // each one of them returned in its own buffer.
+
+        CHECK(mBuffer->range_length() >= mNALLengthSize);
+
+        const uint8_t *src =
+            (const uint8_t *)mBuffer->data() + mBuffer->range_offset();
+
+        size_t nal_size = parseNALSize(src);
+        if (mNALLengthSize > SIZE_MAX - nal_size) {
+            ALOGE("b/24441553, b/24445122");
+        }
+        if (mBuffer->range_length() - mNALLengthSize < nal_size) {
+            ALOGE("incomplete NAL unit.");
+
+            mBuffer->release();
+            mBuffer = NULL;
+
+            return ERROR_MALFORMED;
+        }
+
+        MediaBufferBase *clone = mBuffer->clone();
+        CHECK(clone != NULL);
+        clone->set_range(mBuffer->range_offset() + mNALLengthSize, nal_size);
+
+        CHECK(mBuffer != NULL);
+        mBuffer->set_range(
+                mBuffer->range_offset() + mNALLengthSize + nal_size,
+                mBuffer->range_length() - mNALLengthSize - nal_size);
+
+        if (mBuffer->range_length() == 0) {
+            mBuffer->release();
+            mBuffer = NULL;
+        }
+
+        *out = clone;
+
+        return OK;
+    } else {
+        // Whole NAL units are returned but each fragment is prefixed by
+        // the start code (0x00 00 00 01).
+        ssize_t num_bytes_read = 0;
+        int32_t drm = 0;
+        bool usesDRM = (mFormat.findInt32(kKeyIsDRM, &drm) && drm != 0);
+        if (usesDRM) {
+            num_bytes_read =
+                mDataSource->readAt(offset, (uint8_t*)mBuffer->data(), size);
+        } else {
+            num_bytes_read = mDataSource->readAt(offset, mSrcBuffer, size);
+        }
+
+        if (num_bytes_read < (ssize_t)size) {
+            mBuffer->release();
+            mBuffer = NULL;
+
+            return ERROR_IO;
+        }
+
+        if (usesDRM) {
+            CHECK(mBuffer != NULL);
+            mBuffer->set_range(0, size);
+
+        } else {
+            uint8_t *dstData = (uint8_t *)mBuffer->data();
+            size_t srcOffset = 0;
+            size_t dstOffset = 0;
+
+            while (srcOffset < size) {
+                bool isMalFormed = !isInRange((size_t)0u, size, srcOffset, mNALLengthSize);
+                size_t nalLength = 0;
+                if (!isMalFormed) {
+                    nalLength = parseNALSize(&mSrcBuffer[srcOffset]);
+                    srcOffset += mNALLengthSize;
+                    isMalFormed = !isInRange((size_t)0u, size, srcOffset, nalLength);
+                }
+
+                if (isMalFormed) {
+                    ALOGE("Video is malformed");
+                    mBuffer->release();
+                    mBuffer = NULL;
+                    return ERROR_MALFORMED;
+                }
+
+                if (nalLength == 0) {
+                    continue;
+                }
+
+                if (dstOffset > SIZE_MAX - 4 ||
+                        dstOffset + 4 > SIZE_MAX - nalLength ||
+                        dstOffset + 4 + nalLength > mBuffer->size()) {
+                    ALOGE("b/27208621 : %zu %zu", dstOffset, mBuffer->size());
+                    android_errorWriteLog(0x534e4554, "27208621");
+                    mBuffer->release();
+                    mBuffer = NULL;
+                    return ERROR_MALFORMED;
+                }
+
+                dstData[dstOffset++] = 0;
+                dstData[dstOffset++] = 0;
+                dstData[dstOffset++] = 0;
+                dstData[dstOffset++] = 1;
+                memcpy(&dstData[dstOffset], &mSrcBuffer[srcOffset], nalLength);
+                srcOffset += nalLength;
+                dstOffset += nalLength;
+            }
+            CHECK_EQ(srcOffset, size);
+            CHECK(mBuffer != NULL);
+            mBuffer->set_range(0, dstOffset);
+        }
+
+        mBuffer->meta_data().clear();
+        mBuffer->meta_data().setInt64(
+                kKeyTime, ((int64_t)cts * 1000000) / mTimescale);
+        mBuffer->meta_data().setInt64(
+                kKeyDuration, ((int64_t)stts * 1000000) / mTimescale);
+
+        if (targetSampleTimeUs >= 0) {
+            mBuffer->meta_data().setInt64(
+                    kKeyTargetTime, targetSampleTimeUs);
+        }
+
+        if (mIsAVC) {
+            uint32_t layerId = FindAVCLayerId(
+                    (const uint8_t *)mBuffer->data(), mBuffer->range_length());
+            mBuffer->meta_data().setInt32(kKeyTemporalLayerId, layerId);
+        }
+
+        if (isSyncSample) {
+            mBuffer->meta_data().setInt32(kKeyIsSyncFrame, 1);
+        }
+
+        ++mCurrentSampleIndex;
+
+        *out = mBuffer;
+        mBuffer = NULL;
+
+        return OK;
+    }
+}
+
+status_t MPEG4Source::fragmentedRead(
+        MediaBufferBase **out, const ReadOptions *options) {
+
+    ALOGV("MPEG4Source::fragmentedRead");
+
+    CHECK(mStarted);
+
+    *out = NULL;
+
+    int64_t targetSampleTimeUs = -1;
+
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode mode;
+    if (options && options->getSeekTo(&seekTimeUs, &mode)) {
+
+        int numSidxEntries = mSegments.size();
+        if (numSidxEntries != 0) {
+            int64_t totalTime = 0;
+            off64_t totalOffset = mFirstMoofOffset;
+            for (int i = 0; i < numSidxEntries; i++) {
+                const SidxEntry *se = &mSegments[i];
+                if (totalTime + se->mDurationUs > seekTimeUs) {
+                    // The requested time is somewhere in this segment
+                    if ((mode == ReadOptions::SEEK_NEXT_SYNC && seekTimeUs > totalTime) ||
+                        (mode == ReadOptions::SEEK_CLOSEST_SYNC &&
+                        (seekTimeUs - totalTime) > (totalTime + se->mDurationUs - seekTimeUs))) {
+                        // requested next sync, or closest sync and it was closer to the end of
+                        // this segment
+                        totalTime += se->mDurationUs;
+                        totalOffset += se->mSize;
+                    }
+                    break;
+                }
+                totalTime += se->mDurationUs;
+                totalOffset += se->mSize;
+            }
+            mCurrentMoofOffset = totalOffset;
+            mNextMoofOffset = -1;
+            mCurrentSamples.clear();
+            mCurrentSampleIndex = 0;
+            status_t err = parseChunk(&totalOffset);
+            if (err != OK) {
+                return err;
+            }
+            mCurrentTime = totalTime * mTimescale / 1000000ll;
+        } else {
+            // without sidx boxes, we can only seek to 0
+            mCurrentMoofOffset = mFirstMoofOffset;
+            mNextMoofOffset = -1;
+            mCurrentSamples.clear();
+            mCurrentSampleIndex = 0;
+            off64_t tmp = mCurrentMoofOffset;
+            status_t err = parseChunk(&tmp);
+            if (err != OK) {
+                return err;
+            }
+            mCurrentTime = 0;
+        }
+
+        if (mBuffer != NULL) {
+            mBuffer->release();
+            mBuffer = NULL;
+        }
+
+        // fall through
+    }
+
+    off64_t offset = 0;
+    size_t size = 0;
+    uint32_t cts = 0;
+    bool isSyncSample = false;
+    bool newBuffer = false;
+    if (mBuffer == NULL) {
+        newBuffer = true;
+
+        if (mCurrentSampleIndex >= mCurrentSamples.size()) {
+            // move to next fragment if there is one
+            if (mNextMoofOffset <= mCurrentMoofOffset) {
+                return ERROR_END_OF_STREAM;
+            }
+            off64_t nextMoof = mNextMoofOffset;
+            mCurrentMoofOffset = nextMoof;
+            mCurrentSamples.clear();
+            mCurrentSampleIndex = 0;
+            status_t err = parseChunk(&nextMoof);
+            if (err != OK) {
+                return err;
+            }
+            if (mCurrentSampleIndex >= mCurrentSamples.size()) {
+                return ERROR_END_OF_STREAM;
+            }
+        }
+
+        const Sample *smpl = &mCurrentSamples[mCurrentSampleIndex];
+        offset = smpl->offset;
+        size = smpl->size;
+        cts = mCurrentTime + smpl->compositionOffset;
+        mCurrentTime += smpl->duration;
+        isSyncSample = (mCurrentSampleIndex == 0); // XXX
+
+        status_t err = mGroup->acquire_buffer(&mBuffer);
+
+        if (err != OK) {
+            CHECK(mBuffer == NULL);
+            ALOGV("acquire_buffer returned %d", err);
+            return err;
+        }
+        if (size > mBuffer->size()) {
+            ALOGE("buffer too small: %zu > %zu", size, mBuffer->size());
+            mBuffer->release();
+            mBuffer = NULL;
+            return ERROR_BUFFER_TOO_SMALL;
+        }
+    }
+
+    const Sample *smpl = &mCurrentSamples[mCurrentSampleIndex];
+    MetaDataBase &bufmeta = mBuffer->meta_data();
+    bufmeta.clear();
+    if (smpl->encryptedsizes.size()) {
+        // store clear/encrypted lengths in metadata
+        bufmeta.setData(kKeyPlainSizes, 0,
+                smpl->clearsizes.array(), smpl->clearsizes.size() * 4);
+        bufmeta.setData(kKeyEncryptedSizes, 0,
+                smpl->encryptedsizes.array(), smpl->encryptedsizes.size() * 4);
+        bufmeta.setInt32(kKeyCryptoDefaultIVSize, mDefaultIVSize);
+        bufmeta.setInt32(kKeyCryptoMode, mCryptoMode);
+        bufmeta.setData(kKeyCryptoKey, 0, mCryptoKey, 16);
+        bufmeta.setInt32(kKeyEncryptedByteBlock, mDefaultEncryptedByteBlock);
+        bufmeta.setInt32(kKeySkipByteBlock, mDefaultSkipByteBlock);
+
+        uint32_t type = 0;
+        const void *iv = NULL;
+        size_t ivlength = 0;
+        if (!mFormat.findData(
+                kKeyCryptoIV, &type, &iv, &ivlength)) {
+            iv = smpl->iv;
+            ivlength = 16; // use 16 or the actual size?
+        }
+        bufmeta.setData(kKeyCryptoIV, 0, iv, ivlength);
+
+    }
+
+    if ((!mIsAVC && !mIsHEVC)|| mWantsNALFragments) {
+        if (newBuffer) {
+            if (!isInRange((size_t)0u, mBuffer->size(), size)) {
+                mBuffer->release();
+                mBuffer = NULL;
+
+                ALOGE("fragmentedRead ERROR_MALFORMED size %zu", size);
+                return ERROR_MALFORMED;
+            }
+
+            ssize_t num_bytes_read =
+                mDataSource->readAt(offset, (uint8_t *)mBuffer->data(), size);
+
+            if (num_bytes_read < (ssize_t)size) {
+                mBuffer->release();
+                mBuffer = NULL;
+
+                ALOGE("i/o error");
+                return ERROR_IO;
+            }
+
+            CHECK(mBuffer != NULL);
+            mBuffer->set_range(0, size);
+            mBuffer->meta_data().setInt64(
+                    kKeyTime, ((int64_t)cts * 1000000) / mTimescale);
+            mBuffer->meta_data().setInt64(
+                    kKeyDuration, ((int64_t)smpl->duration * 1000000) / mTimescale);
+
+            if (targetSampleTimeUs >= 0) {
+                mBuffer->meta_data().setInt64(
+                        kKeyTargetTime, targetSampleTimeUs);
+            }
+
+            if (mIsAVC) {
+                uint32_t layerId = FindAVCLayerId(
+                        (const uint8_t *)mBuffer->data(), mBuffer->range_length());
+                mBuffer->meta_data().setInt32(kKeyTemporalLayerId, layerId);
+            }
+
+            if (isSyncSample) {
+                mBuffer->meta_data().setInt32(kKeyIsSyncFrame, 1);
+            }
+
+            ++mCurrentSampleIndex;
+        }
+
+        if (!mIsAVC && !mIsHEVC) {
+            *out = mBuffer;
+            mBuffer = NULL;
+
+            return OK;
+        }
+
+        // Each NAL unit is split up into its constituent fragments and
+        // each one of them returned in its own buffer.
+
+        CHECK(mBuffer->range_length() >= mNALLengthSize);
+
+        const uint8_t *src =
+            (const uint8_t *)mBuffer->data() + mBuffer->range_offset();
+
+        size_t nal_size = parseNALSize(src);
+        if (mNALLengthSize > SIZE_MAX - nal_size) {
+            ALOGE("b/24441553, b/24445122");
+        }
+
+        if (mBuffer->range_length() - mNALLengthSize < nal_size) {
+            ALOGE("incomplete NAL unit.");
+
+            mBuffer->release();
+            mBuffer = NULL;
+
+            return ERROR_MALFORMED;
+        }
+
+        MediaBufferBase *clone = mBuffer->clone();
+        CHECK(clone != NULL);
+        clone->set_range(mBuffer->range_offset() + mNALLengthSize, nal_size);
+
+        CHECK(mBuffer != NULL);
+        mBuffer->set_range(
+                mBuffer->range_offset() + mNALLengthSize + nal_size,
+                mBuffer->range_length() - mNALLengthSize - nal_size);
+
+        if (mBuffer->range_length() == 0) {
+            mBuffer->release();
+            mBuffer = NULL;
+        }
+
+        *out = clone;
+
+        return OK;
+    } else {
+        ALOGV("whole NAL");
+        // Whole NAL units are returned but each fragment is prefixed by
+        // the start code (0x00 00 00 01).
+        ssize_t num_bytes_read = 0;
+        int32_t drm = 0;
+        bool usesDRM = (mFormat.findInt32(kKeyIsDRM, &drm) && drm != 0);
+        void *data = NULL;
+        bool isMalFormed = false;
+        if (usesDRM) {
+            if (mBuffer == NULL || !isInRange((size_t)0u, mBuffer->size(), size)) {
+                isMalFormed = true;
+            } else {
+                data = mBuffer->data();
+            }
+        } else {
+            int32_t max_size;
+            if (!mFormat.findInt32(kKeyMaxInputSize, &max_size)
+                    || !isInRange((size_t)0u, (size_t)max_size, size)) {
+                isMalFormed = true;
+            } else {
+                data = mSrcBuffer;
+            }
+        }
+
+        if (isMalFormed || data == NULL) {
+            ALOGE("isMalFormed size %zu", size);
+            if (mBuffer != NULL) {
+                mBuffer->release();
+                mBuffer = NULL;
+            }
+            return ERROR_MALFORMED;
+        }
+        num_bytes_read = mDataSource->readAt(offset, data, size);
+
+        if (num_bytes_read < (ssize_t)size) {
+            mBuffer->release();
+            mBuffer = NULL;
+
+            ALOGE("i/o error");
+            return ERROR_IO;
+        }
+
+        if (usesDRM) {
+            CHECK(mBuffer != NULL);
+            mBuffer->set_range(0, size);
+
+        } else {
+            uint8_t *dstData = (uint8_t *)mBuffer->data();
+            size_t srcOffset = 0;
+            size_t dstOffset = 0;
+
+            while (srcOffset < size) {
+                isMalFormed = !isInRange((size_t)0u, size, srcOffset, mNALLengthSize);
+                size_t nalLength = 0;
+                if (!isMalFormed) {
+                    nalLength = parseNALSize(&mSrcBuffer[srcOffset]);
+                    srcOffset += mNALLengthSize;
+                    isMalFormed = !isInRange((size_t)0u, size, srcOffset, nalLength)
+                            || !isInRange((size_t)0u, mBuffer->size(), dstOffset, (size_t)4u)
+                            || !isInRange((size_t)0u, mBuffer->size(), dstOffset + 4, nalLength);
+                }
+
+                if (isMalFormed) {
+                    ALOGE("Video is malformed; nalLength %zu", nalLength);
+                    mBuffer->release();
+                    mBuffer = NULL;
+                    return ERROR_MALFORMED;
+                }
+
+                if (nalLength == 0) {
+                    continue;
+                }
+
+                if (dstOffset > SIZE_MAX - 4 ||
+                        dstOffset + 4 > SIZE_MAX - nalLength ||
+                        dstOffset + 4 + nalLength > mBuffer->size()) {
+                    ALOGE("b/26365349 : %zu %zu", dstOffset, mBuffer->size());
+                    android_errorWriteLog(0x534e4554, "26365349");
+                    mBuffer->release();
+                    mBuffer = NULL;
+                    return ERROR_MALFORMED;
+                }
+
+                dstData[dstOffset++] = 0;
+                dstData[dstOffset++] = 0;
+                dstData[dstOffset++] = 0;
+                dstData[dstOffset++] = 1;
+                memcpy(&dstData[dstOffset], &mSrcBuffer[srcOffset], nalLength);
+                srcOffset += nalLength;
+                dstOffset += nalLength;
+            }
+            CHECK_EQ(srcOffset, size);
+            CHECK(mBuffer != NULL);
+            mBuffer->set_range(0, dstOffset);
+        }
+
+        mBuffer->meta_data().setInt64(
+                kKeyTime, ((int64_t)cts * 1000000) / mTimescale);
+        mBuffer->meta_data().setInt64(
+                kKeyDuration, ((int64_t)smpl->duration * 1000000) / mTimescale);
+
+        if (targetSampleTimeUs >= 0) {
+            mBuffer->meta_data().setInt64(
+                    kKeyTargetTime, targetSampleTimeUs);
+        }
+
+        if (isSyncSample) {
+            mBuffer->meta_data().setInt32(kKeyIsSyncFrame, 1);
+        }
+
+        ++mCurrentSampleIndex;
+
+        *out = mBuffer;
+        mBuffer = NULL;
+
+        return OK;
+    }
+}
+
+MPEG4Extractor::Track *MPEG4Extractor::findTrackByMimePrefix(
+        const char *mimePrefix) {
+    for (Track *track = mFirstTrack; track != NULL; track = track->next) {
+        const char *mime;
+        if (track->meta.findCString(kKeyMIMEType, &mime)
+                && !strncasecmp(mime, mimePrefix, strlen(mimePrefix))) {
+            return track;
+        }
+    }
+
+    return NULL;
+}
+
+static bool LegacySniffMPEG4(DataSourceBase *source, float *confidence) {
+    uint8_t header[8];
+
+    ssize_t n = source->readAt(4, header, sizeof(header));
+    if (n < (ssize_t)sizeof(header)) {
+        return false;
+    }
+
+    if (!memcmp(header, "ftyp3gp", 7) || !memcmp(header, "ftypmp42", 8)
+        || !memcmp(header, "ftyp3gr6", 8) || !memcmp(header, "ftyp3gs6", 8)
+        || !memcmp(header, "ftyp3ge6", 8) || !memcmp(header, "ftyp3gg6", 8)
+        || !memcmp(header, "ftypisom", 8) || !memcmp(header, "ftypM4V ", 8)
+        || !memcmp(header, "ftypM4A ", 8) || !memcmp(header, "ftypf4v ", 8)
+        || !memcmp(header, "ftypkddi", 8) || !memcmp(header, "ftypM4VP", 8)
+        || !memcmp(header, "ftypmif1", 8) || !memcmp(header, "ftypheic", 8)
+        || !memcmp(header, "ftypmsf1", 8) || !memcmp(header, "ftyphevc", 8)) {
+        *confidence = 0.4;
+
+        return true;
+    }
+
+    return false;
+}
+
+static bool isCompatibleBrand(uint32_t fourcc) {
+    static const uint32_t kCompatibleBrands[] = {
+        FOURCC('i', 's', 'o', 'm'),
+        FOURCC('i', 's', 'o', '2'),
+        FOURCC('a', 'v', 'c', '1'),
+        FOURCC('h', 'v', 'c', '1'),
+        FOURCC('h', 'e', 'v', '1'),
+        FOURCC('3', 'g', 'p', '4'),
+        FOURCC('m', 'p', '4', '1'),
+        FOURCC('m', 'p', '4', '2'),
+        FOURCC('d', 'a', 's', 'h'),
+
+        // Won't promise that the following file types can be played.
+        // Just give these file types a chance.
+        FOURCC('q', 't', ' ', ' '),  // Apple's QuickTime
+        FOURCC('M', 'S', 'N', 'V'),  // Sony's PSP
+
+        FOURCC('3', 'g', '2', 'a'),  // 3GPP2
+        FOURCC('3', 'g', '2', 'b'),
+        FOURCC('m', 'i', 'f', '1'),  // HEIF image
+        FOURCC('h', 'e', 'i', 'c'),  // HEIF image
+        FOURCC('m', 's', 'f', '1'),  // HEIF image sequence
+        FOURCC('h', 'e', 'v', 'c'),  // HEIF image sequence
+    };
+
+    for (size_t i = 0;
+         i < sizeof(kCompatibleBrands) / sizeof(kCompatibleBrands[0]);
+         ++i) {
+        if (kCompatibleBrands[i] == fourcc) {
+            return true;
+        }
+    }
+
+    return false;
+}
+
+// Attempt to actually parse the 'ftyp' atom and determine if a suitable
+// compatible brand is present.
+// Also try to identify where this file's metadata ends
+// (end of the 'moov' atom) and report it to the caller as part of
+// the metadata.
+static bool BetterSniffMPEG4(DataSourceBase *source, float *confidence) {
+    // We scan up to 128 bytes to identify this file as an MP4.
+    static const off64_t kMaxScanOffset = 128ll;
+
+    off64_t offset = 0ll;
+    bool foundGoodFileType = false;
+    off64_t moovAtomEndOffset = -1ll;
+    bool done = false;
+
+    while (!done && offset < kMaxScanOffset) {
+        uint32_t hdr[2];
+        if (source->readAt(offset, hdr, 8) < 8) {
+            return false;
+        }
+
+        uint64_t chunkSize = ntohl(hdr[0]);
+        uint32_t chunkType = ntohl(hdr[1]);
+        off64_t chunkDataOffset = offset + 8;
+
+        if (chunkSize == 1) {
+            if (source->readAt(offset + 8, &chunkSize, 8) < 8) {
+                return false;
+            }
+
+            chunkSize = ntoh64(chunkSize);
+            chunkDataOffset += 8;
+
+            if (chunkSize < 16) {
+                // The smallest valid chunk is 16 bytes long in this case.
+                return false;
+            }
+
+        } else if (chunkSize < 8) {
+            // The smallest valid chunk is 8 bytes long.
+            return false;
+        }
+
+        // (data_offset - offset) is either 8 or 16
+        off64_t chunkDataSize = chunkSize - (chunkDataOffset - offset);
+        if (chunkDataSize < 0) {
+            ALOGE("b/23540914");
+            return false;
+        }
+
+        char chunkstring[5];
+        MakeFourCCString(chunkType, chunkstring);
+        ALOGV("saw chunk type %s, size %" PRIu64 " @ %lld", chunkstring, chunkSize, (long long)offset);
+        switch (chunkType) {
+            case FOURCC('f', 't', 'y', 'p'):
+            {
+                if (chunkDataSize < 8) {
+                    return false;
+                }
+
+                uint32_t numCompatibleBrands = (chunkDataSize - 8) / 4;
+                for (size_t i = 0; i < numCompatibleBrands + 2; ++i) {
+                    if (i == 1) {
+                        // Skip this index, it refers to the minorVersion,
+                        // not a brand.
+                        continue;
+                    }
+
+                    uint32_t brand;
+                    if (source->readAt(
+                                chunkDataOffset + 4 * i, &brand, 4) < 4) {
+                        return false;
+                    }
+
+                    brand = ntohl(brand);
+
+                    if (isCompatibleBrand(brand)) {
+                        foundGoodFileType = true;
+                        break;
+                    }
+                }
+
+                if (!foundGoodFileType) {
+                    return false;
+                }
+
+                break;
+            }
+
+            case FOURCC('m', 'o', 'o', 'v'):
+            {
+                moovAtomEndOffset = offset + chunkSize;
+
+                done = true;
+                break;
+            }
+
+            default:
+                break;
+        }
+
+        offset += chunkSize;
+    }
+
+    if (!foundGoodFileType) {
+        return false;
+    }
+
+    *confidence = 0.4f;
+
+    return true;
+}
+
+static MediaExtractor* CreateExtractor(DataSourceBase *source, void *) {
+    return new MPEG4Extractor(source);
+}
+
+static MediaExtractor::CreatorFunc Sniff(
+        DataSourceBase *source, float *confidence, void **,
+        MediaExtractor::FreeMetaFunc *) {
+    if (BetterSniffMPEG4(source, confidence)) {
+        return CreateExtractor;
+    }
+
+    if (LegacySniffMPEG4(source, confidence)) {
+        ALOGW("Identified supported mpeg4 through LegacySniffMPEG4.");
+        return CreateExtractor;
+    }
+
+    return NULL;
+}
+
+extern "C" {
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
+MediaExtractor::ExtractorDef GETEXTRACTORDEF() {
+    return {
+        MediaExtractor::EXTRACTORDEF_VERSION,
+        UUID("27575c67-4417-4c54-8d3d-8e626985a164"),
+        1, // version
+        "MP4 Extractor",
+        Sniff
+    };
+}
+
+} // extern "C"
+
+}  // namespace android

diff --git a/media/extractors/mp4/MPEG4Extractor.h b/media/extractors/mp4/MPEG4Extractor.h
new file mode 100644
index 0000000..3ea0963
--- /dev/null
+++ b/media/extractors/mp4/MPEG4Extractor.h

@@ -0,0 +1,155 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MPEG4_EXTRACTOR_H_
+
+#define MPEG4_EXTRACTOR_H_
+
+#include <arpa/inet.h>
+
+#include <media/DataSourceBase.h>
+#include <media/MediaExtractor.h>
+#include <media/stagefright/MetaDataBase.h>
+#include <media/stagefright/foundation/AString.h>
+#include <utils/KeyedVector.h>
+#include <utils/List.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+
+namespace android {
+struct AMessage;
+class DataSourceBase;
+struct CachedRangedDataSource;
+class SampleTable;
+class String8;
+namespace heif {
+class ItemTable;
+}
+using heif::ItemTable;
+
+struct SidxEntry {
+    size_t mSize;
+    uint32_t mDurationUs;
+};
+
+struct Trex {
+    uint32_t track_ID;
+    uint32_t default_sample_description_index;
+    uint32_t default_sample_duration;
+    uint32_t default_sample_size;
+    uint32_t default_sample_flags;
+};
+
+class MPEG4Extractor : public MediaExtractor {
+public:
+    explicit MPEG4Extractor(DataSourceBase *source, const char *mime = NULL);
+
+    virtual size_t countTracks();
+    virtual MediaTrack *getTrack(size_t index);
+    virtual status_t getTrackMetaData(MetaDataBase& meta, size_t index, uint32_t flags);
+
+    virtual status_t getMetaData(MetaDataBase& meta);
+    virtual uint32_t flags() const;
+    virtual const char * name() { return "MPEG4Extractor"; }
+
+protected:
+    virtual ~MPEG4Extractor();
+
+private:
+
+    struct PsshInfo {
+        uint8_t uuid[16];
+        uint32_t datalen;
+        uint8_t *data;
+    };
+    struct Track {
+        Track *next;
+        MetaDataBase meta;
+        uint32_t timescale;
+        sp<SampleTable> sampleTable;
+        bool includes_expensive_metadata;
+        bool skipTrack;
+        bool has_elst;
+        int64_t elst_media_time;
+        uint64_t elst_segment_duration;
+        bool subsample_encryption;
+    };
+
+    Vector<SidxEntry> mSidxEntries;
+    off64_t mMoofOffset;
+    bool mMoofFound;
+    bool mMdatFound;
+
+    Vector<PsshInfo> mPssh;
+
+    Vector<Trex> mTrex;
+
+    DataSourceBase *mDataSource;
+    CachedRangedDataSource *mCachedSource;
+    status_t mInitCheck;
+    uint32_t mHeaderTimescale;
+    bool mIsQT;
+    bool mIsHeif;
+    bool mHasMoovBox;
+    bool mPreferHeif;
+
+    Track *mFirstTrack, *mLastTrack;
+
+    MetaDataBase mFileMetaData;
+
+    Vector<uint32_t> mPath;
+    String8 mLastCommentMean;
+    String8 mLastCommentName;
+    String8 mLastCommentData;
+
+    KeyedVector<uint32_t, AString> mMetaKeyMap;
+
+    status_t readMetaData();
+    status_t parseChunk(off64_t *offset, int depth);
+    status_t parseITunesMetaData(off64_t offset, size_t size);
+    status_t parseColorInfo(off64_t offset, size_t size);
+    status_t parse3GPPMetaData(off64_t offset, size_t size, int depth);
+    void parseID3v2MetaData(off64_t offset);
+    status_t parseQTMetaKey(off64_t data_offset, size_t data_size);
+    status_t parseQTMetaVal(int32_t keyId, off64_t data_offset, size_t data_size);
+
+    status_t updateAudioTrackInfoFromESDS_MPEG4Audio(
+            const void *esds_data, size_t esds_size);
+
+    static status_t verifyTrack(Track *track);
+
+    sp<ItemTable> mItemTable;
+
+    status_t parseTrackHeader(off64_t data_offset, off64_t data_size);
+
+    status_t parseSegmentIndex(off64_t data_offset, size_t data_size);
+
+    Track *findTrackByMimePrefix(const char *mimePrefix);
+
+    status_t parseAC3SampleEntry(off64_t offset);
+    status_t parseAC3SpecificBox(off64_t offset, uint16_t sampleRate);
+
+    MPEG4Extractor(const MPEG4Extractor &);
+    MPEG4Extractor &operator=(const MPEG4Extractor &);
+};
+
+bool SniffMPEG4(
+        DataSourceBase *source, String8 *mimeType, float *confidence,
+        sp<AMessage> *);
+
+}  // namespace android
+
+#endif  // MPEG4_EXTRACTOR_H_

diff --git a/media/libstagefright/matroska/NOTICE b/media/extractors/mp4/NOTICE
similarity index 100%
copy from media/libstagefright/matroska/NOTICE
copy to media/extractors/mp4/NOTICE


diff --git a/media/extractors/mp4/SampleIterator.cpp b/media/extractors/mp4/SampleIterator.cpp
new file mode 100644
index 0000000..93ee7c6
--- /dev/null
+++ b/media/extractors/mp4/SampleIterator.cpp

@@ -0,0 +1,352 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "SampleIterator"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include "SampleIterator.h"
+
+#include <arpa/inet.h>
+
+#include <media/DataSourceBase.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ByteUtils.h>
+
+#include "SampleTable.h"
+
+namespace android {
+
+SampleIterator::SampleIterator(SampleTable *table)
+    : mTable(table),
+      mInitialized(false),
+      mTimeToSampleIndex(0),
+      mTTSSampleIndex(0),
+      mTTSSampleTime(0),
+      mTTSCount(0),
+      mTTSDuration(0) {
+    reset();
+}
+
+void SampleIterator::reset() {
+    mSampleToChunkIndex = 0;
+    mFirstChunk = 0;
+    mFirstChunkSampleIndex = 0;
+    mStopChunk = 0;
+    mStopChunkSampleIndex = 0;
+    mSamplesPerChunk = 0;
+    mChunkDesc = 0;
+}
+
+status_t SampleIterator::seekTo(uint32_t sampleIndex) {
+    ALOGV("seekTo(%d)", sampleIndex);
+
+    if (sampleIndex >= mTable->mNumSampleSizes) {
+        return ERROR_END_OF_STREAM;
+    }
+
+    if (mTable->mSampleToChunkOffset < 0
+            || mTable->mChunkOffsetOffset < 0
+            || mTable->mSampleSizeOffset < 0
+            || mTable->mTimeToSampleCount == 0) {
+
+        return ERROR_MALFORMED;
+    }
+
+    if (mInitialized && mCurrentSampleIndex == sampleIndex) {
+        return OK;
+    }
+
+    if (!mInitialized || sampleIndex < mFirstChunkSampleIndex) {
+        reset();
+    }
+
+    if (sampleIndex >= mStopChunkSampleIndex) {
+        status_t err;
+        if ((err = findChunkRange(sampleIndex)) != OK) {
+            ALOGE("findChunkRange failed");
+            return err;
+        }
+    }
+
+    CHECK(sampleIndex < mStopChunkSampleIndex);
+
+    if (mSamplesPerChunk == 0) {
+        ALOGE("b/22802344");
+        return ERROR_MALFORMED;
+    }
+
+    uint32_t chunk =
+        (sampleIndex - mFirstChunkSampleIndex) / mSamplesPerChunk
+        + mFirstChunk;
+
+    if (!mInitialized || chunk != mCurrentChunkIndex) {
+        status_t err;
+        if ((err = getChunkOffset(chunk, &mCurrentChunkOffset)) != OK) {
+            ALOGE("getChunkOffset return error");
+            return err;
+        }
+
+        mCurrentChunkSampleSizes.clear();
+
+        uint32_t firstChunkSampleIndex =
+            mFirstChunkSampleIndex
+                + mSamplesPerChunk * (chunk - mFirstChunk);
+
+        for (uint32_t i = 0; i < mSamplesPerChunk; ++i) {
+            size_t sampleSize;
+            if ((err = getSampleSizeDirect(
+                            firstChunkSampleIndex + i, &sampleSize)) != OK) {
+                ALOGE("getSampleSizeDirect return error");
+                mCurrentChunkSampleSizes.clear();
+                return err;
+            }
+
+            mCurrentChunkSampleSizes.push(sampleSize);
+        }
+
+        mCurrentChunkIndex = chunk;
+    }
+
+    uint32_t chunkRelativeSampleIndex =
+        (sampleIndex - mFirstChunkSampleIndex) % mSamplesPerChunk;
+
+    mCurrentSampleOffset = mCurrentChunkOffset;
+    for (uint32_t i = 0; i < chunkRelativeSampleIndex; ++i) {
+        mCurrentSampleOffset += mCurrentChunkSampleSizes[i];
+    }
+
+    mCurrentSampleSize = mCurrentChunkSampleSizes[chunkRelativeSampleIndex];
+    if (sampleIndex < mTTSSampleIndex) {
+        mTimeToSampleIndex = 0;
+        mTTSSampleIndex = 0;
+        mTTSSampleTime = 0;
+        mTTSCount = 0;
+        mTTSDuration = 0;
+    }
+
+    status_t err;
+    if ((err = findSampleTimeAndDuration(
+            sampleIndex, &mCurrentSampleTime, &mCurrentSampleDuration)) != OK) {
+        ALOGE("findSampleTime return error");
+        return err;
+    }
+
+    mCurrentSampleIndex = sampleIndex;
+
+    mInitialized = true;
+
+    return OK;
+}
+
+status_t SampleIterator::findChunkRange(uint32_t sampleIndex) {
+    CHECK(sampleIndex >= mFirstChunkSampleIndex);
+
+    while (sampleIndex >= mStopChunkSampleIndex) {
+        if (mSampleToChunkIndex == mTable->mNumSampleToChunkOffsets) {
+            return ERROR_OUT_OF_RANGE;
+        }
+
+        mFirstChunkSampleIndex = mStopChunkSampleIndex;
+
+        const SampleTable::SampleToChunkEntry *entry =
+            &mTable->mSampleToChunkEntries[mSampleToChunkIndex];
+
+        mFirstChunk = entry->startChunk;
+        mSamplesPerChunk = entry->samplesPerChunk;
+        mChunkDesc = entry->chunkDesc;
+
+        if (mSampleToChunkIndex + 1 < mTable->mNumSampleToChunkOffsets) {
+            mStopChunk = entry[1].startChunk;
+
+            if (mSamplesPerChunk == 0 || mStopChunk < mFirstChunk ||
+                (mStopChunk - mFirstChunk) > UINT32_MAX / mSamplesPerChunk ||
+                ((mStopChunk - mFirstChunk) * mSamplesPerChunk >
+                 UINT32_MAX - mFirstChunkSampleIndex)) {
+
+                return ERROR_OUT_OF_RANGE;
+            }
+            mStopChunkSampleIndex =
+                mFirstChunkSampleIndex
+                    + (mStopChunk - mFirstChunk) * mSamplesPerChunk;
+        } else {
+            mStopChunk = 0xffffffff;
+            mStopChunkSampleIndex = 0xffffffff;
+        }
+
+        ++mSampleToChunkIndex;
+    }
+
+    return OK;
+}
+
+status_t SampleIterator::getChunkOffset(uint32_t chunk, off64_t *offset) {
+    *offset = 0;
+
+    if (chunk >= mTable->mNumChunkOffsets) {
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    if (mTable->mChunkOffsetType == SampleTable::kChunkOffsetType32) {
+        uint32_t offset32;
+
+        if (mTable->mDataSource->readAt(
+                    mTable->mChunkOffsetOffset + 8 + 4 * chunk,
+                    &offset32,
+                    sizeof(offset32)) < (ssize_t)sizeof(offset32)) {
+            return ERROR_IO;
+        }
+
+        *offset = ntohl(offset32);
+    } else {
+        CHECK_EQ(mTable->mChunkOffsetType, SampleTable::kChunkOffsetType64);
+
+        uint64_t offset64;
+        if (mTable->mDataSource->readAt(
+                    mTable->mChunkOffsetOffset + 8 + 8 * chunk,
+                    &offset64,
+                    sizeof(offset64)) < (ssize_t)sizeof(offset64)) {
+            return ERROR_IO;
+        }
+
+        *offset = ntoh64(offset64);
+    }
+
+    return OK;
+}
+
+status_t SampleIterator::getSampleSizeDirect(
+        uint32_t sampleIndex, size_t *size) {
+    *size = 0;
+
+    if (sampleIndex >= mTable->mNumSampleSizes) {
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    if (mTable->mDefaultSampleSize > 0) {
+        *size = mTable->mDefaultSampleSize;
+        return OK;
+    }
+
+    switch (mTable->mSampleSizeFieldSize) {
+        case 32:
+        {
+            uint32_t x;
+            if (mTable->mDataSource->readAt(
+                        mTable->mSampleSizeOffset + 12 + 4 * sampleIndex,
+                        &x, sizeof(x)) < (ssize_t)sizeof(x)) {
+                return ERROR_IO;
+            }
+
+            *size = ntohl(x);
+            break;
+        }
+
+        case 16:
+        {
+            uint16_t x;
+            if (mTable->mDataSource->readAt(
+                        mTable->mSampleSizeOffset + 12 + 2 * sampleIndex,
+                        &x, sizeof(x)) < (ssize_t)sizeof(x)) {
+                return ERROR_IO;
+            }
+
+            *size = ntohs(x);
+            break;
+        }
+
+        case 8:
+        {
+            uint8_t x;
+            if (mTable->mDataSource->readAt(
+                        mTable->mSampleSizeOffset + 12 + sampleIndex,
+                        &x, sizeof(x)) < (ssize_t)sizeof(x)) {
+                return ERROR_IO;
+            }
+
+            *size = x;
+            break;
+        }
+
+        default:
+        {
+            CHECK_EQ(mTable->mSampleSizeFieldSize, 4u);
+
+            uint8_t x;
+            if (mTable->mDataSource->readAt(
+                        mTable->mSampleSizeOffset + 12 + sampleIndex / 2,
+                        &x, sizeof(x)) < (ssize_t)sizeof(x)) {
+                return ERROR_IO;
+            }
+
+            *size = (sampleIndex & 1) ? x & 0x0f : x >> 4;
+            break;
+        }
+    }
+
+    return OK;
+}
+
+status_t SampleIterator::findSampleTimeAndDuration(
+        uint32_t sampleIndex, uint32_t *time, uint32_t *duration) {
+    if (sampleIndex >= mTable->mNumSampleSizes) {
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    while (true) {
+        if (mTTSSampleIndex > UINT32_MAX - mTTSCount) {
+            return ERROR_OUT_OF_RANGE;
+        }
+        if(sampleIndex < mTTSSampleIndex + mTTSCount) {
+            break;
+        }
+        if (mTimeToSampleIndex == mTable->mTimeToSampleCount ||
+            (mTTSDuration != 0 && mTTSCount > UINT32_MAX / mTTSDuration) ||
+            mTTSSampleTime > UINT32_MAX - (mTTSCount * mTTSDuration)) {
+            return ERROR_OUT_OF_RANGE;
+        }
+
+        mTTSSampleIndex += mTTSCount;
+        mTTSSampleTime += mTTSCount * mTTSDuration;
+
+        mTTSCount = mTable->mTimeToSample[2 * mTimeToSampleIndex];
+        mTTSDuration = mTable->mTimeToSample[2 * mTimeToSampleIndex + 1];
+
+        ++mTimeToSampleIndex;
+    }
+
+    *time = mTTSSampleTime + mTTSDuration * (sampleIndex - mTTSSampleIndex);
+
+    int32_t offset = mTable->getCompositionTimeOffset(sampleIndex);
+    if ((offset < 0 && *time < (offset == INT32_MIN ?
+            INT32_MAX : uint32_t(-offset))) ||
+            (offset > 0 && *time > UINT32_MAX - offset)) {
+        ALOGE("%u + %d would overflow", *time, offset);
+        return ERROR_OUT_OF_RANGE;
+    }
+    if (offset > 0) {
+        *time += offset;
+    } else {
+        *time -= (offset == INT32_MIN ? INT32_MAX : (-offset));
+    }
+
+    *duration = mTTSDuration;
+
+    return OK;
+}
+
+}  // namespace android
+

diff --git a/media/libstagefright/include/SampleIterator.h b/media/extractors/mp4/SampleIterator.h
similarity index 100%
rename from media/libstagefright/include/SampleIterator.h
rename to media/extractors/mp4/SampleIterator.h


diff --git a/media/extractors/mp4/SampleTable.cpp b/media/extractors/mp4/SampleTable.cpp
new file mode 100644
index 0000000..81c353e
--- /dev/null
+++ b/media/extractors/mp4/SampleTable.cpp

@@ -0,0 +1,1008 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "SampleTable"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <limits>
+
+#include "SampleTable.h"
+#include "SampleIterator.h"
+
+#include <arpa/inet.h>
+
+#include <media/DataSourceBase.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ByteUtils.h>
+
+/* TODO: remove after being merged into other branches */
+#ifndef UINT32_MAX
+#define UINT32_MAX       (4294967295U)
+#endif
+
+namespace android {
+
+// static
+const uint32_t SampleTable::kChunkOffsetType32 = FOURCC('s', 't', 'c', 'o');
+// static
+const uint32_t SampleTable::kChunkOffsetType64 = FOURCC('c', 'o', '6', '4');
+// static
+const uint32_t SampleTable::kSampleSizeType32 = FOURCC('s', 't', 's', 'z');
+// static
+const uint32_t SampleTable::kSampleSizeTypeCompact = FOURCC('s', 't', 'z', '2');
+
+////////////////////////////////////////////////////////////////////////////////
+
+const off64_t kMaxOffset = std::numeric_limits<off64_t>::max();
+
+struct SampleTable::CompositionDeltaLookup {
+    CompositionDeltaLookup();
+
+    void setEntries(
+            const int32_t *deltaEntries, size_t numDeltaEntries);
+
+    int32_t getCompositionTimeOffset(uint32_t sampleIndex);
+
+private:
+    Mutex mLock;
+
+    const int32_t *mDeltaEntries;
+    size_t mNumDeltaEntries;
+
+    size_t mCurrentDeltaEntry;
+    size_t mCurrentEntrySampleIndex;
+
+    DISALLOW_EVIL_CONSTRUCTORS(CompositionDeltaLookup);
+};
+
+SampleTable::CompositionDeltaLookup::CompositionDeltaLookup()
+    : mDeltaEntries(NULL),
+      mNumDeltaEntries(0),
+      mCurrentDeltaEntry(0),
+      mCurrentEntrySampleIndex(0) {
+}
+
+void SampleTable::CompositionDeltaLookup::setEntries(
+        const int32_t *deltaEntries, size_t numDeltaEntries) {
+    Mutex::Autolock autolock(mLock);
+
+    mDeltaEntries = deltaEntries;
+    mNumDeltaEntries = numDeltaEntries;
+    mCurrentDeltaEntry = 0;
+    mCurrentEntrySampleIndex = 0;
+}
+
+int32_t SampleTable::CompositionDeltaLookup::getCompositionTimeOffset(
+        uint32_t sampleIndex) {
+    Mutex::Autolock autolock(mLock);
+
+    if (mDeltaEntries == NULL) {
+        return 0;
+    }
+
+    if (sampleIndex < mCurrentEntrySampleIndex) {
+        mCurrentDeltaEntry = 0;
+        mCurrentEntrySampleIndex = 0;
+    }
+
+    while (mCurrentDeltaEntry < mNumDeltaEntries) {
+        uint32_t sampleCount = mDeltaEntries[2 * mCurrentDeltaEntry];
+        if (sampleIndex < mCurrentEntrySampleIndex + sampleCount) {
+            return mDeltaEntries[2 * mCurrentDeltaEntry + 1];
+        }
+
+        mCurrentEntrySampleIndex += sampleCount;
+        ++mCurrentDeltaEntry;
+    }
+
+    return 0;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+SampleTable::SampleTable(DataSourceBase *source)
+    : mDataSource(source),
+      mChunkOffsetOffset(-1),
+      mChunkOffsetType(0),
+      mNumChunkOffsets(0),
+      mSampleToChunkOffset(-1),
+      mNumSampleToChunkOffsets(0),
+      mSampleSizeOffset(-1),
+      mSampleSizeFieldSize(0),
+      mDefaultSampleSize(0),
+      mNumSampleSizes(0),
+      mHasTimeToSample(false),
+      mTimeToSampleCount(0),
+      mTimeToSample(NULL),
+      mSampleTimeEntries(NULL),
+      mCompositionTimeDeltaEntries(NULL),
+      mNumCompositionTimeDeltaEntries(0),
+      mCompositionDeltaLookup(new CompositionDeltaLookup),
+      mSyncSampleOffset(-1),
+      mNumSyncSamples(0),
+      mSyncSamples(NULL),
+      mLastSyncSampleIndex(0),
+      mSampleToChunkEntries(NULL),
+      mTotalSize(0) {
+    mSampleIterator = new SampleIterator(this);
+}
+
+SampleTable::~SampleTable() {
+    delete[] mSampleToChunkEntries;
+    mSampleToChunkEntries = NULL;
+
+    delete[] mSyncSamples;
+    mSyncSamples = NULL;
+
+    delete[] mTimeToSample;
+    mTimeToSample = NULL;
+
+    delete mCompositionDeltaLookup;
+    mCompositionDeltaLookup = NULL;
+
+    delete[] mCompositionTimeDeltaEntries;
+    mCompositionTimeDeltaEntries = NULL;
+
+    delete[] mSampleTimeEntries;
+    mSampleTimeEntries = NULL;
+
+    delete mSampleIterator;
+    mSampleIterator = NULL;
+}
+
+bool SampleTable::isValid() const {
+    return mChunkOffsetOffset >= 0
+        && mSampleToChunkOffset >= 0
+        && mSampleSizeOffset >= 0
+        && mHasTimeToSample;
+}
+
+status_t SampleTable::setChunkOffsetParams(
+        uint32_t type, off64_t data_offset, size_t data_size) {
+    if (mChunkOffsetOffset >= 0) {
+        return ERROR_MALFORMED;
+    }
+
+    CHECK(type == kChunkOffsetType32 || type == kChunkOffsetType64);
+
+    mChunkOffsetOffset = data_offset;
+    mChunkOffsetType = type;
+
+    if (data_size < 8) {
+        return ERROR_MALFORMED;
+    }
+
+    uint8_t header[8];
+    if (mDataSource->readAt(
+                data_offset, header, sizeof(header)) < (ssize_t)sizeof(header)) {
+        return ERROR_IO;
+    }
+
+    if (U32_AT(header) != 0) {
+        // Expected version = 0, flags = 0.
+        return ERROR_MALFORMED;
+    }
+
+    mNumChunkOffsets = U32_AT(&header[4]);
+
+    if (mChunkOffsetType == kChunkOffsetType32) {
+      if ((data_size - 8) / 4 < mNumChunkOffsets) {
+            return ERROR_MALFORMED;
+        }
+    } else {
+      if ((data_size - 8) / 8 < mNumChunkOffsets) {
+            return ERROR_MALFORMED;
+        }
+    }
+
+    return OK;
+}
+
+status_t SampleTable::setSampleToChunkParams(
+        off64_t data_offset, size_t data_size) {
+    if (mSampleToChunkOffset >= 0) {
+        // already set
+        return ERROR_MALFORMED;
+    }
+
+    if (data_offset < 0) {
+        return ERROR_MALFORMED;
+    }
+
+    mSampleToChunkOffset = data_offset;
+
+    if (data_size < 8) {
+        return ERROR_MALFORMED;
+    }
+
+    uint8_t header[8];
+    if (mDataSource->readAt(
+                data_offset, header, sizeof(header)) < (ssize_t)sizeof(header)) {
+        return ERROR_IO;
+    }
+
+    if (U32_AT(header) != 0) {
+        // Expected version = 0, flags = 0.
+        return ERROR_MALFORMED;
+    }
+
+    mNumSampleToChunkOffsets = U32_AT(&header[4]);
+
+    if ((data_size - 8) / sizeof(SampleToChunkEntry) < mNumSampleToChunkOffsets) {
+        return ERROR_MALFORMED;
+    }
+
+    if ((uint64_t)kMaxTotalSize / sizeof(SampleToChunkEntry) <=
+            (uint64_t)mNumSampleToChunkOffsets) {
+        ALOGE("Sample-to-chunk table size too large.");
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    mTotalSize += (uint64_t)mNumSampleToChunkOffsets *
+            sizeof(SampleToChunkEntry);
+    if (mTotalSize > kMaxTotalSize) {
+        ALOGE("Sample-to-chunk table size would make sample table too large.\n"
+              "    Requested sample-to-chunk table size = %llu\n"
+              "    Eventual sample table size >= %llu\n"
+              "    Allowed sample table size = %llu\n",
+              (unsigned long long)mNumSampleToChunkOffsets *
+                      sizeof(SampleToChunkEntry),
+              (unsigned long long)mTotalSize,
+              (unsigned long long)kMaxTotalSize);
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    mSampleToChunkEntries =
+        new (std::nothrow) SampleToChunkEntry[mNumSampleToChunkOffsets];
+    if (!mSampleToChunkEntries) {
+        ALOGE("Cannot allocate sample-to-chunk table with %llu entries.",
+                (unsigned long long)mNumSampleToChunkOffsets);
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    if (mNumSampleToChunkOffsets == 0) {
+        return OK;
+    }
+
+    if ((off64_t)(kMaxOffset - 8 -
+            ((mNumSampleToChunkOffsets - 1) * sizeof(SampleToChunkEntry)))
+            < mSampleToChunkOffset) {
+        return ERROR_MALFORMED;
+    }
+
+    for (uint32_t i = 0; i < mNumSampleToChunkOffsets; ++i) {
+        uint8_t buffer[sizeof(SampleToChunkEntry)];
+
+        if (mDataSource->readAt(
+                    mSampleToChunkOffset + 8 + i * sizeof(SampleToChunkEntry),
+                    buffer,
+                    sizeof(buffer))
+                != (ssize_t)sizeof(buffer)) {
+            return ERROR_IO;
+        }
+        // chunk index is 1 based in the spec.
+        if (U32_AT(buffer) < 1) {
+            ALOGE("b/23534160");
+            return ERROR_OUT_OF_RANGE;
+        }
+
+        // We want the chunk index to be 0-based.
+        mSampleToChunkEntries[i].startChunk = U32_AT(buffer) - 1;
+        mSampleToChunkEntries[i].samplesPerChunk = U32_AT(&buffer[4]);
+        mSampleToChunkEntries[i].chunkDesc = U32_AT(&buffer[8]);
+    }
+
+    return OK;
+}
+
+status_t SampleTable::setSampleSizeParams(
+        uint32_t type, off64_t data_offset, size_t data_size) {
+    if (mSampleSizeOffset >= 0) {
+        return ERROR_MALFORMED;
+    }
+
+    CHECK(type == kSampleSizeType32 || type == kSampleSizeTypeCompact);
+
+    mSampleSizeOffset = data_offset;
+
+    if (data_size < 12) {
+        return ERROR_MALFORMED;
+    }
+
+    uint8_t header[12];
+    if (mDataSource->readAt(
+                data_offset, header, sizeof(header)) < (ssize_t)sizeof(header)) {
+        return ERROR_IO;
+    }
+
+    if (U32_AT(header) != 0) {
+        // Expected version = 0, flags = 0.
+        return ERROR_MALFORMED;
+    }
+
+    mDefaultSampleSize = U32_AT(&header[4]);
+    mNumSampleSizes = U32_AT(&header[8]);
+    if (mNumSampleSizes > (UINT32_MAX - 12) / 16) {
+        ALOGE("b/23247055, mNumSampleSizes(%u)", mNumSampleSizes);
+        return ERROR_MALFORMED;
+    }
+
+    if (type == kSampleSizeType32) {
+        mSampleSizeFieldSize = 32;
+
+        if (mDefaultSampleSize != 0) {
+            return OK;
+        }
+
+        if (data_size < 12 + mNumSampleSizes * 4) {
+            return ERROR_MALFORMED;
+        }
+    } else {
+        if ((mDefaultSampleSize & 0xffffff00) != 0) {
+            // The high 24 bits are reserved and must be 0.
+            return ERROR_MALFORMED;
+        }
+
+        mSampleSizeFieldSize = mDefaultSampleSize & 0xff;
+        mDefaultSampleSize = 0;
+
+        if (mSampleSizeFieldSize != 4 && mSampleSizeFieldSize != 8
+            && mSampleSizeFieldSize != 16) {
+            return ERROR_MALFORMED;
+        }
+
+        if (data_size < 12 + (mNumSampleSizes * mSampleSizeFieldSize + 4) / 8) {
+            return ERROR_MALFORMED;
+        }
+    }
+
+    return OK;
+}
+
+status_t SampleTable::setTimeToSampleParams(
+        off64_t data_offset, size_t data_size) {
+    if (mHasTimeToSample || data_size < 8) {
+        return ERROR_MALFORMED;
+    }
+
+    uint8_t header[8];
+    if (mDataSource->readAt(
+                data_offset, header, sizeof(header)) < (ssize_t)sizeof(header)) {
+        return ERROR_IO;
+    }
+
+    if (U32_AT(header) != 0) {
+        // Expected version = 0, flags = 0.
+        return ERROR_MALFORMED;
+    }
+
+    mTimeToSampleCount = U32_AT(&header[4]);
+    if (mTimeToSampleCount > UINT32_MAX / (2 * sizeof(uint32_t))) {
+        // Choose this bound because
+        // 1) 2 * sizeof(uint32_t) is the amount of memory needed for one
+        //    time-to-sample entry in the time-to-sample table.
+        // 2) mTimeToSampleCount is the number of entries of the time-to-sample
+        //    table.
+        // 3) We hope that the table size does not exceed UINT32_MAX.
+        ALOGE("Time-to-sample table size too large.");
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    // Note: At this point, we know that mTimeToSampleCount * 2 will not
+    // overflow because of the above condition.
+
+    uint64_t allocSize = (uint64_t)mTimeToSampleCount * 2 * sizeof(uint32_t);
+    mTotalSize += allocSize;
+    if (mTotalSize > kMaxTotalSize) {
+        ALOGE("Time-to-sample table size would make sample table too large.\n"
+              "    Requested time-to-sample table size = %llu\n"
+              "    Eventual sample table size >= %llu\n"
+              "    Allowed sample table size = %llu\n",
+              (unsigned long long)allocSize,
+              (unsigned long long)mTotalSize,
+              (unsigned long long)kMaxTotalSize);
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    mTimeToSample = new (std::nothrow) uint32_t[mTimeToSampleCount * 2];
+    if (!mTimeToSample) {
+        ALOGE("Cannot allocate time-to-sample table with %llu entries.",
+                (unsigned long long)mTimeToSampleCount);
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    if (mDataSource->readAt(data_offset + 8, mTimeToSample,
+            (size_t)allocSize) < (ssize_t)allocSize) {
+        ALOGE("Incomplete data read for time-to-sample table.");
+        return ERROR_IO;
+    }
+
+    for (size_t i = 0; i < mTimeToSampleCount * 2; ++i) {
+        mTimeToSample[i] = ntohl(mTimeToSample[i]);
+    }
+
+    mHasTimeToSample = true;
+    return OK;
+}
+
+// NOTE: per 14996-12, version 0 ctts contains unsigned values, while version 1
+// contains signed values, however some software creates version 0 files that
+// contain signed values, so we're always treating the values as signed,
+// regardless of version.
+status_t SampleTable::setCompositionTimeToSampleParams(
+        off64_t data_offset, size_t data_size) {
+    ALOGI("There are reordered frames present.");
+
+    if (mCompositionTimeDeltaEntries != NULL || data_size < 8) {
+        return ERROR_MALFORMED;
+    }
+
+    uint8_t header[8];
+    if (mDataSource->readAt(
+                data_offset, header, sizeof(header))
+            < (ssize_t)sizeof(header)) {
+        return ERROR_IO;
+    }
+
+    uint32_t flags = U32_AT(header);
+    uint32_t version = flags >> 24;
+    flags &= 0xffffff;
+
+    if ((version != 0 && version != 1) || flags != 0) {
+        // Expected version = 0 or 1, flags = 0.
+        return ERROR_MALFORMED;
+    }
+
+    size_t numEntries = U32_AT(&header[4]);
+
+    if (((SIZE_MAX / 8) - 1 < numEntries) || (data_size != (numEntries + 1) * 8)) {
+        return ERROR_MALFORMED;
+    }
+
+    mNumCompositionTimeDeltaEntries = numEntries;
+    uint64_t allocSize = (uint64_t)numEntries * 2 * sizeof(int32_t);
+    if (allocSize > kMaxTotalSize) {
+        ALOGE("Composition-time-to-sample table size too large.");
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    mTotalSize += allocSize;
+    if (mTotalSize > kMaxTotalSize) {
+        ALOGE("Composition-time-to-sample table would make sample table too large.\n"
+              "    Requested composition-time-to-sample table size = %llu\n"
+              "    Eventual sample table size >= %llu\n"
+              "    Allowed sample table size = %llu\n",
+              (unsigned long long)allocSize,
+              (unsigned long long)mTotalSize,
+              (unsigned long long)kMaxTotalSize);
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    mCompositionTimeDeltaEntries = new (std::nothrow) int32_t[2 * numEntries];
+    if (!mCompositionTimeDeltaEntries) {
+        ALOGE("Cannot allocate composition-time-to-sample table with %llu "
+                "entries.", (unsigned long long)numEntries);
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    if (mDataSource->readAt(data_offset + 8, mCompositionTimeDeltaEntries,
+            (size_t)allocSize) < (ssize_t)allocSize) {
+        delete[] mCompositionTimeDeltaEntries;
+        mCompositionTimeDeltaEntries = NULL;
+
+        return ERROR_IO;
+    }
+
+    for (size_t i = 0; i < 2 * numEntries; ++i) {
+        mCompositionTimeDeltaEntries[i] = ntohl(mCompositionTimeDeltaEntries[i]);
+    }
+
+    mCompositionDeltaLookup->setEntries(
+            mCompositionTimeDeltaEntries, mNumCompositionTimeDeltaEntries);
+
+    return OK;
+}
+
+status_t SampleTable::setSyncSampleParams(off64_t data_offset, size_t data_size) {
+    if (mSyncSampleOffset >= 0 || data_size < 8) {
+        return ERROR_MALFORMED;
+    }
+
+    uint8_t header[8];
+    if (mDataSource->readAt(
+                data_offset, header, sizeof(header)) < (ssize_t)sizeof(header)) {
+        return ERROR_IO;
+    }
+
+    if (U32_AT(header) != 0) {
+        // Expected version = 0, flags = 0.
+        return ERROR_MALFORMED;
+    }
+
+    uint32_t numSyncSamples = U32_AT(&header[4]);
+
+    if (numSyncSamples < 2) {
+        ALOGV("Table of sync samples is empty or has only a single entry!");
+    }
+
+    uint64_t allocSize = (uint64_t)numSyncSamples * sizeof(uint32_t);
+    if (allocSize > kMaxTotalSize) {
+        ALOGE("Sync sample table size too large.");
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    mTotalSize += allocSize;
+    if (mTotalSize > kMaxTotalSize) {
+        ALOGE("Sync sample table size would make sample table too large.\n"
+              "    Requested sync sample table size = %llu\n"
+              "    Eventual sample table size >= %llu\n"
+              "    Allowed sample table size = %llu\n",
+              (unsigned long long)allocSize,
+              (unsigned long long)mTotalSize,
+              (unsigned long long)kMaxTotalSize);
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    mSyncSamples = new (std::nothrow) uint32_t[numSyncSamples];
+    if (!mSyncSamples) {
+        ALOGE("Cannot allocate sync sample table with %llu entries.",
+                (unsigned long long)numSyncSamples);
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    if (mDataSource->readAt(data_offset + 8, mSyncSamples,
+            (size_t)allocSize) != (ssize_t)allocSize) {
+        delete[] mSyncSamples;
+        mSyncSamples = NULL;
+        return ERROR_IO;
+    }
+
+    for (size_t i = 0; i < numSyncSamples; ++i) {
+        if (mSyncSamples[i] == 0) {
+            ALOGE("b/32423862, unexpected zero value in stss");
+            continue;
+        }
+        mSyncSamples[i] = ntohl(mSyncSamples[i]) - 1;
+    }
+
+    mSyncSampleOffset = data_offset;
+    mNumSyncSamples = numSyncSamples;
+
+    return OK;
+}
+
+uint32_t SampleTable::countChunkOffsets() const {
+    return mNumChunkOffsets;
+}
+
+uint32_t SampleTable::countSamples() const {
+    return mNumSampleSizes;
+}
+
+status_t SampleTable::getMaxSampleSize(size_t *max_size) {
+    Mutex::Autolock autoLock(mLock);
+
+    *max_size = 0;
+
+    for (uint32_t i = 0; i < mNumSampleSizes; ++i) {
+        size_t sample_size;
+        status_t err = getSampleSize_l(i, &sample_size);
+
+        if (err != OK) {
+            return err;
+        }
+
+        if (sample_size > *max_size) {
+            *max_size = sample_size;
+        }
+    }
+
+    return OK;
+}
+
+uint32_t abs_difference(uint32_t time1, uint32_t time2) {
+    return time1 > time2 ? time1 - time2 : time2 - time1;
+}
+
+// static
+int SampleTable::CompareIncreasingTime(const void *_a, const void *_b) {
+    const SampleTimeEntry *a = (const SampleTimeEntry *)_a;
+    const SampleTimeEntry *b = (const SampleTimeEntry *)_b;
+
+    if (a->mCompositionTime < b->mCompositionTime) {
+        return -1;
+    } else if (a->mCompositionTime > b->mCompositionTime) {
+        return 1;
+    }
+
+    return 0;
+}
+
+void SampleTable::buildSampleEntriesTable() {
+    Mutex::Autolock autoLock(mLock);
+
+    if (mSampleTimeEntries != NULL || mNumSampleSizes == 0) {
+        if (mNumSampleSizes == 0) {
+            ALOGE("b/23247055, mNumSampleSizes(%u)", mNumSampleSizes);
+        }
+        return;
+    }
+
+    mTotalSize += (uint64_t)mNumSampleSizes * sizeof(SampleTimeEntry);
+    if (mTotalSize > kMaxTotalSize) {
+        ALOGE("Sample entry table size would make sample table too large.\n"
+              "    Requested sample entry table size = %llu\n"
+              "    Eventual sample table size >= %llu\n"
+              "    Allowed sample table size = %llu\n",
+              (unsigned long long)mNumSampleSizes * sizeof(SampleTimeEntry),
+              (unsigned long long)mTotalSize,
+              (unsigned long long)kMaxTotalSize);
+        return;
+    }
+
+    mSampleTimeEntries = new (std::nothrow) SampleTimeEntry[mNumSampleSizes];
+    if (!mSampleTimeEntries) {
+        ALOGE("Cannot allocate sample entry table with %llu entries.",
+                (unsigned long long)mNumSampleSizes);
+        return;
+    }
+
+    uint32_t sampleIndex = 0;
+    uint32_t sampleTime = 0;
+
+    for (uint32_t i = 0; i < mTimeToSampleCount; ++i) {
+        uint32_t n = mTimeToSample[2 * i];
+        uint32_t delta = mTimeToSample[2 * i + 1];
+
+        for (uint32_t j = 0; j < n; ++j) {
+            if (sampleIndex < mNumSampleSizes) {
+                // Technically this should always be the case if the file
+                // is well-formed, but you know... there's (gasp) malformed
+                // content out there.
+
+                mSampleTimeEntries[sampleIndex].mSampleIndex = sampleIndex;
+
+                int32_t compTimeDelta =
+                    mCompositionDeltaLookup->getCompositionTimeOffset(
+                            sampleIndex);
+
+                if ((compTimeDelta < 0 && sampleTime <
+                        (compTimeDelta == INT32_MIN ?
+                                INT32_MAX : uint32_t(-compTimeDelta)))
+                        || (compTimeDelta > 0 &&
+                                sampleTime > UINT32_MAX - compTimeDelta)) {
+                    ALOGE("%u + %d would overflow, clamping",
+                            sampleTime, compTimeDelta);
+                    if (compTimeDelta < 0) {
+                        sampleTime = 0;
+                    } else {
+                        sampleTime = UINT32_MAX;
+                    }
+                    compTimeDelta = 0;
+                }
+
+                mSampleTimeEntries[sampleIndex].mCompositionTime =
+                        compTimeDelta > 0 ? sampleTime + compTimeDelta:
+                                sampleTime - (-compTimeDelta);
+            }
+
+            ++sampleIndex;
+            if (sampleTime > UINT32_MAX - delta) {
+                ALOGE("%u + %u would overflow, clamping",
+                    sampleTime, delta);
+                sampleTime = UINT32_MAX;
+            } else {
+                sampleTime += delta;
+            }
+        }
+    }
+
+    qsort(mSampleTimeEntries, mNumSampleSizes, sizeof(SampleTimeEntry),
+          CompareIncreasingTime);
+}
+
+status_t SampleTable::findSampleAtTime(
+        uint64_t req_time, uint64_t scale_num, uint64_t scale_den,
+        uint32_t *sample_index, uint32_t flags) {
+    buildSampleEntriesTable();
+
+    if (mSampleTimeEntries == NULL) {
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    if (flags == kFlagFrameIndex) {
+        if (req_time >= mNumSampleSizes) {
+            return ERROR_OUT_OF_RANGE;
+        }
+        *sample_index = mSampleTimeEntries[req_time].mSampleIndex;
+        return OK;
+    }
+
+    uint32_t left = 0;
+    uint32_t right_plus_one = mNumSampleSizes;
+    while (left < right_plus_one) {
+        uint32_t center = left + (right_plus_one - left) / 2;
+        uint64_t centerTime =
+            getSampleTime(center, scale_num, scale_den);
+
+        if (req_time < centerTime) {
+            right_plus_one = center;
+        } else if (req_time > centerTime) {
+            left = center + 1;
+        } else {
+            *sample_index = mSampleTimeEntries[center].mSampleIndex;
+            return OK;
+        }
+    }
+
+    uint32_t closestIndex = left;
+
+    if (closestIndex == mNumSampleSizes) {
+        if (flags == kFlagAfter) {
+            return ERROR_OUT_OF_RANGE;
+        }
+        flags = kFlagBefore;
+    } else if (closestIndex == 0) {
+        if (flags == kFlagBefore) {
+            // normally we should return out of range, but that is
+            // treated as end-of-stream.  instead return first sample
+            //
+            // return ERROR_OUT_OF_RANGE;
+        }
+        flags = kFlagAfter;
+    }
+
+    switch (flags) {
+        case kFlagBefore:
+        {
+            --closestIndex;
+            break;
+        }
+
+        case kFlagAfter:
+        {
+            // nothing to do
+            break;
+        }
+
+        default:
+        {
+            CHECK(flags == kFlagClosest);
+            // pick closest based on timestamp. use abs_difference for safety
+            if (abs_difference(
+                    getSampleTime(closestIndex, scale_num, scale_den), req_time) >
+                abs_difference(
+                    req_time, getSampleTime(closestIndex - 1, scale_num, scale_den))) {
+                --closestIndex;
+            }
+            break;
+        }
+    }
+
+    *sample_index = mSampleTimeEntries[closestIndex].mSampleIndex;
+    return OK;
+}
+
+status_t SampleTable::findSyncSampleNear(
+        uint32_t start_sample_index, uint32_t *sample_index, uint32_t flags) {
+    Mutex::Autolock autoLock(mLock);
+
+    *sample_index = 0;
+
+    if (mSyncSampleOffset < 0) {
+        // All samples are sync-samples.
+        *sample_index = start_sample_index;
+        return OK;
+    }
+
+    if (mNumSyncSamples == 0) {
+        *sample_index = 0;
+        return OK;
+    }
+
+    uint32_t left = 0;
+    uint32_t right_plus_one = mNumSyncSamples;
+    while (left < right_plus_one) {
+        uint32_t center = left + (right_plus_one - left) / 2;
+        uint32_t x = mSyncSamples[center];
+
+        if (start_sample_index < x) {
+            right_plus_one = center;
+        } else if (start_sample_index > x) {
+            left = center + 1;
+        } else {
+            *sample_index = x;
+            return OK;
+        }
+    }
+
+    if (left == mNumSyncSamples) {
+        if (flags == kFlagAfter) {
+            ALOGE("tried to find a sync frame after the last one: %d", left);
+            return ERROR_OUT_OF_RANGE;
+        }
+        flags = kFlagBefore;
+    }
+    else if (left == 0) {
+        if (flags == kFlagBefore) {
+            ALOGE("tried to find a sync frame before the first one: %d", left);
+
+            // normally we should return out of range, but that is
+            // treated as end-of-stream.  instead seek to first sync
+            //
+            // return ERROR_OUT_OF_RANGE;
+        }
+        flags = kFlagAfter;
+    }
+
+    // Now ssi[left - 1] <(=) start_sample_index <= ssi[left]
+    switch (flags) {
+        case kFlagBefore:
+        {
+            --left;
+            break;
+        }
+        case kFlagAfter:
+        {
+            // nothing to do
+            break;
+        }
+        default:
+        {
+            // this route is not used, but implement it nonetheless
+            CHECK(flags == kFlagClosest);
+
+            status_t err = mSampleIterator->seekTo(start_sample_index);
+            if (err != OK) {
+                return err;
+            }
+            uint32_t sample_time = mSampleIterator->getSampleTime();
+
+            err = mSampleIterator->seekTo(mSyncSamples[left]);
+            if (err != OK) {
+                return err;
+            }
+            uint32_t upper_time = mSampleIterator->getSampleTime();
+
+            err = mSampleIterator->seekTo(mSyncSamples[left - 1]);
+            if (err != OK) {
+                return err;
+            }
+            uint32_t lower_time = mSampleIterator->getSampleTime();
+
+            // use abs_difference for safety
+            if (abs_difference(upper_time, sample_time) >
+                abs_difference(sample_time, lower_time)) {
+                --left;
+            }
+            break;
+        }
+    }
+
+    *sample_index = mSyncSamples[left];
+    return OK;
+}
+
+status_t SampleTable::findThumbnailSample(uint32_t *sample_index) {
+    Mutex::Autolock autoLock(mLock);
+
+    if (mSyncSampleOffset < 0) {
+        // All samples are sync-samples.
+        *sample_index = 0;
+        return OK;
+    }
+
+    uint32_t bestSampleIndex = 0;
+    size_t maxSampleSize = 0;
+
+    static const size_t kMaxNumSyncSamplesToScan = 20;
+
+    // Consider the first kMaxNumSyncSamplesToScan sync samples and
+    // pick the one with the largest (compressed) size as the thumbnail.
+
+    size_t numSamplesToScan = mNumSyncSamples;
+    if (numSamplesToScan > kMaxNumSyncSamplesToScan) {
+        numSamplesToScan = kMaxNumSyncSamplesToScan;
+    }
+
+    for (size_t i = 0; i < numSamplesToScan; ++i) {
+        uint32_t x = mSyncSamples[i];
+
+        // Now x is a sample index.
+        size_t sampleSize;
+        status_t err = getSampleSize_l(x, &sampleSize);
+        if (err != OK) {
+            return err;
+        }
+
+        if (i == 0 || sampleSize > maxSampleSize) {
+            bestSampleIndex = x;
+            maxSampleSize = sampleSize;
+        }
+    }
+
+    *sample_index = bestSampleIndex;
+
+    return OK;
+}
+
+status_t SampleTable::getSampleSize_l(
+        uint32_t sampleIndex, size_t *sampleSize) {
+    return mSampleIterator->getSampleSizeDirect(
+            sampleIndex, sampleSize);
+}
+
+status_t SampleTable::getMetaDataForSample(
+        uint32_t sampleIndex,
+        off64_t *offset,
+        size_t *size,
+        uint32_t *compositionTime,
+        bool *isSyncSample,
+        uint32_t *sampleDuration) {
+    Mutex::Autolock autoLock(mLock);
+
+    status_t err;
+    if ((err = mSampleIterator->seekTo(sampleIndex)) != OK) {
+        return err;
+    }
+
+    if (offset) {
+        *offset = mSampleIterator->getSampleOffset();
+    }
+
+    if (size) {
+        *size = mSampleIterator->getSampleSize();
+    }
+
+    if (compositionTime) {
+        *compositionTime = mSampleIterator->getSampleTime();
+    }
+
+    if (isSyncSample) {
+        *isSyncSample = false;
+        if (mSyncSampleOffset < 0) {
+            // Every sample is a sync sample.
+            *isSyncSample = true;
+        } else {
+            size_t i = (mLastSyncSampleIndex < mNumSyncSamples)
+                    && (mSyncSamples[mLastSyncSampleIndex] <= sampleIndex)
+                ? mLastSyncSampleIndex : 0;
+
+            while (i < mNumSyncSamples && mSyncSamples[i] < sampleIndex) {
+                ++i;
+            }
+
+            if (i < mNumSyncSamples && mSyncSamples[i] == sampleIndex) {
+                *isSyncSample = true;
+            }
+
+            mLastSyncSampleIndex = i;
+        }
+    }
+
+    if (sampleDuration) {
+        *sampleDuration = mSampleIterator->getSampleDuration();
+    }
+
+    return OK;
+}
+
+int32_t SampleTable::getCompositionTimeOffset(uint32_t sampleIndex) {
+    return mCompositionDeltaLookup->getCompositionTimeOffset(sampleIndex);
+}
+
+}  // namespace android

diff --git a/media/extractors/mp4/SampleTable.h b/media/extractors/mp4/SampleTable.h
new file mode 100644
index 0000000..e4e974b
--- /dev/null
+++ b/media/extractors/mp4/SampleTable.h

@@ -0,0 +1,171 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SAMPLE_TABLE_H_
+
+#define SAMPLE_TABLE_H_
+
+#include <sys/types.h>
+#include <stdint.h>
+
+#include <media/stagefright/MediaErrors.h>
+#include <utils/RefBase.h>
+#include <utils/threads.h>
+
+namespace android {
+
+class DataSourceBase;
+struct SampleIterator;
+
+class SampleTable : public RefBase {
+public:
+    explicit SampleTable(DataSourceBase *source);
+
+    bool isValid() const;
+
+    // type can be 'stco' or 'co64'.
+    status_t setChunkOffsetParams(
+            uint32_t type, off64_t data_offset, size_t data_size);
+
+    status_t setSampleToChunkParams(off64_t data_offset, size_t data_size);
+
+    // type can be 'stsz' or 'stz2'.
+    status_t setSampleSizeParams(
+            uint32_t type, off64_t data_offset, size_t data_size);
+
+    status_t setTimeToSampleParams(off64_t data_offset, size_t data_size);
+
+    status_t setCompositionTimeToSampleParams(
+            off64_t data_offset, size_t data_size);
+
+    status_t setSyncSampleParams(off64_t data_offset, size_t data_size);
+
+    ////////////////////////////////////////////////////////////////////////////
+
+    uint32_t countChunkOffsets() const;
+
+    uint32_t countSamples() const;
+
+    status_t getMaxSampleSize(size_t *size);
+
+    status_t getMetaDataForSample(
+            uint32_t sampleIndex,
+            off64_t *offset,
+            size_t *size,
+            uint32_t *compositionTime,
+            bool *isSyncSample = NULL,
+            uint32_t *sampleDuration = NULL);
+
+    enum {
+        kFlagBefore,
+        kFlagAfter,
+        kFlagClosest,
+        kFlagFrameIndex,
+    };
+    status_t findSampleAtTime(
+            uint64_t req_time, uint64_t scale_num, uint64_t scale_den,
+            uint32_t *sample_index, uint32_t flags);
+
+    status_t findSyncSampleNear(
+            uint32_t start_sample_index, uint32_t *sample_index,
+            uint32_t flags);
+
+    status_t findThumbnailSample(uint32_t *sample_index);
+
+protected:
+    ~SampleTable();
+
+private:
+    struct CompositionDeltaLookup;
+
+    static const uint32_t kChunkOffsetType32;
+    static const uint32_t kChunkOffsetType64;
+    static const uint32_t kSampleSizeType32;
+    static const uint32_t kSampleSizeTypeCompact;
+
+    // Limit the total size of all internal tables to 200MiB.
+    static const size_t kMaxTotalSize = 200 * (1 << 20);
+
+    DataSourceBase *mDataSource;
+    Mutex mLock;
+
+    off64_t mChunkOffsetOffset;
+    uint32_t mChunkOffsetType;
+    uint32_t mNumChunkOffsets;
+
+    off64_t mSampleToChunkOffset;
+    uint32_t mNumSampleToChunkOffsets;
+
+    off64_t mSampleSizeOffset;
+    uint32_t mSampleSizeFieldSize;
+    uint32_t mDefaultSampleSize;
+    uint32_t mNumSampleSizes;
+
+    bool mHasTimeToSample;
+    uint32_t mTimeToSampleCount;
+    uint32_t* mTimeToSample;
+
+    struct SampleTimeEntry {
+        uint32_t mSampleIndex;
+        uint32_t mCompositionTime;
+    };
+    SampleTimeEntry *mSampleTimeEntries;
+
+    int32_t *mCompositionTimeDeltaEntries;
+    size_t mNumCompositionTimeDeltaEntries;
+    CompositionDeltaLookup *mCompositionDeltaLookup;
+
+    off64_t mSyncSampleOffset;
+    uint32_t mNumSyncSamples;
+    uint32_t *mSyncSamples;
+    size_t mLastSyncSampleIndex;
+
+    SampleIterator *mSampleIterator;
+
+    struct SampleToChunkEntry {
+        uint32_t startChunk;
+        uint32_t samplesPerChunk;
+        uint32_t chunkDesc;
+    };
+    SampleToChunkEntry *mSampleToChunkEntries;
+
+    // Approximate size of all tables combined.
+    uint64_t mTotalSize;
+
+    friend struct SampleIterator;
+
+    // normally we don't round
+    inline uint64_t getSampleTime(
+            size_t sample_index, uint64_t scale_num, uint64_t scale_den) const {
+        return (sample_index < (size_t)mNumSampleSizes && mSampleTimeEntries != NULL
+                && scale_den != 0)
+                ? (mSampleTimeEntries[sample_index].mCompositionTime * scale_num) / scale_den : 0;
+    }
+
+    status_t getSampleSize_l(uint32_t sample_index, size_t *sample_size);
+    int32_t getCompositionTimeOffset(uint32_t sampleIndex);
+
+    static int CompareIncreasingTime(const void *, const void *);
+
+    void buildSampleEntriesTable();
+
+    SampleTable(const SampleTable &);
+    SampleTable &operator=(const SampleTable &);
+};
+
+}  // namespace android
+
+#endif  // SAMPLE_TABLE_H_

diff --git a/media/extractors/mp4/exports.lds b/media/extractors/mp4/exports.lds
new file mode 100644
index 0000000..b1309ee
--- /dev/null
+++ b/media/extractors/mp4/exports.lds

@@ -0,0 +1 @@
+{ global: GETEXTRACTORDEF; local: *; };

diff --git a/media/extractors/mpeg2/Android.bp b/media/extractors/mpeg2/Android.bp
new file mode 100644
index 0000000..5e4a592
--- /dev/null
+++ b/media/extractors/mpeg2/Android.bp

@@ -0,0 +1,56 @@
+cc_library_shared {
+
+    srcs: [
+        "ExtractorBundle.cpp",
+        "MPEG2PSExtractor.cpp",
+        "MPEG2TSExtractor.cpp",
+    ],
+
+    include_dirs: [
+        "frameworks/av/media/libstagefright",
+        "frameworks/av/media/libstagefright/include",
+    ],
+
+    shared_libs: [
+        "android.hardware.cas@1.0",
+        "android.hardware.cas.native@1.0",
+        "android.hidl.token@1.0-utils",
+        "libbinder",
+        "libcrypto",
+        "libcutils",
+        "libhidlallocatorutils",
+        "libhidlbase",
+        "liblog",
+        "libmediaextractor",
+        "libstagefright_foundation",
+    ],
+
+    static_libs: [
+        "libstagefright_mpeg2support",
+        "libutils",
+    ],
+
+    name: "libmpeg2extractor",
+    relative_install_path: "extractors",
+
+    compile_multilib: "first",
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+        "-fvisibility=hidden",
+    ],
+    version_script: "exports.lds",
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+        diag: {
+            cfi: true,
+        },
+    },
+
+}

diff --git a/media/extractors/mpeg2/ExtractorBundle.cpp b/media/extractors/mpeg2/ExtractorBundle.cpp
new file mode 100644
index 0000000..8a0fa03
--- /dev/null
+++ b/media/extractors/mpeg2/ExtractorBundle.cpp

@@ -0,0 +1,59 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MPEG2ExtractorBundle"
+#include <utils/Log.h>
+
+#include <media/MediaExtractor.h>
+#include "MPEG2PSExtractor.h"
+#include "MPEG2TSExtractor.h"
+
+namespace android {
+
+extern "C" {
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
+MediaExtractor::ExtractorDef GETEXTRACTORDEF() {
+    return {
+        MediaExtractor::EXTRACTORDEF_VERSION,
+        UUID("3d1dcfeb-e40a-436d-a574-c2438a555e5f"),
+        1,
+        "MPEG2-PS/TS Extractor",
+        [](
+                DataSourceBase *source,
+                float *confidence,
+                void **,
+                MediaExtractor::FreeMetaFunc *) -> MediaExtractor::CreatorFunc {
+            if (SniffMPEG2TS(source, confidence)) {
+                return [](
+                        DataSourceBase *source,
+                        void *) -> MediaExtractor* {
+                    return new MPEG2TSExtractor(source);};
+            } else if (SniffMPEG2PS(source, confidence)) {
+                        return [](
+                                DataSourceBase *source,
+                                void *) -> MediaExtractor* {
+                            return new MPEG2PSExtractor(source);};
+            }
+            return NULL;
+        }
+    };
+}
+
+} // extern "C"
+
+} // namespace android

diff --git a/media/libstagefright/matroska/MODULE_LICENSE_APACHE2 b/media/extractors/mpeg2/MODULE_LICENSE_APACHE2
similarity index 100%
copy from media/libstagefright/matroska/MODULE_LICENSE_APACHE2
copy to media/extractors/mpeg2/MODULE_LICENSE_APACHE2


diff --git a/media/extractors/mpeg2/MPEG2PSExtractor.cpp b/media/extractors/mpeg2/MPEG2PSExtractor.cpp
new file mode 100644
index 0000000..6980b82
--- /dev/null
+++ b/media/extractors/mpeg2/MPEG2PSExtractor.cpp

@@ -0,0 +1,772 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MPEG2PSExtractor"
+#include <utils/Log.h>
+
+#include "MPEG2PSExtractor.h"
+
+#include "mpeg2ts/AnotherPacketSource.h"
+#include "mpeg2ts/ESQueue.h"
+
+#include <media/DataSourceBase.h>
+#include <media/MediaTrack.h>
+#include <media/stagefright/foundation/ABitReader.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/ByteUtils.h>
+#include <media/stagefright/foundation/hexdump.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <utils/String8.h>
+
+#include <inttypes.h>
+
+namespace android {
+
+struct MPEG2PSExtractor::Track : public MediaTrack, public RefBase {
+    Track(MPEG2PSExtractor *extractor,
+          unsigned stream_id, unsigned stream_type);
+
+    virtual status_t start(MetaDataBase *params);
+    virtual status_t stop();
+    virtual status_t getFormat(MetaDataBase &);
+
+    virtual status_t read(
+            MediaBufferBase **buffer, const ReadOptions *options);
+
+protected:
+    virtual ~Track();
+
+private:
+    friend struct MPEG2PSExtractor;
+
+    MPEG2PSExtractor *mExtractor;
+
+    unsigned mStreamID;
+    unsigned mStreamType;
+    ElementaryStreamQueue *mQueue;
+    sp<AnotherPacketSource> mSource;
+
+    status_t appendPESData(
+            unsigned PTS_DTS_flags,
+            uint64_t PTS, uint64_t DTS,
+            const uint8_t *data, size_t size);
+
+    DISALLOW_EVIL_CONSTRUCTORS(Track);
+};
+
+struct MPEG2PSExtractor::WrappedTrack : public MediaTrack {
+    WrappedTrack(MPEG2PSExtractor *extractor, const sp<Track> &track);
+
+    virtual status_t start(MetaDataBase *params);
+    virtual status_t stop();
+    virtual status_t getFormat(MetaDataBase &);
+
+    virtual status_t read(
+            MediaBufferBase **buffer, const ReadOptions *options);
+
+protected:
+    virtual ~WrappedTrack();
+
+private:
+    MPEG2PSExtractor *mExtractor;
+    sp<MPEG2PSExtractor::Track> mTrack;
+
+    DISALLOW_EVIL_CONSTRUCTORS(WrappedTrack);
+};
+
+////////////////////////////////////////////////////////////////////////////////
+
+MPEG2PSExtractor::MPEG2PSExtractor(DataSourceBase *source)
+    : mDataSource(source),
+      mOffset(0),
+      mFinalResult(OK),
+      mBuffer(new ABuffer(0)),
+      mScanning(true),
+      mProgramStreamMapValid(false) {
+    for (size_t i = 0; i < 500; ++i) {
+        if (feedMore() != OK) {
+            break;
+        }
+    }
+
+    // Remove all tracks that were unable to determine their format.
+    MetaDataBase meta;
+    for (size_t i = mTracks.size(); i > 0;) {
+        i--;
+        if (mTracks.valueAt(i)->getFormat(meta) != OK) {
+            mTracks.removeItemsAt(i);
+        }
+    }
+
+    mScanning = false;
+}
+
+MPEG2PSExtractor::~MPEG2PSExtractor() {
+}
+
+size_t MPEG2PSExtractor::countTracks() {
+    return mTracks.size();
+}
+
+MediaTrack *MPEG2PSExtractor::getTrack(size_t index) {
+    if (index >= mTracks.size()) {
+        return NULL;
+    }
+
+    return new WrappedTrack(this, mTracks.valueAt(index));
+}
+
+status_t MPEG2PSExtractor::getTrackMetaData(
+        MetaDataBase &meta,
+        size_t index, uint32_t /* flags */) {
+    if (index >= mTracks.size()) {
+        return UNKNOWN_ERROR;
+    }
+
+    return mTracks.valueAt(index)->getFormat(meta);
+}
+
+status_t MPEG2PSExtractor::getMetaData(MetaDataBase &meta) {
+    meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_CONTAINER_MPEG2PS);
+
+    return OK;
+}
+
+uint32_t MPEG2PSExtractor::flags() const {
+    return CAN_PAUSE;
+}
+
+status_t MPEG2PSExtractor::feedMore() {
+    Mutex::Autolock autoLock(mLock);
+
+    // How much data we're reading at a time
+    static const size_t kChunkSize = 8192;
+
+    for (;;) {
+        status_t err = dequeueChunk();
+
+        if (err == -EAGAIN && mFinalResult == OK) {
+            memmove(mBuffer->base(), mBuffer->data(), mBuffer->size());
+            mBuffer->setRange(0, mBuffer->size());
+
+            if (mBuffer->size() + kChunkSize > mBuffer->capacity()) {
+                size_t newCapacity = mBuffer->capacity() + kChunkSize;
+                sp<ABuffer> newBuffer = new ABuffer(newCapacity);
+                memcpy(newBuffer->data(), mBuffer->data(), mBuffer->size());
+                newBuffer->setRange(0, mBuffer->size());
+                mBuffer = newBuffer;
+            }
+
+            ssize_t n = mDataSource->readAt(
+                    mOffset, mBuffer->data() + mBuffer->size(), kChunkSize);
+
+            if (n < (ssize_t)kChunkSize) {
+                mFinalResult = (n < 0) ? (status_t)n : ERROR_END_OF_STREAM;
+                return mFinalResult;
+            }
+
+            mBuffer->setRange(mBuffer->offset(), mBuffer->size() + n);
+            mOffset += n;
+        } else if (err != OK) {
+            mFinalResult = err;
+            return err;
+        } else {
+            return OK;
+        }
+    }
+}
+
+status_t MPEG2PSExtractor::dequeueChunk() {
+    if (mBuffer->size() < 4) {
+        return -EAGAIN;
+    }
+
+    if (memcmp("\x00\x00\x01", mBuffer->data(), 3)) {
+        return ERROR_MALFORMED;
+    }
+
+    unsigned chunkType = mBuffer->data()[3];
+
+    ssize_t res;
+
+    switch (chunkType) {
+        case 0xba:
+        {
+            res = dequeuePack();
+            break;
+        }
+
+        case 0xbb:
+        {
+            res = dequeueSystemHeader();
+            break;
+        }
+
+        default:
+        {
+            res = dequeuePES();
+            break;
+        }
+    }
+
+    if (res > 0) {
+        if (mBuffer->size() < (size_t)res) {
+            return -EAGAIN;
+        }
+
+        mBuffer->setRange(mBuffer->offset() + res, mBuffer->size() - res);
+        res = OK;
+    }
+
+    return res;
+}
+
+ssize_t MPEG2PSExtractor::dequeuePack() {
+    // 32 + 2 + 3 + 1 + 15 + 1 + 15+ 1 + 9 + 1 + 22 + 1 + 1 | +5
+
+    if (mBuffer->size() < 14) {
+        return -EAGAIN;
+    }
+
+    unsigned pack_stuffing_length = mBuffer->data()[13] & 7;
+
+    return pack_stuffing_length + 14;
+}
+
+ssize_t MPEG2PSExtractor::dequeueSystemHeader() {
+    if (mBuffer->size() < 6) {
+        return -EAGAIN;
+    }
+
+    unsigned header_length = U16_AT(mBuffer->data() + 4);
+
+    return header_length + 6;
+}
+
+ssize_t MPEG2PSExtractor::dequeuePES() {
+    if (mBuffer->size() < 6) {
+        return -EAGAIN;
+    }
+
+    unsigned PES_packet_length = U16_AT(mBuffer->data() + 4);
+    if (PES_packet_length == 0u) {
+        ALOGE("PES_packet_length is 0");
+        return -EAGAIN;
+    }
+
+    size_t n = PES_packet_length + 6;
+
+    if (mBuffer->size() < n) {
+        return -EAGAIN;
+    }
+
+    ABitReader br(mBuffer->data(), n);
+
+    unsigned packet_startcode_prefix = br.getBits(24);
+
+    ALOGV("packet_startcode_prefix = 0x%08x", packet_startcode_prefix);
+
+    if (packet_startcode_prefix != 1) {
+        ALOGV("Supposedly payload_unit_start=1 unit does not start "
+             "with startcode.");
+
+        return ERROR_MALFORMED;
+    }
+
+    if (packet_startcode_prefix != 0x000001u) {
+        ALOGE("Wrong PES prefix");
+        return ERROR_MALFORMED;
+    }
+
+    unsigned stream_id = br.getBits(8);
+    ALOGV("stream_id = 0x%02x", stream_id);
+
+    /* unsigned PES_packet_length = */br.getBits(16);
+
+    if (stream_id == 0xbc) {
+        // program_stream_map
+
+        if (!mScanning) {
+            return n;
+        }
+
+        mStreamTypeByESID.clear();
+
+        /* unsigned current_next_indicator = */br.getBits(1);
+        /* unsigned reserved = */br.getBits(2);
+        /* unsigned program_stream_map_version = */br.getBits(5);
+        /* unsigned reserved = */br.getBits(7);
+        /* unsigned marker_bit = */br.getBits(1);
+        unsigned program_stream_info_length = br.getBits(16);
+
+        size_t offset = 0;
+        while (offset < program_stream_info_length) {
+            if (offset + 2 > program_stream_info_length) {
+                return ERROR_MALFORMED;
+            }
+
+            unsigned descriptor_tag = br.getBits(8);
+            unsigned descriptor_length = br.getBits(8);
+
+            ALOGI("found descriptor tag 0x%02x of length %u",
+                 descriptor_tag, descriptor_length);
+
+            if (offset + 2 + descriptor_length > program_stream_info_length) {
+                return ERROR_MALFORMED;
+            }
+
+            br.skipBits(8 * descriptor_length);
+
+            offset += 2 + descriptor_length;
+        }
+
+        unsigned elementary_stream_map_length = br.getBits(16);
+
+        offset = 0;
+        while (offset < elementary_stream_map_length) {
+            if (offset + 4 > elementary_stream_map_length) {
+                return ERROR_MALFORMED;
+            }
+
+            unsigned stream_type = br.getBits(8);
+            unsigned elementary_stream_id = br.getBits(8);
+
+            ALOGI("elementary stream id 0x%02x has stream type 0x%02x",
+                 elementary_stream_id, stream_type);
+
+            mStreamTypeByESID.add(elementary_stream_id, stream_type);
+
+            unsigned elementary_stream_info_length = br.getBits(16);
+
+            if (offset + 4 + elementary_stream_info_length
+                    > elementary_stream_map_length) {
+                return ERROR_MALFORMED;
+            }
+
+            offset += 4 + elementary_stream_info_length;
+        }
+
+        /* unsigned CRC32 = */br.getBits(32);
+
+        mProgramStreamMapValid = true;
+    } else if (stream_id != 0xbe  // padding_stream
+            && stream_id != 0xbf  // private_stream_2
+            && stream_id != 0xf0  // ECM
+            && stream_id != 0xf1  // EMM
+            && stream_id != 0xff  // program_stream_directory
+            && stream_id != 0xf2  // DSMCC
+            && stream_id != 0xf8) {  // H.222.1 type E
+        /* unsigned PES_marker_bits = */br.getBits(2);  // should be 0x2(hex)
+        /* unsigned PES_scrambling_control = */br.getBits(2);
+        /* unsigned PES_priority = */br.getBits(1);
+        /* unsigned data_alignment_indicator = */br.getBits(1);
+        /* unsigned copyright = */br.getBits(1);
+        /* unsigned original_or_copy = */br.getBits(1);
+
+        unsigned PTS_DTS_flags = br.getBits(2);
+        ALOGV("PTS_DTS_flags = %u", PTS_DTS_flags);
+
+        unsigned ESCR_flag = br.getBits(1);
+        ALOGV("ESCR_flag = %u", ESCR_flag);
+
+        unsigned ES_rate_flag = br.getBits(1);
+        ALOGV("ES_rate_flag = %u", ES_rate_flag);
+
+        unsigned DSM_trick_mode_flag = br.getBits(1);
+        ALOGV("DSM_trick_mode_flag = %u", DSM_trick_mode_flag);
+
+        unsigned additional_copy_info_flag = br.getBits(1);
+        ALOGV("additional_copy_info_flag = %u", additional_copy_info_flag);
+
+        /* unsigned PES_CRC_flag = */br.getBits(1);
+        /* PES_extension_flag = */br.getBits(1);
+
+        unsigned PES_header_data_length = br.getBits(8);
+        ALOGV("PES_header_data_length = %u", PES_header_data_length);
+
+        unsigned optional_bytes_remaining = PES_header_data_length;
+
+        uint64_t PTS = 0, DTS = 0;
+
+        if (PTS_DTS_flags == 2 || PTS_DTS_flags == 3) {
+            if (optional_bytes_remaining < 5u) {
+                return ERROR_MALFORMED;
+            }
+
+            if (br.getBits(4) != PTS_DTS_flags) {
+                return ERROR_MALFORMED;
+            }
+
+            PTS = ((uint64_t)br.getBits(3)) << 30;
+            if (br.getBits(1) != 1u) {
+                return ERROR_MALFORMED;
+            }
+            PTS |= ((uint64_t)br.getBits(15)) << 15;
+            if (br.getBits(1) != 1u) {
+                return ERROR_MALFORMED;
+            }
+            PTS |= br.getBits(15);
+            if (br.getBits(1) != 1u) {
+                return ERROR_MALFORMED;
+            }
+
+            ALOGV("PTS = %" PRIu64, PTS);
+            // ALOGI("PTS = %.2f secs", PTS / 90000.0f);
+
+            optional_bytes_remaining -= 5;
+
+            if (PTS_DTS_flags == 3) {
+                if (optional_bytes_remaining < 5u) {
+                    return ERROR_MALFORMED;
+                }
+
+                if (br.getBits(4) != 1u) {
+                    return ERROR_MALFORMED;
+                }
+
+                DTS = ((uint64_t)br.getBits(3)) << 30;
+                if (br.getBits(1) != 1u) {
+                    return ERROR_MALFORMED;
+                }
+                DTS |= ((uint64_t)br.getBits(15)) << 15;
+                if (br.getBits(1) != 1u) {
+                    return ERROR_MALFORMED;
+                }
+                DTS |= br.getBits(15);
+                if (br.getBits(1) != 1u) {
+                    return ERROR_MALFORMED;
+                }
+
+                ALOGV("DTS = %" PRIu64, DTS);
+
+                optional_bytes_remaining -= 5;
+            }
+        }
+
+        if (ESCR_flag) {
+            if (optional_bytes_remaining < 6u) {
+                return ERROR_MALFORMED;
+            }
+
+            br.getBits(2);
+
+            uint64_t ESCR = ((uint64_t)br.getBits(3)) << 30;
+            if (br.getBits(1) != 1u) {
+                return ERROR_MALFORMED;
+            }
+            ESCR |= ((uint64_t)br.getBits(15)) << 15;
+            if (br.getBits(1) != 1u) {
+                return ERROR_MALFORMED;
+            }
+            ESCR |= br.getBits(15);
+            if (br.getBits(1) != 1u) {
+                return ERROR_MALFORMED;
+            }
+
+            ALOGV("ESCR = %" PRIu64, ESCR);
+            /* unsigned ESCR_extension = */br.getBits(9);
+
+            if (br.getBits(1) != 1u) {
+                return ERROR_MALFORMED;
+            }
+
+            optional_bytes_remaining -= 6;
+        }
+
+        if (ES_rate_flag) {
+            if (optional_bytes_remaining < 3u) {
+                return ERROR_MALFORMED;
+            }
+
+            if (br.getBits(1) != 1u) {
+                return ERROR_MALFORMED;
+            }
+            /* unsigned ES_rate = */br.getBits(22);
+            if (br.getBits(1) != 1u) {
+                return ERROR_MALFORMED;
+            }
+
+            optional_bytes_remaining -= 3;
+        }
+
+        if (br.numBitsLeft() < optional_bytes_remaining * 8) {
+            return ERROR_MALFORMED;
+        }
+
+        br.skipBits(optional_bytes_remaining * 8);
+
+        // ES data follows.
+
+        if (PES_packet_length < PES_header_data_length + 3) {
+            return ERROR_MALFORMED;
+        }
+
+        unsigned dataLength =
+            PES_packet_length - 3 - PES_header_data_length;
+
+        if (br.numBitsLeft() < dataLength * 8) {
+            ALOGE("PES packet does not carry enough data to contain "
+                 "payload. (numBitsLeft = %zu, required = %u)",
+                 br.numBitsLeft(), dataLength * 8);
+
+            return ERROR_MALFORMED;
+        }
+
+        if (br.numBitsLeft() < dataLength * 8) {
+            return ERROR_MALFORMED;
+        }
+
+        ssize_t index = mTracks.indexOfKey(stream_id);
+        if (index < 0 && mScanning) {
+            unsigned streamType;
+
+            ssize_t streamTypeIndex;
+            if (mProgramStreamMapValid
+                    && (streamTypeIndex =
+                            mStreamTypeByESID.indexOfKey(stream_id)) >= 0) {
+                streamType = mStreamTypeByESID.valueAt(streamTypeIndex);
+            } else if ((stream_id & ~0x1f) == 0xc0) {
+                // ISO/IEC 13818-3 or ISO/IEC 11172-3 or ISO/IEC 13818-7
+                // or ISO/IEC 14496-3 audio
+                streamType = ATSParser::STREAMTYPE_MPEG2_AUDIO;
+            } else if ((stream_id & ~0x0f) == 0xe0) {
+                // ISO/IEC 13818-2 or ISO/IEC 11172-2 or ISO/IEC 14496-2 video
+                streamType = ATSParser::STREAMTYPE_MPEG2_VIDEO;
+            } else {
+                streamType = ATSParser::STREAMTYPE_RESERVED;
+            }
+
+            index = mTracks.add(
+                    stream_id, new Track(this, stream_id, streamType));
+        }
+
+        status_t err = OK;
+
+        if (index >= 0) {
+            err =
+                mTracks.editValueAt(index)->appendPESData(
+                    PTS_DTS_flags, PTS, DTS, br.data(), dataLength);
+        }
+
+        br.skipBits(dataLength * 8);
+
+        if (err != OK) {
+            return err;
+        }
+    } else if (stream_id == 0xbe) {  // padding_stream
+        if (PES_packet_length == 0u) {
+            return ERROR_MALFORMED;
+        }
+        br.skipBits(PES_packet_length * 8);
+    } else {
+        if (PES_packet_length == 0u) {
+            return ERROR_MALFORMED;
+        }
+        br.skipBits(PES_packet_length * 8);
+    }
+
+    return n;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+MPEG2PSExtractor::Track::Track(
+        MPEG2PSExtractor *extractor, unsigned stream_id, unsigned stream_type)
+    : mExtractor(extractor),
+      mStreamID(stream_id),
+      mStreamType(stream_type),
+      mQueue(NULL) {
+    bool supported = true;
+    ElementaryStreamQueue::Mode mode;
+
+    switch (mStreamType) {
+        case ATSParser::STREAMTYPE_H264:
+            mode = ElementaryStreamQueue::H264;
+            break;
+        case ATSParser::STREAMTYPE_MPEG2_AUDIO_ADTS:
+            mode = ElementaryStreamQueue::AAC;
+            break;
+        case ATSParser::STREAMTYPE_MPEG1_AUDIO:
+        case ATSParser::STREAMTYPE_MPEG2_AUDIO:
+            mode = ElementaryStreamQueue::MPEG_AUDIO;
+            break;
+
+        case ATSParser::STREAMTYPE_MPEG1_VIDEO:
+        case ATSParser::STREAMTYPE_MPEG2_VIDEO:
+            mode = ElementaryStreamQueue::MPEG_VIDEO;
+            break;
+
+        case ATSParser::STREAMTYPE_MPEG4_VIDEO:
+            mode = ElementaryStreamQueue::MPEG4_VIDEO;
+            break;
+
+        default:
+            supported = false;
+            break;
+    }
+
+    if (supported) {
+        mQueue = new ElementaryStreamQueue(mode);
+    } else {
+        ALOGI("unsupported stream ID 0x%02x", stream_id);
+    }
+}
+
+MPEG2PSExtractor::Track::~Track() {
+    delete mQueue;
+    mQueue = NULL;
+}
+
+status_t MPEG2PSExtractor::Track::start(MetaDataBase *) {
+    if (mSource == NULL) {
+        return NO_INIT;
+    }
+
+    return mSource->start(NULL); // AnotherPacketSource::start doesn't use its argument
+}
+
+status_t MPEG2PSExtractor::Track::stop() {
+    if (mSource == NULL) {
+        return NO_INIT;
+    }
+
+    return mSource->stop();
+}
+
+status_t MPEG2PSExtractor::Track::getFormat(MetaDataBase &meta) {
+    if (mSource == NULL) {
+        return NO_INIT;
+    }
+
+    sp<MetaData> sourceMeta = mSource->getFormat();
+    meta = *sourceMeta;
+    return OK;
+}
+
+status_t MPEG2PSExtractor::Track::read(
+        MediaBufferBase **buffer, const ReadOptions *options) {
+    if (mSource == NULL) {
+        return NO_INIT;
+    }
+
+    status_t finalResult;
+    while (!mSource->hasBufferAvailable(&finalResult)) {
+        if (finalResult != OK) {
+            return ERROR_END_OF_STREAM;
+        }
+
+        status_t err = mExtractor->feedMore();
+
+        if (err != OK) {
+            mSource->signalEOS(err);
+        }
+    }
+
+    return mSource->read(buffer, options);
+}
+
+status_t MPEG2PSExtractor::Track::appendPESData(
+        unsigned PTS_DTS_flags,
+        uint64_t PTS, uint64_t /* DTS */,
+        const uint8_t *data, size_t size) {
+    if (mQueue == NULL) {
+        return OK;
+    }
+
+    int64_t timeUs;
+    if (PTS_DTS_flags == 2 || PTS_DTS_flags == 3) {
+        timeUs = (PTS * 100) / 9;
+    } else {
+        timeUs = 0;
+    }
+
+    status_t err = mQueue->appendData(data, size, timeUs);
+
+    if (err != OK) {
+        return err;
+    }
+
+    sp<ABuffer> accessUnit;
+    while ((accessUnit = mQueue->dequeueAccessUnit()) != NULL) {
+        if (mSource == NULL) {
+            sp<MetaData> meta = mQueue->getFormat();
+
+            if (meta != NULL) {
+                ALOGV("Stream ID 0x%02x now has data.", mStreamID);
+
+                mSource = new AnotherPacketSource(meta);
+                mSource->queueAccessUnit(accessUnit);
+            }
+        } else if (mQueue->getFormat() != NULL) {
+            mSource->queueAccessUnit(accessUnit);
+        }
+    }
+
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+MPEG2PSExtractor::WrappedTrack::WrappedTrack(
+        MPEG2PSExtractor *extractor, const sp<Track> &track)
+    : mExtractor(extractor),
+      mTrack(track) {
+}
+
+MPEG2PSExtractor::WrappedTrack::~WrappedTrack() {
+}
+
+status_t MPEG2PSExtractor::WrappedTrack::start(MetaDataBase *params) {
+    return mTrack->start(params);
+}
+
+status_t MPEG2PSExtractor::WrappedTrack::stop() {
+    return mTrack->stop();
+}
+
+status_t MPEG2PSExtractor::WrappedTrack::getFormat(MetaDataBase &meta) {
+    return mTrack->getFormat(meta);
+}
+
+status_t MPEG2PSExtractor::WrappedTrack::read(
+        MediaBufferBase **buffer, const ReadOptions *options) {
+    return mTrack->read(buffer, options);
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+bool SniffMPEG2PS(
+        DataSourceBase *source, float *confidence) {
+    uint8_t header[5];
+    if (source->readAt(0, header, sizeof(header)) < (ssize_t)sizeof(header)) {
+        return false;
+    }
+
+    if (memcmp("\x00\x00\x01\xba", header, 4) || (header[4] >> 6) != 1) {
+        return false;
+    }
+
+    *confidence = 0.25f;  // Slightly larger than .mp3 extractor's confidence
+
+    return true;
+}
+
+}  // namespace android

diff --git a/media/extractors/mpeg2/MPEG2PSExtractor.h b/media/extractors/mpeg2/MPEG2PSExtractor.h
new file mode 100644
index 0000000..8b9dad9
--- /dev/null
+++ b/media/extractors/mpeg2/MPEG2PSExtractor.h

@@ -0,0 +1,80 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MPEG2_PS_EXTRACTOR_H_
+
+#define MPEG2_PS_EXTRACTOR_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/MediaExtractor.h>
+#include <media/stagefright/MetaDataBase.h>
+#include <utils/threads.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+struct ABuffer;
+struct AMessage;
+struct Track;
+class String8;
+
+struct MPEG2PSExtractor : public MediaExtractor {
+    explicit MPEG2PSExtractor(DataSourceBase *source);
+
+    virtual size_t countTracks();
+    virtual MediaTrack *getTrack(size_t index);
+    virtual status_t getTrackMetaData(MetaDataBase& meta, size_t index, uint32_t flags);
+
+    virtual status_t getMetaData(MetaDataBase& meta);
+
+    virtual uint32_t flags() const;
+    virtual const char * name() { return "MPEG2PSExtractor"; }
+
+protected:
+    virtual ~MPEG2PSExtractor();
+
+private:
+    struct Track;
+    struct WrappedTrack;
+
+    mutable Mutex mLock;
+    DataSourceBase *mDataSource;
+
+    off64_t mOffset;
+    status_t mFinalResult;
+    sp<ABuffer> mBuffer;
+    KeyedVector<unsigned, sp<Track> > mTracks;
+    bool mScanning;
+
+    bool mProgramStreamMapValid;
+    KeyedVector<unsigned, unsigned> mStreamTypeByESID;
+
+    status_t feedMore();
+
+    status_t dequeueChunk();
+    ssize_t dequeuePack();
+    ssize_t dequeueSystemHeader();
+    ssize_t dequeuePES();
+
+    DISALLOW_EVIL_CONSTRUCTORS(MPEG2PSExtractor);
+};
+
+bool SniffMPEG2PS(DataSourceBase *source, float *confidence);
+
+}  // namespace android
+
+#endif  // MPEG2_PS_EXTRACTOR_H_
+

diff --git a/media/extractors/mpeg2/MPEG2TSExtractor.cpp b/media/extractors/mpeg2/MPEG2TSExtractor.cpp
new file mode 100644
index 0000000..c83f7ce
--- /dev/null
+++ b/media/extractors/mpeg2/MPEG2TSExtractor.cpp

@@ -0,0 +1,669 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MPEG2TSExtractor"
+
+#include <inttypes.h>
+#include <utils/Log.h>
+
+#include "MPEG2TSExtractor.h"
+
+#include <media/DataSourceBase.h>
+#include <media/IStreamSource.h>
+#include <media/MediaTrack.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AUtils.h>
+#include <media/stagefright/foundation/MediaKeys.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <utils/String8.h>
+
+#include "mpeg2ts/AnotherPacketSource.h"
+#include "mpeg2ts/ATSParser.h"
+
+#include <hidl/HybridInterface.h>
+#include <android/hardware/cas/1.0/ICas.h>
+
+namespace android {
+
+using hardware::cas::V1_0::ICas;
+
+static const size_t kTSPacketSize = 188;
+static const int kMaxDurationReadSize = 250000LL;
+static const int kMaxDurationRetry = 6;
+
+struct MPEG2TSSource : public MediaTrack {
+    MPEG2TSSource(
+            MPEG2TSExtractor *extractor,
+            const sp<AnotherPacketSource> &impl,
+            bool doesSeek);
+    virtual ~MPEG2TSSource();
+
+    virtual status_t start(MetaDataBase *params = NULL);
+    virtual status_t stop();
+    virtual status_t getFormat(MetaDataBase &);
+
+    virtual status_t read(
+            MediaBufferBase **buffer, const ReadOptions *options = NULL);
+
+private:
+    MPEG2TSExtractor *mExtractor;
+    sp<AnotherPacketSource> mImpl;
+
+    // If there are both audio and video streams, only the video stream
+    // will signal seek on the extractor; otherwise the single stream will seek.
+    bool mDoesSeek;
+
+    DISALLOW_EVIL_CONSTRUCTORS(MPEG2TSSource);
+};
+
+MPEG2TSSource::MPEG2TSSource(
+        MPEG2TSExtractor *extractor,
+        const sp<AnotherPacketSource> &impl,
+        bool doesSeek)
+    : mExtractor(extractor),
+      mImpl(impl),
+      mDoesSeek(doesSeek) {
+}
+
+MPEG2TSSource::~MPEG2TSSource() {
+}
+
+status_t MPEG2TSSource::start(MetaDataBase *) {
+    return mImpl->start(NULL); // AnotherPacketSource::start() doesn't use its argument
+}
+
+status_t MPEG2TSSource::stop() {
+    return mImpl->stop();
+}
+
+status_t MPEG2TSSource::getFormat(MetaDataBase &meta) {
+    sp<MetaData> implMeta = mImpl->getFormat();
+    meta = *implMeta;
+    return OK;
+}
+
+status_t MPEG2TSSource::read(
+        MediaBufferBase **out, const ReadOptions *options) {
+    *out = NULL;
+
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode seekMode;
+    if (mDoesSeek && options && options->getSeekTo(&seekTimeUs, &seekMode)) {
+        // seek is needed
+        status_t err = mExtractor->seek(seekTimeUs, seekMode);
+        if (err != OK) {
+            return err;
+        }
+    }
+
+    if (mExtractor->feedUntilBufferAvailable(mImpl) != OK) {
+        return ERROR_END_OF_STREAM;
+    }
+
+    return mImpl->read(out, options);
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+MPEG2TSExtractor::MPEG2TSExtractor(DataSourceBase *source)
+    : mDataSource(source),
+      mParser(new ATSParser),
+      mLastSyncEvent(0),
+      mOffset(0) {
+    init();
+}
+
+size_t MPEG2TSExtractor::countTracks() {
+    return mSourceImpls.size();
+}
+
+MediaTrack *MPEG2TSExtractor::getTrack(size_t index) {
+    if (index >= mSourceImpls.size()) {
+        return NULL;
+    }
+
+    // The seek reference track (video if present; audio otherwise) performs
+    // seek requests, while other tracks ignore requests.
+    return new MPEG2TSSource(this, mSourceImpls.editItemAt(index),
+            (mSeekSyncPoints == &mSyncPoints.editItemAt(index)));
+}
+
+status_t MPEG2TSExtractor::getTrackMetaData(
+        MetaDataBase &meta,
+        size_t index, uint32_t /* flags */) {
+    sp<MetaData> implMeta = index < mSourceImpls.size()
+        ? mSourceImpls.editItemAt(index)->getFormat() : NULL;
+    if (implMeta == NULL) {
+        return UNKNOWN_ERROR;
+    }
+    meta = *implMeta;
+    return OK;
+}
+
+status_t MPEG2TSExtractor::getMetaData(MetaDataBase &meta) {
+    meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_CONTAINER_MPEG2TS);
+
+    return OK;
+}
+
+//static
+bool MPEG2TSExtractor::isScrambledFormat(MetaDataBase &format) {
+    const char *mime;
+    return format.findCString(kKeyMIMEType, &mime)
+            && (!strcasecmp(MEDIA_MIMETYPE_VIDEO_SCRAMBLED, mime)
+                    || !strcasecmp(MEDIA_MIMETYPE_AUDIO_SCRAMBLED, mime));
+}
+
+status_t MPEG2TSExtractor::setMediaCas(const uint8_t* casToken, size_t size) {
+    HalToken halToken;
+    halToken.setToExternal((uint8_t*)casToken, size);
+    sp<ICas> cas = ICas::castFrom(retrieveHalInterface(halToken));
+    ALOGD("setMediaCas: %p", cas.get());
+
+    status_t err = mParser->setMediaCas(cas);
+    if (err == OK) {
+        ALOGI("All tracks now have descramblers");
+        init();
+    }
+    return err;
+}
+
+void MPEG2TSExtractor::addSource(const sp<AnotherPacketSource> &impl) {
+    bool found = false;
+    for (size_t i = 0; i < mSourceImpls.size(); i++) {
+        if (mSourceImpls[i] == impl) {
+            found = true;
+            break;
+        }
+    }
+    if (!found) {
+        mSourceImpls.push(impl);
+    }
+}
+
+void MPEG2TSExtractor::init() {
+    bool haveAudio = false;
+    bool haveVideo = false;
+    int64_t startTime = ALooper::GetNowUs();
+
+    status_t err;
+    while ((err = feedMore(true /* isInit */)) == OK
+            || err == ERROR_DRM_DECRYPT_UNIT_NOT_INITIALIZED) {
+        if (haveAudio && haveVideo) {
+            addSyncPoint_l(mLastSyncEvent);
+            mLastSyncEvent.reset();
+            break;
+        }
+        if (!haveVideo) {
+            sp<AnotherPacketSource> impl = mParser->getSource(ATSParser::VIDEO);
+
+            if (impl != NULL) {
+                sp<MetaData> format = impl->getFormat();
+                if (format != NULL) {
+                    haveVideo = true;
+                    addSource(impl);
+                    if (!isScrambledFormat(*(format.get()))) {
+                        mSyncPoints.push();
+                        mSeekSyncPoints = &mSyncPoints.editTop();
+                    }
+                }
+            }
+        }
+
+        if (!haveAudio) {
+            sp<AnotherPacketSource> impl = mParser->getSource(ATSParser::AUDIO);
+
+            if (impl != NULL) {
+                sp<MetaData> format = impl->getFormat();
+                if (format != NULL) {
+                    haveAudio = true;
+                    addSource(impl);
+                    if (!isScrambledFormat(*(format.get()))) {
+                        mSyncPoints.push();
+                        if (!haveVideo) {
+                            mSeekSyncPoints = &mSyncPoints.editTop();
+                        }
+                    }
+                }
+            }
+        }
+
+        addSyncPoint_l(mLastSyncEvent);
+        mLastSyncEvent.reset();
+
+        // ERROR_DRM_DECRYPT_UNIT_NOT_INITIALIZED is returned when the mpeg2ts
+        // is scrambled but we don't have a MediaCas object set. The extraction
+        // will only continue when setMediaCas() is called successfully.
+        if (err == ERROR_DRM_DECRYPT_UNIT_NOT_INITIALIZED) {
+            ALOGI("stopped parsing scrambled content, "
+                  "haveAudio=%d, haveVideo=%d, elaspedTime=%" PRId64,
+                    haveAudio, haveVideo, ALooper::GetNowUs() - startTime);
+            return;
+        }
+
+        // Wait only for 2 seconds to detect audio/video streams.
+        if (ALooper::GetNowUs() - startTime > 2000000ll) {
+            break;
+        }
+    }
+
+    off64_t size;
+    if (mDataSource->getSize(&size) == OK && (haveAudio || haveVideo)) {
+        sp<AnotherPacketSource> impl = haveVideo
+                ? mParser->getSource(ATSParser::VIDEO)
+                : mParser->getSource(ATSParser::AUDIO);
+        size_t prevSyncSize = 1;
+        int64_t durationUs = -1;
+        List<int64_t> durations;
+        // Estimate duration --- stabilize until you get <500ms deviation.
+        while (feedMore() == OK
+                && ALooper::GetNowUs() - startTime <= 2000000ll) {
+            if (mSeekSyncPoints->size() > prevSyncSize) {
+                prevSyncSize = mSeekSyncPoints->size();
+                int64_t diffUs = mSeekSyncPoints->keyAt(prevSyncSize - 1)
+                        - mSeekSyncPoints->keyAt(0);
+                off64_t diffOffset = mSeekSyncPoints->valueAt(prevSyncSize - 1)
+                        - mSeekSyncPoints->valueAt(0);
+                int64_t currentDurationUs = size * diffUs / diffOffset;
+                durations.push_back(currentDurationUs);
+                if (durations.size() > 5) {
+                    durations.erase(durations.begin());
+                    int64_t min = *durations.begin();
+                    int64_t max = *durations.begin();
+                    for (auto duration : durations) {
+                        if (min > duration) {
+                            min = duration;
+                        }
+                        if (max < duration) {
+                            max = duration;
+                        }
+                    }
+                    if (max - min < 500 * 1000) {
+                        durationUs = currentDurationUs;
+                        break;
+                    }
+                }
+            }
+        }
+        status_t err;
+        int64_t bufferedDurationUs;
+        bufferedDurationUs = impl->getBufferedDurationUs(&err);
+        if (err == ERROR_END_OF_STREAM) {
+            durationUs = bufferedDurationUs;
+        }
+        if (durationUs > 0) {
+            const sp<MetaData> meta = impl->getFormat();
+            meta->setInt64(kKeyDuration, durationUs);
+            impl->setFormat(meta);
+        } else {
+            estimateDurationsFromTimesUsAtEnd();
+        }
+    }
+
+    ALOGI("haveAudio=%d, haveVideo=%d, elaspedTime=%" PRId64,
+            haveAudio, haveVideo, ALooper::GetNowUs() - startTime);
+}
+
+status_t MPEG2TSExtractor::feedMore(bool isInit) {
+    Mutex::Autolock autoLock(mLock);
+
+    uint8_t packet[kTSPacketSize];
+    ssize_t n = mDataSource->readAt(mOffset, packet, kTSPacketSize);
+
+    if (n < (ssize_t)kTSPacketSize) {
+        if (n >= 0) {
+            mParser->signalEOS(ERROR_END_OF_STREAM);
+        }
+        return (n < 0) ? (status_t)n : ERROR_END_OF_STREAM;
+    }
+
+    ATSParser::SyncEvent event(mOffset);
+    mOffset += n;
+    status_t err = mParser->feedTSPacket(packet, kTSPacketSize, &event);
+    if (event.hasReturnedData()) {
+        if (isInit) {
+            mLastSyncEvent = event;
+        } else {
+            addSyncPoint_l(event);
+        }
+    }
+    return err;
+}
+
+void MPEG2TSExtractor::addSyncPoint_l(const ATSParser::SyncEvent &event) {
+    if (!event.hasReturnedData()) {
+        return;
+    }
+
+    for (size_t i = 0; i < mSourceImpls.size(); ++i) {
+        if (mSourceImpls[i].get() == event.getMediaSource().get()) {
+            KeyedVector<int64_t, off64_t> *syncPoints = &mSyncPoints.editItemAt(i);
+            syncPoints->add(event.getTimeUs(), event.getOffset());
+            // We're keeping the size of the sync points at most 5mb per a track.
+            size_t size = syncPoints->size();
+            if (size >= 327680) {
+                int64_t firstTimeUs = syncPoints->keyAt(0);
+                int64_t lastTimeUs = syncPoints->keyAt(size - 1);
+                if (event.getTimeUs() - firstTimeUs > lastTimeUs - event.getTimeUs()) {
+                    syncPoints->removeItemsAt(0, 4096);
+                } else {
+                    syncPoints->removeItemsAt(size - 4096, 4096);
+                }
+            }
+            break;
+        }
+    }
+}
+
+status_t MPEG2TSExtractor::estimateDurationsFromTimesUsAtEnd()  {
+    if (!(mDataSource->flags() & DataSourceBase::kIsLocalFileSource)) {
+        return ERROR_UNSUPPORTED;
+    }
+
+    off64_t size = 0;
+    status_t err = mDataSource->getSize(&size);
+    if (err != OK) {
+        return err;
+    }
+
+    uint8_t packet[kTSPacketSize];
+    const off64_t zero = 0;
+    off64_t offset = max(zero, size - kMaxDurationReadSize);
+    if (mDataSource->readAt(offset, &packet, 0) < 0) {
+        return ERROR_IO;
+    }
+
+    int retry = 0;
+    bool allDurationsFound = false;
+    int64_t timeAnchorUs = mParser->getFirstPTSTimeUs();
+    do {
+        int bytesRead = 0;
+        sp<ATSParser> parser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE);
+        ATSParser::SyncEvent ev(0);
+        offset = max(zero, size - (kMaxDurationReadSize << retry));
+        offset = (offset / kTSPacketSize) * kTSPacketSize;
+        for (;;) {
+            if (bytesRead >= kMaxDurationReadSize << max(0, retry - 1)) {
+                break;
+            }
+
+            ssize_t n = mDataSource->readAt(offset, packet, kTSPacketSize);
+            if (n < 0) {
+                return n;
+            } else if (n < (ssize_t)kTSPacketSize) {
+                break;
+            }
+
+            offset += kTSPacketSize;
+            bytesRead += kTSPacketSize;
+            err = parser->feedTSPacket(packet, kTSPacketSize, &ev);
+            if (err != OK) {
+                return err;
+            }
+
+            if (ev.hasReturnedData()) {
+                int64_t durationUs = ev.getTimeUs();
+                ATSParser::SourceType type = ev.getType();
+                ev.reset();
+
+                int64_t firstTimeUs;
+                sp<AnotherPacketSource> src = mParser->getSource(type);
+                if (src == NULL || src->nextBufferTime(&firstTimeUs) != OK) {
+                    continue;
+                }
+                durationUs += src->getEstimatedBufferDurationUs();
+                durationUs -= timeAnchorUs;
+                durationUs -= firstTimeUs;
+                if (durationUs > 0) {
+                    int64_t origDurationUs, lastDurationUs;
+                    const sp<MetaData> meta = src->getFormat();
+                    const uint32_t kKeyLastDuration = 'ldur';
+                    // Require two consecutive duration calculations to be within 1 sec before
+                    // updating; use MetaData to store previous duration estimate in per-stream
+                    // context.
+                    if (!meta->findInt64(kKeyDuration, &origDurationUs)
+                            || !meta->findInt64(kKeyLastDuration, &lastDurationUs)
+                            || (origDurationUs < durationUs
+                             && abs(durationUs - lastDurationUs) < 60000000)) {
+                        meta->setInt64(kKeyDuration, durationUs);
+                    }
+                    meta->setInt64(kKeyLastDuration, durationUs);
+                }
+            }
+        }
+
+        if (!allDurationsFound) {
+            allDurationsFound = true;
+            for (auto t: {ATSParser::VIDEO, ATSParser::AUDIO}) {
+                sp<AnotherPacketSource> src = mParser->getSource(t);
+                if (src == NULL) {
+                    continue;
+                }
+                int64_t durationUs;
+                const sp<MetaData> meta = src->getFormat();
+                if (!meta->findInt64(kKeyDuration, &durationUs)) {
+                    allDurationsFound = false;
+                    break;
+                }
+            }
+        }
+
+        ++retry;
+    } while(!allDurationsFound && offset > 0 && retry <= kMaxDurationRetry);
+
+    return allDurationsFound? OK : ERROR_UNSUPPORTED;
+}
+
+uint32_t MPEG2TSExtractor::flags() const {
+    return CAN_PAUSE | CAN_SEEK_BACKWARD | CAN_SEEK_FORWARD;
+}
+
+status_t MPEG2TSExtractor::seek(int64_t seekTimeUs,
+        const MediaTrack::ReadOptions::SeekMode &seekMode) {
+    if (mSeekSyncPoints == NULL || mSeekSyncPoints->isEmpty()) {
+        ALOGW("No sync point to seek to.");
+        // ... and therefore we have nothing useful to do here.
+        return OK;
+    }
+
+    // Determine whether we're seeking beyond the known area.
+    bool shouldSeekBeyond =
+            (seekTimeUs > mSeekSyncPoints->keyAt(mSeekSyncPoints->size() - 1));
+
+    // Determine the sync point to seek.
+    size_t index = 0;
+    for (; index < mSeekSyncPoints->size(); ++index) {
+        int64_t timeUs = mSeekSyncPoints->keyAt(index);
+        if (timeUs > seekTimeUs) {
+            break;
+        }
+    }
+
+    switch (seekMode) {
+        case MediaTrack::ReadOptions::SEEK_NEXT_SYNC:
+            if (index == mSeekSyncPoints->size()) {
+                ALOGW("Next sync not found; starting from the latest sync.");
+                --index;
+            }
+            break;
+        case MediaTrack::ReadOptions::SEEK_CLOSEST_SYNC:
+        case MediaTrack::ReadOptions::SEEK_CLOSEST:
+            ALOGW("seekMode not supported: %d; falling back to PREVIOUS_SYNC",
+                    seekMode);
+            // fall-through
+        case MediaTrack::ReadOptions::SEEK_PREVIOUS_SYNC:
+            if (index == 0) {
+                ALOGW("Previous sync not found; starting from the earliest "
+                        "sync.");
+            } else {
+                --index;
+            }
+            break;
+        default:
+            return ERROR_UNSUPPORTED;
+    }
+    if (!shouldSeekBeyond || mOffset <= mSeekSyncPoints->valueAt(index)) {
+        int64_t actualSeekTimeUs = mSeekSyncPoints->keyAt(index);
+        mOffset = mSeekSyncPoints->valueAt(index);
+        status_t err = queueDiscontinuityForSeek(actualSeekTimeUs);
+        if (err != OK) {
+            return err;
+        }
+    }
+
+    if (shouldSeekBeyond) {
+        status_t err = seekBeyond(seekTimeUs);
+        if (err != OK) {
+            return err;
+        }
+    }
+
+    // Fast-forward to sync frame.
+    for (size_t i = 0; i < mSourceImpls.size(); ++i) {
+        const sp<AnotherPacketSource> &impl = mSourceImpls[i];
+        status_t err;
+        feedUntilBufferAvailable(impl);
+        while (impl->hasBufferAvailable(&err)) {
+            sp<AMessage> meta = impl->getMetaAfterLastDequeued(0);
+            sp<ABuffer> buffer;
+            if (meta == NULL) {
+                return UNKNOWN_ERROR;
+            }
+            int32_t sync;
+            if (meta->findInt32("isSync", &sync) && sync) {
+                break;
+            }
+            err = impl->dequeueAccessUnit(&buffer);
+            if (err != OK) {
+                return err;
+            }
+            feedUntilBufferAvailable(impl);
+        }
+    }
+
+    return OK;
+}
+
+status_t MPEG2TSExtractor::queueDiscontinuityForSeek(int64_t actualSeekTimeUs) {
+    // Signal discontinuity
+    sp<AMessage> extra(new AMessage);
+    extra->setInt64(kATSParserKeyMediaTimeUs, actualSeekTimeUs);
+    mParser->signalDiscontinuity(ATSParser::DISCONTINUITY_TIME, extra);
+
+    // After discontinuity, impl should only have discontinuities
+    // with the last being what we queued. Dequeue them all here.
+    for (size_t i = 0; i < mSourceImpls.size(); ++i) {
+        const sp<AnotherPacketSource> &impl = mSourceImpls.itemAt(i);
+        sp<ABuffer> buffer;
+        status_t err;
+        while (impl->hasBufferAvailable(&err)) {
+            if (err != OK) {
+                return err;
+            }
+            err = impl->dequeueAccessUnit(&buffer);
+            // If the source contains anything but discontinuity, that's
+            // a programming mistake.
+            CHECK(err == INFO_DISCONTINUITY);
+        }
+    }
+
+    // Feed until we have a buffer for each source.
+    for (size_t i = 0; i < mSourceImpls.size(); ++i) {
+        const sp<AnotherPacketSource> &impl = mSourceImpls.itemAt(i);
+        sp<ABuffer> buffer;
+        status_t err = feedUntilBufferAvailable(impl);
+        if (err != OK) {
+            return err;
+        }
+    }
+
+    return OK;
+}
+
+status_t MPEG2TSExtractor::seekBeyond(int64_t seekTimeUs) {
+    // If we're seeking beyond where we know --- read until we reach there.
+    size_t syncPointsSize = mSeekSyncPoints->size();
+
+    while (seekTimeUs > mSeekSyncPoints->keyAt(
+            mSeekSyncPoints->size() - 1)) {
+        status_t err;
+        if (syncPointsSize < mSeekSyncPoints->size()) {
+            syncPointsSize = mSeekSyncPoints->size();
+            int64_t syncTimeUs = mSeekSyncPoints->keyAt(syncPointsSize - 1);
+            // Dequeue buffers before sync point in order to avoid too much
+            // cache building up.
+            sp<ABuffer> buffer;
+            for (size_t i = 0; i < mSourceImpls.size(); ++i) {
+                const sp<AnotherPacketSource> &impl = mSourceImpls[i];
+                int64_t timeUs;
+                while ((err = impl->nextBufferTime(&timeUs)) == OK) {
+                    if (timeUs < syncTimeUs) {
+                        impl->dequeueAccessUnit(&buffer);
+                    } else {
+                        break;
+                    }
+                }
+                if (err != OK && err != -EWOULDBLOCK) {
+                    return err;
+                }
+            }
+        }
+        if (feedMore() != OK) {
+            return ERROR_END_OF_STREAM;
+        }
+    }
+
+    return OK;
+}
+
+status_t MPEG2TSExtractor::feedUntilBufferAvailable(
+        const sp<AnotherPacketSource> &impl) {
+    status_t finalResult;
+    while (!impl->hasBufferAvailable(&finalResult)) {
+        if (finalResult != OK) {
+            return finalResult;
+        }
+
+        status_t err = feedMore();
+        if (err != OK) {
+            impl->signalEOS(err);
+        }
+    }
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+bool SniffMPEG2TS(DataSourceBase *source, float *confidence) {
+    for (int i = 0; i < 5; ++i) {
+        char header;
+        if (source->readAt(kTSPacketSize * i, &header, 1) != 1
+                || header != 0x47) {
+            return false;
+        }
+    }
+
+    *confidence = 0.1f;
+
+    return true;
+}
+
+}  // namespace android

diff --git a/media/extractors/mpeg2/MPEG2TSExtractor.h b/media/extractors/mpeg2/MPEG2TSExtractor.h
new file mode 100644
index 0000000..cbdd3cb
--- /dev/null
+++ b/media/extractors/mpeg2/MPEG2TSExtractor.h

@@ -0,0 +1,108 @@
+/*
+
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MPEG2_TS_EXTRACTOR_H_
+
+#define MPEG2_TS_EXTRACTOR_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/MediaExtractor.h>
+#include <media/MediaTrack.h>
+#include <media/stagefright/MetaDataBase.h>
+#include <utils/threads.h>
+#include <utils/KeyedVector.h>
+#include <utils/Vector.h>
+
+#include "mpeg2ts/ATSParser.h"
+
+namespace android {
+
+struct AMessage;
+struct AnotherPacketSource;
+struct ATSParser;
+class DataSourceBase;
+struct MPEG2TSSource;
+class String8;
+
+struct MPEG2TSExtractor : public MediaExtractor {
+    explicit MPEG2TSExtractor(DataSourceBase *source);
+
+    virtual size_t countTracks();
+    virtual MediaTrack *getTrack(size_t index);
+    virtual status_t getTrackMetaData(MetaDataBase &meta, size_t index, uint32_t flags);
+
+    virtual status_t getMetaData(MetaDataBase& meta);
+
+    virtual status_t setMediaCas(const uint8_t* /*casToken*/, size_t /*size*/) override;
+
+    virtual uint32_t flags() const;
+    virtual const char * name() { return "MPEG2TSExtractor"; }
+
+private:
+    friend struct MPEG2TSSource;
+
+    mutable Mutex mLock;
+
+    DataSourceBase *mDataSource;
+
+    sp<ATSParser> mParser;
+
+    // Used to remember SyncEvent occurred in feedMore() when called from init(),
+    // because init() needs to update |mSourceImpls| before adding SyncPoint.
+    ATSParser::SyncEvent mLastSyncEvent;
+
+    Vector<sp<AnotherPacketSource> > mSourceImpls;
+
+    Vector<KeyedVector<int64_t, off64_t> > mSyncPoints;
+    // Sync points used for seeking --- normally one for video track is used.
+    // If no video track is present, audio track will be used instead.
+    KeyedVector<int64_t, off64_t> *mSeekSyncPoints;
+
+    off64_t mOffset;
+
+    static bool isScrambledFormat(MetaDataBase &format);
+
+    void init();
+    void addSource(const sp<AnotherPacketSource> &impl);
+    // Try to feed more data from source to parser.
+    // |isInit| means this function is called inside init(). This is a signal to
+    // save SyncEvent so that init() can add SyncPoint after it updates |mSourceImpls|.
+    // This function returns OK if expected amount of data is fed from DataSourceBase to
+    // parser and is successfully parsed. Otherwise, various error codes could be
+    // returned, e.g., ERROR_END_OF_STREAM, or no data availalbe from DataSourceBase, or
+    // the data has syntax error during parsing, etc.
+    status_t feedMore(bool isInit = false);
+    status_t seek(int64_t seekTimeUs,
+            const MediaSource::ReadOptions::SeekMode& seekMode);
+    status_t queueDiscontinuityForSeek(int64_t actualSeekTimeUs);
+    status_t seekBeyond(int64_t seekTimeUs);
+
+    status_t feedUntilBufferAvailable(const sp<AnotherPacketSource> &impl);
+
+    // Add a SynPoint derived from |event|.
+    void addSyncPoint_l(const ATSParser::SyncEvent &event);
+
+    status_t  estimateDurationsFromTimesUsAtEnd();
+
+    DISALLOW_EVIL_CONSTRUCTORS(MPEG2TSExtractor);
+};
+
+bool SniffMPEG2TS(DataSourceBase *source, float *confidence);
+
+}  // namespace android
+
+#endif  // MPEG2_TS_EXTRACTOR_H_

diff --git a/media/libstagefright/matroska/NOTICE b/media/extractors/mpeg2/NOTICE
similarity index 100%
copy from media/libstagefright/matroska/NOTICE
copy to media/extractors/mpeg2/NOTICE


diff --git a/media/extractors/mpeg2/exports.lds b/media/extractors/mpeg2/exports.lds
new file mode 100644
index 0000000..b1309ee
--- /dev/null
+++ b/media/extractors/mpeg2/exports.lds

@@ -0,0 +1 @@
+{ global: GETEXTRACTORDEF; local: *; };

diff --git a/media/extractors/ogg/Android.bp b/media/extractors/ogg/Android.bp
new file mode 100644
index 0000000..7c6fc75
--- /dev/null
+++ b/media/extractors/ogg/Android.bp

@@ -0,0 +1,44 @@
+cc_library_shared {
+
+    srcs: ["OggExtractor.cpp"],
+
+    include_dirs: [
+        "frameworks/av/media/libstagefright/include",
+        "external/tremolo",
+    ],
+
+    shared_libs: [
+        "liblog",
+        "libmediaextractor",
+    ],
+
+    static_libs: [
+        "libstagefright_foundation",
+        "libutils",
+        "libvorbisidec",
+    ],
+
+    name: "liboggextractor",
+    relative_install_path: "extractors",
+
+    compile_multilib: "first",
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+        "-fvisibility=hidden",
+    ],
+    version_script: "exports.lds",
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+        diag: {
+            cfi: true,
+        },
+    },
+
+}

diff --git a/media/libstagefright/matroska/MODULE_LICENSE_APACHE2 b/media/extractors/ogg/MODULE_LICENSE_APACHE2
similarity index 100%
copy from media/libstagefright/matroska/MODULE_LICENSE_APACHE2
copy to media/extractors/ogg/MODULE_LICENSE_APACHE2


diff --git a/media/libstagefright/matroska/NOTICE b/media/extractors/ogg/NOTICE
similarity index 100%
copy from media/libstagefright/matroska/NOTICE
copy to media/extractors/ogg/NOTICE


diff --git a/media/extractors/ogg/OggExtractor.cpp b/media/extractors/ogg/OggExtractor.cpp
new file mode 100644
index 0000000..b2fe69c
--- /dev/null
+++ b/media/extractors/ogg/OggExtractor.cpp

@@ -0,0 +1,1287 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "OggExtractor"
+#include <utils/Log.h>
+
+#include "OggExtractor.h"
+
+#include <cutils/properties.h>
+#include <media/DataSourceBase.h>
+#include <media/ExtractorUtils.h>
+#include <media/MediaTrack.h>
+#include <media/VorbisComment.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/base64.h>
+#include <media/stagefright/foundation/ByteUtils.h>
+#include <media/stagefright/MediaBufferBase.h>
+#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaDataBase.h>
+#include <utils/String8.h>
+
+extern "C" {
+    #include <Tremolo/codec_internal.h>
+
+    int _vorbis_unpack_books(vorbis_info *vi,oggpack_buffer *opb);
+    int _vorbis_unpack_info(vorbis_info *vi,oggpack_buffer *opb);
+    int _vorbis_unpack_comment(vorbis_comment *vc,oggpack_buffer *opb);
+    long vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op);
+}
+
+namespace android {
+
+struct OggSource : public MediaTrack {
+    explicit OggSource(OggExtractor *extractor);
+
+    virtual status_t getFormat(MetaDataBase &);
+
+    virtual status_t start(MetaDataBase *params = NULL);
+    virtual status_t stop();
+
+    virtual status_t read(
+            MediaBufferBase **buffer, const ReadOptions *options = NULL);
+
+protected:
+    virtual ~OggSource();
+
+private:
+    OggExtractor *mExtractor;
+    bool mStarted;
+
+    OggSource(const OggSource &);
+    OggSource &operator=(const OggSource &);
+};
+
+struct MyOggExtractor {
+    MyOggExtractor(
+            DataSourceBase *source,
+            const char *mimeType,
+            size_t numHeaders,
+            int64_t seekPreRollUs);
+    virtual ~MyOggExtractor();
+
+    status_t getFormat(MetaDataBase &) const;
+
+    // Returns an approximate bitrate in bits per second.
+    virtual uint64_t approxBitrate() const = 0;
+
+    status_t seekToTime(int64_t timeUs);
+    status_t seekToOffset(off64_t offset);
+    virtual status_t readNextPacket(MediaBufferBase **buffer) = 0;
+
+    status_t init();
+
+    status_t getFileMetaData(MetaDataBase &meta) {
+        meta = mFileMeta;
+        return OK;
+    }
+
+protected:
+    struct Page {
+        uint64_t mGranulePosition;
+        int32_t mPrevPacketSize;
+        uint64_t mPrevPacketPos;
+        uint32_t mSerialNo;
+        uint32_t mPageNo;
+        uint8_t mFlags;
+        uint8_t mNumSegments;
+        uint8_t mLace[255];
+    };
+
+    struct TOCEntry {
+        off64_t mPageOffset;
+        int64_t mTimeUs;
+    };
+
+    DataSourceBase *mSource;
+    off64_t mOffset;
+    Page mCurrentPage;
+    uint64_t mCurGranulePosition;
+    uint64_t mPrevGranulePosition;
+    size_t mCurrentPageSize;
+    bool mFirstPacketInPage;
+    uint64_t mCurrentPageSamples;
+    size_t mNextLaceIndex;
+
+    const char *mMimeType;
+    size_t mNumHeaders;
+    int64_t mSeekPreRollUs;
+
+    off64_t mFirstDataOffset;
+
+    vorbis_info mVi;
+    vorbis_comment mVc;
+
+    MetaDataBase mMeta;
+    MetaDataBase mFileMeta;
+
+    Vector<TOCEntry> mTableOfContents;
+
+    ssize_t readPage(off64_t offset, Page *page);
+    status_t findNextPage(off64_t startOffset, off64_t *pageOffset);
+
+    virtual int64_t getTimeUsOfGranule(uint64_t granulePos) const = 0;
+
+    // Extract codec format, metadata tags, and various codec specific data;
+    // the format and CSD's are required to setup the decoders for the enclosed media content.
+    //
+    // Valid values for `type` are:
+    // 1 - bitstream identification header
+    // 3 - comment header
+    // 5 - codec setup header (Vorbis only)
+    virtual status_t verifyHeader(MediaBufferBase *buffer, uint8_t type) = 0;
+
+    // Read the next ogg packet from the underlying data source; optionally
+    // calculate the timestamp for the output packet whilst pretending
+    // that we are parsing an Ogg Vorbis stream.
+    //
+    // *buffer is NULL'ed out immediately upon entry, and if successful a new buffer is allocated;
+    // clients are responsible for releasing the original buffer.
+    status_t _readNextPacket(MediaBufferBase **buffer, bool calcVorbisTimestamp);
+
+    int32_t getPacketBlockSize(MediaBufferBase *buffer);
+
+    void parseFileMetaData();
+
+    status_t findPrevGranulePosition(off64_t pageOffset, uint64_t *granulePos);
+
+    void buildTableOfContents();
+
+    MyOggExtractor(const MyOggExtractor &);
+    MyOggExtractor &operator=(const MyOggExtractor &);
+};
+
+struct MyVorbisExtractor : public MyOggExtractor {
+    explicit MyVorbisExtractor(DataSourceBase *source)
+        : MyOggExtractor(source,
+                MEDIA_MIMETYPE_AUDIO_VORBIS,
+                /* numHeaders */ 3,
+                /* seekPreRollUs */ 0) {
+    }
+
+    virtual uint64_t approxBitrate() const;
+
+    virtual status_t readNextPacket(MediaBufferBase **buffer) {
+        return _readNextPacket(buffer, /* calcVorbisTimestamp = */ true);
+    }
+
+protected:
+    virtual int64_t getTimeUsOfGranule(uint64_t granulePos) const {
+        if (granulePos > INT64_MAX / 1000000ll) {
+            return INT64_MAX;
+        }
+        return granulePos * 1000000ll / mVi.rate;
+    }
+
+    virtual status_t verifyHeader(MediaBufferBase *buffer, uint8_t type);
+};
+
+struct MyOpusExtractor : public MyOggExtractor {
+    static const int32_t kOpusSampleRate = 48000;
+    static const int64_t kOpusSeekPreRollUs = 80000; // 80 ms
+
+    explicit MyOpusExtractor(DataSourceBase *source)
+        : MyOggExtractor(source, MEDIA_MIMETYPE_AUDIO_OPUS, /*numHeaders*/ 2, kOpusSeekPreRollUs),
+          mChannelCount(0),
+          mCodecDelay(0),
+          mStartGranulePosition(-1) {
+    }
+
+    virtual uint64_t approxBitrate() const {
+        return 0;
+    }
+
+    virtual status_t readNextPacket(MediaBufferBase **buffer);
+
+protected:
+    virtual int64_t getTimeUsOfGranule(uint64_t granulePos) const;
+    virtual status_t verifyHeader(MediaBufferBase *buffer, uint8_t type);
+
+private:
+    status_t verifyOpusHeader(MediaBufferBase *buffer);
+    status_t verifyOpusComments(MediaBufferBase *buffer);
+    uint32_t getNumSamplesInPacket(MediaBufferBase *buffer) const;
+
+    uint8_t mChannelCount;
+    uint16_t mCodecDelay;
+    int64_t mStartGranulePosition;
+};
+
+////////////////////////////////////////////////////////////////////////////////
+
+OggSource::OggSource(OggExtractor *extractor)
+    : mExtractor(extractor),
+      mStarted(false) {
+}
+
+OggSource::~OggSource() {
+    if (mStarted) {
+        stop();
+    }
+}
+
+status_t OggSource::getFormat(MetaDataBase &meta) {
+    return mExtractor->mImpl->getFormat(meta);
+}
+
+status_t OggSource::start(MetaDataBase * /* params */) {
+    if (mStarted) {
+        return INVALID_OPERATION;
+    }
+
+    mStarted = true;
+
+    return OK;
+}
+
+status_t OggSource::stop() {
+    mStarted = false;
+
+    return OK;
+}
+
+status_t OggSource::read(
+        MediaBufferBase **out, const ReadOptions *options) {
+    *out = NULL;
+
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode mode;
+    if (options && options->getSeekTo(&seekTimeUs, &mode)) {
+        status_t err = mExtractor->mImpl->seekToTime(seekTimeUs);
+        if (err != OK) {
+            return err;
+        }
+    }
+
+    MediaBufferBase *packet;
+    status_t err = mExtractor->mImpl->readNextPacket(&packet);
+
+    if (err != OK) {
+        return err;
+    }
+
+#if 0
+    int64_t timeUs;
+    if (packet->meta_data().findInt64(kKeyTime, &timeUs)) {
+        ALOGI("found time = %lld us", timeUs);
+    } else {
+        ALOGI("NO time");
+    }
+#endif
+
+    packet->meta_data().setInt32(kKeyIsSyncFrame, 1);
+
+    *out = packet;
+
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+MyOggExtractor::MyOggExtractor(
+        DataSourceBase *source,
+        const char *mimeType,
+        size_t numHeaders,
+        int64_t seekPreRollUs)
+    : mSource(source),
+      mOffset(0),
+      mCurGranulePosition(0),
+      mPrevGranulePosition(0),
+      mCurrentPageSize(0),
+      mFirstPacketInPage(true),
+      mCurrentPageSamples(0),
+      mNextLaceIndex(0),
+      mMimeType(mimeType),
+      mNumHeaders(numHeaders),
+      mSeekPreRollUs(seekPreRollUs),
+      mFirstDataOffset(-1) {
+    mCurrentPage.mNumSegments = 0;
+
+    vorbis_info_init(&mVi);
+    vorbis_comment_init(&mVc);
+}
+
+MyOggExtractor::~MyOggExtractor() {
+    vorbis_comment_clear(&mVc);
+    vorbis_info_clear(&mVi);
+}
+
+status_t MyOggExtractor::getFormat(MetaDataBase &meta) const {
+    meta = mMeta;
+    return OK;
+}
+
+status_t MyOggExtractor::findNextPage(
+        off64_t startOffset, off64_t *pageOffset) {
+    *pageOffset = startOffset;
+
+    for (;;) {
+        char signature[4];
+        ssize_t n = mSource->readAt(*pageOffset, &signature, 4);
+
+        if (n < 4) {
+            *pageOffset = 0;
+
+            return (n < 0) ? n : (status_t)ERROR_END_OF_STREAM;
+        }
+
+        if (!memcmp(signature, "OggS", 4)) {
+            if (*pageOffset > startOffset) {
+                ALOGV("skipped %lld bytes of junk to reach next frame",
+                     (long long)(*pageOffset - startOffset));
+            }
+
+            return OK;
+        }
+
+        ++*pageOffset;
+    }
+}
+
+// Given the offset of the "current" page, find the page immediately preceding
+// it (if any) and return its granule position.
+// To do this we back up from the "current" page's offset until we find any
+// page preceding it and then scan forward to just before the current page.
+status_t MyOggExtractor::findPrevGranulePosition(
+        off64_t pageOffset, uint64_t *granulePos) {
+    *granulePos = 0;
+
+    off64_t prevPageOffset = 0;
+    off64_t prevGuess = pageOffset;
+    for (;;) {
+        if (prevGuess >= 5000) {
+            prevGuess -= 5000;
+        } else {
+            prevGuess = 0;
+        }
+
+        ALOGV("backing up %lld bytes", (long long)(pageOffset - prevGuess));
+
+        status_t err = findNextPage(prevGuess, &prevPageOffset);
+        if (err == ERROR_END_OF_STREAM) {
+            // We are at the last page and didn't back off enough;
+            // back off 5000 bytes more and try again.
+            continue;
+        } else if (err != OK) {
+            return err;
+        }
+
+        if (prevPageOffset < pageOffset || prevGuess == 0) {
+            break;
+        }
+    }
+
+    if (prevPageOffset == pageOffset) {
+        // We did not find a page preceding this one.
+        return UNKNOWN_ERROR;
+    }
+
+    ALOGV("prevPageOffset at %lld, pageOffset at %lld",
+            (long long)prevPageOffset, (long long)pageOffset);
+
+    for (;;) {
+        Page prevPage;
+        ssize_t n = readPage(prevPageOffset, &prevPage);
+
+        if (n <= 0) {
+            return (status_t)n;
+        }
+
+        prevPageOffset += n;
+
+        if (prevPageOffset == pageOffset) {
+            *granulePos = prevPage.mGranulePosition;
+            return OK;
+        }
+    }
+}
+
+status_t MyOggExtractor::seekToTime(int64_t timeUs) {
+    timeUs -= mSeekPreRollUs;
+    if (timeUs < 0) {
+        timeUs = 0;
+    }
+
+    if (mTableOfContents.isEmpty()) {
+        // Perform approximate seeking based on avg. bitrate.
+        uint64_t bps = approxBitrate();
+        if (bps <= 0) {
+            return INVALID_OPERATION;
+        }
+
+        off64_t pos = timeUs * bps / 8000000ll;
+
+        ALOGV("seeking to offset %lld", (long long)pos);
+        return seekToOffset(pos);
+    }
+
+    size_t left = 0;
+    size_t right_plus_one = mTableOfContents.size();
+    while (left < right_plus_one) {
+        size_t center = left + (right_plus_one - left) / 2;
+
+        const TOCEntry &entry = mTableOfContents.itemAt(center);
+
+        if (timeUs < entry.mTimeUs) {
+            right_plus_one = center;
+        } else if (timeUs > entry.mTimeUs) {
+            left = center + 1;
+        } else {
+            left = center;
+            break;
+        }
+    }
+
+    if (left == mTableOfContents.size()) {
+        --left;
+    }
+
+    const TOCEntry &entry = mTableOfContents.itemAt(left);
+
+    ALOGV("seeking to entry %zu / %zu at offset %lld",
+         left, mTableOfContents.size(), (long long)entry.mPageOffset);
+
+    return seekToOffset(entry.mPageOffset);
+}
+
+status_t MyOggExtractor::seekToOffset(off64_t offset) {
+    if (mFirstDataOffset >= 0 && offset < mFirstDataOffset) {
+        // Once we know where the actual audio data starts (past the headers)
+        // don't ever seek to anywhere before that.
+        offset = mFirstDataOffset;
+    }
+
+    off64_t pageOffset;
+    status_t err = findNextPage(offset, &pageOffset);
+
+    if (err != OK) {
+        return err;
+    }
+
+    // We found the page we wanted to seek to, but we'll also need
+    // the page preceding it to determine how many valid samples are on
+    // this page.
+    findPrevGranulePosition(pageOffset, &mPrevGranulePosition);
+
+    mOffset = pageOffset;
+
+    mCurrentPageSize = 0;
+    mFirstPacketInPage = true;
+    mCurrentPageSamples = 0;
+    mCurrentPage.mNumSegments = 0;
+    mCurrentPage.mPrevPacketSize = -1;
+    mNextLaceIndex = 0;
+
+    // XXX what if new page continues packet from last???
+
+    return OK;
+}
+
+ssize_t MyOggExtractor::readPage(off64_t offset, Page *page) {
+    uint8_t header[27];
+    ssize_t n;
+    if ((n = mSource->readAt(offset, header, sizeof(header)))
+            < (ssize_t)sizeof(header)) {
+        ALOGV("failed to read %zu bytes at offset %#016llx, got %zd bytes",
+                sizeof(header), (long long)offset, n);
+
+        if (n < 0) {
+            return n;
+        } else if (n == 0) {
+            return ERROR_END_OF_STREAM;
+        } else {
+            return ERROR_IO;
+        }
+    }
+
+    if (memcmp(header, "OggS", 4)) {
+        return ERROR_MALFORMED;
+    }
+
+    if (header[4] != 0) {
+        // Wrong version.
+
+        return ERROR_UNSUPPORTED;
+    }
+
+    page->mFlags = header[5];
+
+    if (page->mFlags & ~7) {
+        // Only bits 0-2 are defined in version 0.
+        return ERROR_MALFORMED;
+    }
+
+    page->mGranulePosition = U64LE_AT(&header[6]);
+
+#if 0
+    printf("granulePosition = %llu (0x%llx)\n",
+           page->mGranulePosition, page->mGranulePosition);
+#endif
+
+    page->mSerialNo = U32LE_AT(&header[14]);
+    page->mPageNo = U32LE_AT(&header[18]);
+
+    page->mNumSegments = header[26];
+    if (mSource->readAt(
+                offset + sizeof(header), page->mLace, page->mNumSegments)
+            < (ssize_t)page->mNumSegments) {
+        return ERROR_IO;
+    }
+
+    size_t totalSize = 0;;
+    for (size_t i = 0; i < page->mNumSegments; ++i) {
+        totalSize += page->mLace[i];
+    }
+
+#if 0
+    String8 tmp;
+    for (size_t i = 0; i < page->mNumSegments; ++i) {
+        char x[32];
+        sprintf(x, "%s%u", i > 0 ? ", " : "", (unsigned)page->mLace[i]);
+
+        tmp.append(x);
+    }
+
+    ALOGV("%c %s", page->mFlags & 1 ? '+' : ' ', tmp.string());
+#endif
+
+    return sizeof(header) + page->mNumSegments + totalSize;
+}
+
+status_t MyOpusExtractor::readNextPacket(MediaBufferBase **out) {
+    if (mOffset <= mFirstDataOffset && mStartGranulePosition < 0) {
+        // The first sample might not start at time 0; find out where by subtracting
+        // the number of samples on the first page from the granule position
+        // (position of last complete sample) of the first page. This happens
+        // the first time before we attempt to read a packet from the first page.
+        MediaBufferBase *mBuf;
+        uint32_t numSamples = 0;
+        uint64_t curGranulePosition = 0;
+        while (true) {
+            status_t err = _readNextPacket(&mBuf, /* calcVorbisTimestamp = */false);
+            if (err != OK && err != ERROR_END_OF_STREAM) {
+                return err;
+            }
+            // First two pages are header pages.
+            if (err == ERROR_END_OF_STREAM || mCurrentPage.mPageNo > 2) {
+                if (mBuf != NULL) {
+                    mBuf->release();
+                    mBuf = NULL;
+                }
+                break;
+            }
+            curGranulePosition = mCurrentPage.mGranulePosition;
+            numSamples += getNumSamplesInPacket(mBuf);
+            mBuf->release();
+            mBuf = NULL;
+        }
+
+        if (curGranulePosition > numSamples) {
+            mStartGranulePosition = curGranulePosition - numSamples;
+        } else {
+            mStartGranulePosition = 0;
+        }
+        seekToOffset(0);
+    }
+
+    status_t err = _readNextPacket(out, /* calcVorbisTimestamp = */false);
+    if (err != OK) {
+        return err;
+    }
+
+    int32_t currentPageSamples;
+    // Calculate timestamps by accumulating durations starting from the first sample of a page;
+    // We assume that we only seek to page boundaries.
+    if ((*out)->meta_data().findInt32(kKeyValidSamples, &currentPageSamples)) {
+        // first packet in page
+        if (mOffset == mFirstDataOffset) {
+            currentPageSamples -= mStartGranulePosition;
+            (*out)->meta_data().setInt32(kKeyValidSamples, currentPageSamples);
+        }
+        mCurGranulePosition = mCurrentPage.mGranulePosition - currentPageSamples;
+    }
+
+    int64_t timeUs = getTimeUsOfGranule(mCurGranulePosition);
+    (*out)->meta_data().setInt64(kKeyTime, timeUs);
+
+    uint32_t frames = getNumSamplesInPacket(*out);
+    mCurGranulePosition += frames;
+    return OK;
+}
+
+uint32_t MyOpusExtractor::getNumSamplesInPacket(MediaBufferBase *buffer) const {
+    if (buffer == NULL || buffer->range_length() < 1) {
+        return 0;
+    }
+
+    uint8_t *data = (uint8_t *)buffer->data() + buffer->range_offset();
+    uint8_t toc = data[0];
+    uint8_t config = (toc >> 3) & 0x1f;
+    uint32_t frameSizesUs[] = {
+        10000, 20000, 40000, 60000, // 0...3
+        10000, 20000, 40000, 60000, // 4...7
+        10000, 20000, 40000, 60000, // 8...11
+        10000, 20000,               // 12...13
+        10000, 20000,               // 14...15
+        2500, 5000, 10000, 20000,   // 16...19
+        2500, 5000, 10000, 20000,   // 20...23
+        2500, 5000, 10000, 20000,   // 24...27
+        2500, 5000, 10000, 20000    // 28...31
+    };
+    uint32_t frameSizeUs = frameSizesUs[config];
+
+    uint32_t numFrames;
+    uint8_t c = toc & 3;
+    switch (c) {
+    case 0:
+        numFrames = 1;
+        break;
+    case 1:
+    case 2:
+        numFrames = 2;
+        break;
+    case 3:
+        if (buffer->range_length() < 3) {
+            numFrames = 0;
+        } else {
+            numFrames = data[2] & 0x3f;
+        }
+        break;
+    default:
+        TRESPASS();
+    }
+
+    uint32_t numSamples = frameSizeUs * numFrames * kOpusSampleRate / 1000000;
+    return numSamples;
+}
+
+status_t MyOggExtractor::_readNextPacket(MediaBufferBase **out, bool calcVorbisTimestamp) {
+    *out = NULL;
+
+    MediaBufferBase *buffer = NULL;
+    int64_t timeUs = -1;
+
+    for (;;) {
+        size_t i;
+        size_t packetSize = 0;
+        bool gotFullPacket = false;
+        for (i = mNextLaceIndex; i < mCurrentPage.mNumSegments; ++i) {
+            uint8_t lace = mCurrentPage.mLace[i];
+
+            packetSize += lace;
+
+            if (lace < 255) {
+                gotFullPacket = true;
+                ++i;
+                break;
+            }
+        }
+
+        if (mNextLaceIndex < mCurrentPage.mNumSegments) {
+            off64_t dataOffset = mOffset + 27 + mCurrentPage.mNumSegments;
+            for (size_t j = 0; j < mNextLaceIndex; ++j) {
+                dataOffset += mCurrentPage.mLace[j];
+            }
+
+            size_t fullSize = packetSize;
+            if (buffer != NULL) {
+                fullSize += buffer->range_length();
+            }
+            if (fullSize > 16 * 1024 * 1024) { // arbitrary limit of 16 MB packet size
+                if (buffer != NULL) {
+                    buffer->release();
+                }
+                ALOGE("b/36592202");
+                return ERROR_MALFORMED;
+            }
+            MediaBufferBase *tmp = MediaBufferBase::Create(fullSize);
+            if (tmp == NULL) {
+                if (buffer != NULL) {
+                    buffer->release();
+                }
+                ALOGE("b/36592202");
+                return ERROR_MALFORMED;
+            }
+            if (buffer != NULL) {
+                memcpy(tmp->data(), buffer->data(), buffer->range_length());
+                tmp->set_range(0, buffer->range_length());
+                buffer->release();
+            } else {
+                tmp->set_range(0, 0);
+            }
+            buffer = tmp;
+
+            ssize_t n = mSource->readAt(
+                    dataOffset,
+                    (uint8_t *)buffer->data() + buffer->range_length(),
+                    packetSize);
+
+            if (n < (ssize_t)packetSize) {
+                buffer->release();
+                ALOGV("failed to read %zu bytes at %#016llx, got %zd bytes",
+                        packetSize, (long long)dataOffset, n);
+                return ERROR_IO;
+            }
+
+            buffer->set_range(0, fullSize);
+
+            mNextLaceIndex = i;
+
+            if (gotFullPacket) {
+                // We've just read the entire packet.
+
+                if (mFirstPacketInPage) {
+                    buffer->meta_data().setInt32(
+                            kKeyValidSamples, mCurrentPageSamples);
+                    mFirstPacketInPage = false;
+                }
+
+                if (calcVorbisTimestamp) {
+                    int32_t curBlockSize = getPacketBlockSize(buffer);
+                    if (mCurrentPage.mPrevPacketSize < 0) {
+                        mCurrentPage.mPrevPacketSize = curBlockSize;
+                        mCurrentPage.mPrevPacketPos =
+                                mCurrentPage.mGranulePosition - mCurrentPageSamples;
+                        timeUs = mCurrentPage.mPrevPacketPos * 1000000ll / mVi.rate;
+                    } else {
+                        // The effective block size is the average of the two overlapped blocks
+                        int32_t actualBlockSize =
+                                (curBlockSize + mCurrentPage.mPrevPacketSize) / 2;
+                        timeUs = mCurrentPage.mPrevPacketPos * 1000000ll / mVi.rate;
+                        // The actual size output by the decoder will be half the effective
+                        // size, due to the overlap
+                        mCurrentPage.mPrevPacketPos += actualBlockSize / 2;
+                        mCurrentPage.mPrevPacketSize = curBlockSize;
+                    }
+                    buffer->meta_data().setInt64(kKeyTime, timeUs);
+                }
+                *out = buffer;
+
+                return OK;
+            }
+
+            // fall through, the buffer now contains the start of the packet.
+        }
+
+        CHECK_EQ(mNextLaceIndex, mCurrentPage.mNumSegments);
+
+        mOffset += mCurrentPageSize;
+        ssize_t n = readPage(mOffset, &mCurrentPage);
+
+        if (n <= 0) {
+            if (buffer) {
+                buffer->release();
+                buffer = NULL;
+            }
+
+            ALOGV("readPage returned %zd", n);
+
+            return n < 0 ? n : (status_t)ERROR_END_OF_STREAM;
+        }
+
+        // Prevent a harmless unsigned integer overflow by clamping to 0
+        if (mCurrentPage.mGranulePosition >= mPrevGranulePosition) {
+            mCurrentPageSamples =
+                    mCurrentPage.mGranulePosition - mPrevGranulePosition;
+        } else {
+            mCurrentPageSamples = 0;
+        }
+        mFirstPacketInPage = true;
+
+        mPrevGranulePosition = mCurrentPage.mGranulePosition;
+
+        mCurrentPageSize = n;
+        mNextLaceIndex = 0;
+
+        if (buffer != NULL) {
+            if ((mCurrentPage.mFlags & 1) == 0) {
+                // This page does not continue the packet, i.e. the packet
+                // is already complete.
+
+                if (timeUs >= 0) {
+                    buffer->meta_data().setInt64(kKeyTime, timeUs);
+                }
+
+                buffer->meta_data().setInt32(
+                        kKeyValidSamples, mCurrentPageSamples);
+                mFirstPacketInPage = false;
+
+                *out = buffer;
+
+                return OK;
+            }
+        }
+    }
+}
+
+status_t MyOggExtractor::init() {
+    mMeta.setCString(kKeyMIMEType, mMimeType);
+
+    status_t err;
+    MediaBufferBase *packet;
+    for (size_t i = 0; i < mNumHeaders; ++i) {
+        // ignore timestamp for configuration packets
+        if ((err = _readNextPacket(&packet, /* calcVorbisTimestamp = */ false)) != OK) {
+            return err;
+        }
+        ALOGV("read packet of size %zu\n", packet->range_length());
+        err = verifyHeader(packet, /* type = */ i * 2 + 1);
+        packet->release();
+        packet = NULL;
+        if (err != OK) {
+            return err;
+        }
+    }
+
+    mFirstDataOffset = mOffset + mCurrentPageSize;
+
+    off64_t size;
+    uint64_t lastGranulePosition;
+    if (!(mSource->flags() & DataSourceBase::kIsCachingDataSource)
+            && mSource->getSize(&size) == OK
+            && findPrevGranulePosition(size, &lastGranulePosition) == OK) {
+        // Let's assume it's cheap to seek to the end.
+        // The granule position of the final page in the stream will
+        // give us the exact duration of the content, something that
+        // we can only approximate using avg. bitrate if seeking to
+        // the end is too expensive or impossible (live streaming).
+
+        int64_t durationUs = getTimeUsOfGranule(lastGranulePosition);
+
+        mMeta.setInt64(kKeyDuration, durationUs);
+
+        buildTableOfContents();
+    }
+
+    return OK;
+}
+
+void MyOggExtractor::buildTableOfContents() {
+    off64_t offset = mFirstDataOffset;
+    Page page;
+    ssize_t pageSize;
+    while ((pageSize = readPage(offset, &page)) > 0) {
+        mTableOfContents.push();
+
+        TOCEntry &entry =
+            mTableOfContents.editItemAt(mTableOfContents.size() - 1);
+
+        entry.mPageOffset = offset;
+        entry.mTimeUs = getTimeUsOfGranule(page.mGranulePosition);
+
+        offset += (size_t)pageSize;
+    }
+
+    // Limit the maximum amount of RAM we spend on the table of contents,
+    // if necessary thin out the table evenly to trim it down to maximum
+    // size.
+
+    static const size_t kMaxTOCSize = 8192;
+    static const size_t kMaxNumTOCEntries = kMaxTOCSize / sizeof(TOCEntry);
+
+    size_t numerator = mTableOfContents.size();
+
+    if (numerator > kMaxNumTOCEntries) {
+        size_t denom = numerator - kMaxNumTOCEntries;
+
+        size_t accum = 0;
+        for (ssize_t i = mTableOfContents.size() - 1; i >= 0; --i) {
+            accum += denom;
+            if (accum >= numerator) {
+                mTableOfContents.removeAt(i);
+                accum -= numerator;
+            }
+        }
+    }
+}
+
+int32_t MyOggExtractor::getPacketBlockSize(MediaBufferBase *buffer) {
+    const uint8_t *data =
+        (const uint8_t *)buffer->data() + buffer->range_offset();
+
+    size_t size = buffer->range_length();
+
+    ogg_buffer buf;
+    buf.data = (uint8_t *)data;
+    buf.size = size;
+    buf.refcount = 1;
+    buf.ptr.owner = NULL;
+
+    ogg_reference ref;
+    ref.buffer = &buf;
+    ref.begin = 0;
+    ref.length = size;
+    ref.next = NULL;
+
+    ogg_packet pack;
+    pack.packet = &ref;
+    pack.bytes = ref.length;
+    pack.b_o_s = 0;
+    pack.e_o_s = 0;
+    pack.granulepos = 0;
+    pack.packetno = 0;
+
+    return vorbis_packet_blocksize(&mVi, &pack);
+}
+
+int64_t MyOpusExtractor::getTimeUsOfGranule(uint64_t granulePos) const {
+    uint64_t pcmSamplePosition = 0;
+    if (granulePos > mCodecDelay) {
+        pcmSamplePosition = granulePos - mCodecDelay;
+    }
+    if (pcmSamplePosition > INT64_MAX / 1000000ll) {
+        return INT64_MAX;
+    }
+    return pcmSamplePosition * 1000000ll / kOpusSampleRate;
+}
+
+status_t MyOpusExtractor::verifyHeader(MediaBufferBase *buffer, uint8_t type) {
+    switch (type) {
+        // there are actually no header types defined in the Opus spec; we choose 1 and 3 to mean
+        // header and comments such that we can share code with MyVorbisExtractor.
+        case 1:
+            return verifyOpusHeader(buffer);
+        case 3:
+            return verifyOpusComments(buffer);
+        default:
+            return INVALID_OPERATION;
+    }
+}
+
+status_t MyOpusExtractor::verifyOpusHeader(MediaBufferBase *buffer) {
+    const size_t kOpusHeaderSize = 19;
+    const uint8_t *data =
+        (const uint8_t *)buffer->data() + buffer->range_offset();
+
+    size_t size = buffer->range_length();
+
+    if (size < kOpusHeaderSize
+            || memcmp(data, "OpusHead", 8)
+            || /* version = */ data[8] != 1) {
+        return ERROR_MALFORMED;
+    }
+
+    mChannelCount = data[9];
+    mCodecDelay = U16LE_AT(&data[10]);
+
+    mMeta.setData(kKeyOpusHeader, 0, data, size);
+    mMeta.setInt32(kKeySampleRate, kOpusSampleRate);
+    mMeta.setInt32(kKeyChannelCount, mChannelCount);
+    mMeta.setInt64(kKeyOpusSeekPreRoll /* ns */, kOpusSeekPreRollUs * 1000 /* = 80 ms*/);
+    mMeta.setInt64(kKeyOpusCodecDelay /* ns */,
+            mCodecDelay /* sample/s */ * 1000000000ll / kOpusSampleRate);
+
+    return OK;
+}
+
+status_t MyOpusExtractor::verifyOpusComments(MediaBufferBase *buffer) {
+    // add artificial framing bit so we can reuse _vorbis_unpack_comment
+    int32_t commentSize = buffer->range_length() + 1;
+    auto tmp = heapbuffer<uint8_t>(commentSize);
+    uint8_t *commentData = tmp.get();
+    if (commentData == nullptr) {
+        return ERROR_MALFORMED;
+    }
+
+    memcpy(commentData,
+            (uint8_t *)buffer->data() + buffer->range_offset(),
+            buffer->range_length());
+
+    ogg_buffer buf;
+    buf.data = commentData;
+    buf.size = commentSize;
+    buf.refcount = 1;
+    buf.ptr.owner = NULL;
+
+    ogg_reference ref;
+    ref.buffer = &buf;
+    ref.begin = 0;
+    ref.length = commentSize;
+    ref.next = NULL;
+
+    oggpack_buffer bits;
+    oggpack_readinit(&bits, &ref);
+
+    // skip 'OpusTags'
+    const char *OpusTags = "OpusTags";
+    const int32_t headerLen = strlen(OpusTags);
+    int32_t framingBitOffset = headerLen;
+    for (int i = 0; i < headerLen; ++i) {
+        char chr = oggpack_read(&bits, 8);
+        if (chr != OpusTags[i]) {
+            return ERROR_MALFORMED;
+        }
+    }
+
+    int32_t vendorLen = oggpack_read(&bits, 32);
+    framingBitOffset += 4;
+    if (vendorLen < 0 || vendorLen > commentSize - 8) {
+        return ERROR_MALFORMED;
+    }
+    // skip vendor string
+    framingBitOffset += vendorLen;
+    for (int i = 0; i < vendorLen; ++i) {
+        oggpack_read(&bits, 8);
+    }
+
+    int32_t n = oggpack_read(&bits, 32);
+    framingBitOffset += 4;
+    if (n < 0 || n > ((commentSize - oggpack_bytes(&bits)) >> 2)) {
+        return ERROR_MALFORMED;
+    }
+    for (int i = 0; i < n; ++i) {
+        int32_t len = oggpack_read(&bits, 32);
+        framingBitOffset += 4;
+        if (len  < 0 || len  > (commentSize - oggpack_bytes(&bits))) {
+            return ERROR_MALFORMED;
+        }
+        framingBitOffset += len;
+        for (int j = 0; j < len; ++j) {
+            oggpack_read(&bits, 8);
+        }
+    }
+    if (framingBitOffset < 0 || framingBitOffset >= commentSize) {
+        return ERROR_MALFORMED;
+    }
+    commentData[framingBitOffset] = 1;
+
+    buf.data = commentData + headerLen;
+    buf.size = commentSize - headerLen;
+    buf.refcount = 1;
+    buf.ptr.owner = NULL;
+
+    ref.buffer = &buf;
+    ref.begin = 0;
+    ref.length = commentSize - headerLen;
+    ref.next = NULL;
+
+    oggpack_readinit(&bits, &ref);
+    int err = _vorbis_unpack_comment(&mVc, &bits);
+    if (0 != err) {
+        return ERROR_MALFORMED;
+    }
+
+    parseFileMetaData();
+    return OK;
+}
+
+status_t MyVorbisExtractor::verifyHeader(
+        MediaBufferBase *buffer, uint8_t type) {
+    const uint8_t *data =
+        (const uint8_t *)buffer->data() + buffer->range_offset();
+
+    size_t size = buffer->range_length();
+
+    if (size < 7 || data[0] != type || memcmp(&data[1], "vorbis", 6)) {
+        return ERROR_MALFORMED;
+    }
+
+    ogg_buffer buf;
+    buf.data = (uint8_t *)data;
+    buf.size = size;
+    buf.refcount = 1;
+    buf.ptr.owner = NULL;
+
+    ogg_reference ref;
+    ref.buffer = &buf;
+    ref.begin = 0;
+    ref.length = size;
+    ref.next = NULL;
+
+    oggpack_buffer bits;
+    oggpack_readinit(&bits, &ref);
+
+    if (oggpack_read(&bits, 8) != type) {
+        return ERROR_MALFORMED;
+    }
+    for (size_t i = 0; i < 6; ++i) {
+        oggpack_read(&bits, 8);  // skip 'vorbis'
+    }
+
+    switch (type) {
+        case 1:
+        {
+            if (0 != _vorbis_unpack_info(&mVi, &bits)) {
+                return ERROR_MALFORMED;
+            }
+
+            mMeta.setData(kKeyVorbisInfo, 0, data, size);
+            mMeta.setInt32(kKeySampleRate, mVi.rate);
+            mMeta.setInt32(kKeyChannelCount, mVi.channels);
+            mMeta.setInt32(kKeyBitRate, mVi.bitrate_nominal);
+
+            ALOGV("lower-bitrate = %ld", mVi.bitrate_lower);
+            ALOGV("upper-bitrate = %ld", mVi.bitrate_upper);
+            ALOGV("nominal-bitrate = %ld", mVi.bitrate_nominal);
+            ALOGV("window-bitrate = %ld", mVi.bitrate_window);
+            ALOGV("blocksizes: %d/%d",
+                    vorbis_info_blocksize(&mVi, 0),
+                    vorbis_info_blocksize(&mVi, 1)
+                    );
+
+            off64_t size;
+            if (mSource->getSize(&size) == OK) {
+                uint64_t bps = approxBitrate();
+                if (bps != 0) {
+                    mMeta.setInt64(kKeyDuration, size * 8000000ll / bps);
+                }
+            }
+            break;
+        }
+
+        case 3:
+        {
+            if (0 != _vorbis_unpack_comment(&mVc, &bits)) {
+                return ERROR_MALFORMED;
+            }
+
+            parseFileMetaData();
+            break;
+        }
+
+        case 5:
+        {
+            if (0 != _vorbis_unpack_books(&mVi, &bits)) {
+                return ERROR_MALFORMED;
+            }
+
+            mMeta.setData(kKeyVorbisBooks, 0, data, size);
+            break;
+        }
+    }
+
+    return OK;
+}
+
+uint64_t MyVorbisExtractor::approxBitrate() const {
+    if (mVi.bitrate_nominal != 0) {
+        return mVi.bitrate_nominal;
+    }
+
+    return (mVi.bitrate_lower + mVi.bitrate_upper) / 2;
+}
+
+
+void MyOggExtractor::parseFileMetaData() {
+    mFileMeta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_CONTAINER_OGG);
+
+    for (int i = 0; i < mVc.comments; ++i) {
+        const char *comment = mVc.user_comments[i];
+        size_t commentLength = mVc.comment_lengths[i];
+        parseVorbisComment(&mFileMeta, comment, commentLength);
+        //ALOGI("comment #%d: '%s'", i + 1, mVc.user_comments[i]);
+    }
+}
+
+
+////////////////////////////////////////////////////////////////////////////////
+
+OggExtractor::OggExtractor(DataSourceBase *source)
+    : mDataSource(source),
+      mInitCheck(NO_INIT),
+      mImpl(NULL) {
+    for (int i = 0; i < 2; ++i) {
+        if (mImpl != NULL) {
+            delete mImpl;
+        }
+        if (i == 0) {
+            mImpl = new MyVorbisExtractor(mDataSource);
+        } else {
+            mImpl = new MyOpusExtractor(mDataSource);
+        }
+        mInitCheck = mImpl->seekToOffset(0);
+
+        if (mInitCheck == OK) {
+            mInitCheck = mImpl->init();
+            if (mInitCheck == OK) {
+                break;
+            }
+        }
+    }
+}
+
+OggExtractor::~OggExtractor() {
+    delete mImpl;
+    mImpl = NULL;
+}
+
+size_t OggExtractor::countTracks() {
+    return mInitCheck != OK ? 0 : 1;
+}
+
+MediaTrack *OggExtractor::getTrack(size_t index) {
+    if (index >= 1) {
+        return NULL;
+    }
+
+    return new OggSource(this);
+}
+
+status_t OggExtractor::getTrackMetaData(
+        MetaDataBase &meta,
+        size_t index, uint32_t /* flags */) {
+    if (index >= 1) {
+        return UNKNOWN_ERROR;
+    }
+
+    return mImpl->getFormat(meta);
+}
+
+status_t OggExtractor::getMetaData(MetaDataBase &meta) {
+    return mImpl->getFileMetaData(meta);
+}
+
+static MediaExtractor* CreateExtractor(
+        DataSourceBase *source,
+        void *) {
+    return new OggExtractor(source);
+}
+
+static MediaExtractor::CreatorFunc Sniff(
+        DataSourceBase *source,
+        float *confidence,
+        void **,
+        MediaExtractor::FreeMetaFunc *) {
+    char tmp[4];
+    if (source->readAt(0, tmp, 4) < 4 || memcmp(tmp, "OggS", 4)) {
+        return NULL;
+    }
+
+    *confidence = 0.2f;
+
+    return CreateExtractor;
+}
+
+extern "C" {
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
+MediaExtractor::ExtractorDef GETEXTRACTORDEF() {
+    return {
+        MediaExtractor::EXTRACTORDEF_VERSION,
+        UUID("8cc5cd06-f772-495e-8a62-cba9649374e9"),
+        1, // version
+        "Ogg Extractor",
+        Sniff
+    };
+}
+
+} // extern "C"
+
+}  // namespace android

diff --git a/media/extractors/ogg/OggExtractor.h b/media/extractors/ogg/OggExtractor.h
new file mode 100644
index 0000000..9fe2944
--- /dev/null
+++ b/media/extractors/ogg/OggExtractor.h

@@ -0,0 +1,60 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef OGG_EXTRACTOR_H_
+
+#define OGG_EXTRACTOR_H_
+
+#include <utils/Errors.h>
+#include <media/MediaExtractor.h>
+
+namespace android {
+
+struct AMessage;
+class DataSourceBase;
+class String8;
+
+struct MyOggExtractor;
+struct OggSource;
+
+struct OggExtractor : public MediaExtractor {
+    explicit OggExtractor(DataSourceBase *source);
+
+    virtual size_t countTracks();
+    virtual MediaTrack *getTrack(size_t index);
+    virtual status_t getTrackMetaData(MetaDataBase& meta, size_t index, uint32_t flags);
+
+    virtual status_t getMetaData(MetaDataBase& meta);
+    virtual const char * name() { return "OggExtractor"; }
+
+protected:
+    virtual ~OggExtractor();
+
+private:
+    friend struct OggSource;
+
+    DataSourceBase *mDataSource;
+    status_t mInitCheck;
+
+    MyOggExtractor *mImpl;
+
+    OggExtractor(const OggExtractor &);
+    OggExtractor &operator=(const OggExtractor &);
+};
+
+}  // namespace android
+
+#endif  // OGG_EXTRACTOR_H_

diff --git a/media/extractors/ogg/exports.lds b/media/extractors/ogg/exports.lds
new file mode 100644
index 0000000..b1309ee
--- /dev/null
+++ b/media/extractors/ogg/exports.lds

@@ -0,0 +1 @@
+{ global: GETEXTRACTORDEF; local: *; };

diff --git a/media/extractors/wav/Android.bp b/media/extractors/wav/Android.bp
new file mode 100644
index 0000000..067933e
--- /dev/null
+++ b/media/extractors/wav/Android.bp

@@ -0,0 +1,42 @@
+cc_library_shared {
+
+    srcs: ["WAVExtractor.cpp"],
+
+    include_dirs: [
+        "frameworks/av/media/libstagefright/include",
+    ],
+
+    shared_libs: [
+        "liblog",
+        "libmediaextractor",
+    ],
+
+    static_libs: [
+        "libfifo",
+        "libstagefright_foundation",
+    ],
+
+    name: "libwavextractor",
+    relative_install_path: "extractors",
+
+    compile_multilib: "first",
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+        "-fvisibility=hidden",
+    ],
+    version_script: "exports.lds",
+
+    sanitize: {
+        cfi: true,
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+        diag: {
+            cfi: true,
+        },
+    },
+
+}

diff --git a/media/libstagefright/matroska/MODULE_LICENSE_APACHE2 b/media/extractors/wav/MODULE_LICENSE_APACHE2
similarity index 100%
copy from media/libstagefright/matroska/MODULE_LICENSE_APACHE2
copy to media/extractors/wav/MODULE_LICENSE_APACHE2


diff --git a/media/libstagefright/matroska/NOTICE b/media/extractors/wav/NOTICE
similarity index 100%
copy from media/libstagefright/matroska/NOTICE
copy to media/extractors/wav/NOTICE


diff --git a/media/extractors/wav/WAVExtractor.cpp b/media/extractors/wav/WAVExtractor.cpp
new file mode 100644
index 0000000..f5a1b01
--- /dev/null
+++ b/media/extractors/wav/WAVExtractor.cpp

@@ -0,0 +1,594 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "WAVExtractor"
+#include <utils/Log.h>
+
+#include "WAVExtractor.h"
+
+#include <audio_utils/primitives.h>
+#include <media/DataSourceBase.h>
+#include <media/MediaTrack.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <utils/String8.h>
+#include <cutils/bitops.h>
+
+#define CHANNEL_MASK_USE_CHANNEL_ORDER 0
+
+namespace android {
+
+enum {
+    WAVE_FORMAT_PCM        = 0x0001,
+    WAVE_FORMAT_IEEE_FLOAT = 0x0003,
+    WAVE_FORMAT_ALAW       = 0x0006,
+    WAVE_FORMAT_MULAW      = 0x0007,
+    WAVE_FORMAT_MSGSM      = 0x0031,
+    WAVE_FORMAT_EXTENSIBLE = 0xFFFE
+};
+
+static const char* WAVEEXT_SUBFORMAT = "\x00\x00\x00\x00\x10\x00\x80\x00\x00\xAA\x00\x38\x9B\x71";
+static const char* AMBISONIC_SUBFORMAT = "\x00\x00\x21\x07\xD3\x11\x86\x44\xC8\xC1\xCA\x00\x00\x00";
+
+static uint32_t U32_LE_AT(const uint8_t *ptr) {
+    return ptr[3] << 24 | ptr[2] << 16 | ptr[1] << 8 | ptr[0];
+}
+
+static uint16_t U16_LE_AT(const uint8_t *ptr) {
+    return ptr[1] << 8 | ptr[0];
+}
+
+struct WAVSource : public MediaTrack {
+    WAVSource(
+            DataSourceBase *dataSource,
+            MetaDataBase &meta,
+            uint16_t waveFormat,
+            int32_t bitsPerSample,
+            off64_t offset, size_t size);
+
+    virtual status_t start(MetaDataBase *params = NULL);
+    virtual status_t stop();
+    virtual status_t getFormat(MetaDataBase &meta);
+
+    virtual status_t read(
+            MediaBufferBase **buffer, const ReadOptions *options = NULL);
+
+    virtual bool supportNonblockingRead() { return true; }
+
+protected:
+    virtual ~WAVSource();
+
+private:
+    static const size_t kMaxFrameSize;
+
+    DataSourceBase *mDataSource;
+    MetaDataBase &mMeta;
+    uint16_t mWaveFormat;
+    int32_t mSampleRate;
+    int32_t mNumChannels;
+    int32_t mBitsPerSample;
+    off64_t mOffset;
+    size_t mSize;
+    bool mStarted;
+    MediaBufferGroup *mGroup;
+    off64_t mCurrentPos;
+
+    WAVSource(const WAVSource &);
+    WAVSource &operator=(const WAVSource &);
+};
+
+WAVExtractor::WAVExtractor(DataSourceBase *source)
+    : mDataSource(source),
+      mValidFormat(false),
+      mChannelMask(CHANNEL_MASK_USE_CHANNEL_ORDER) {
+    mInitCheck = init();
+}
+
+WAVExtractor::~WAVExtractor() {
+}
+
+status_t WAVExtractor::getMetaData(MetaDataBase &meta) {
+    meta.clear();
+    if (mInitCheck == OK) {
+        meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_CONTAINER_WAV);
+    }
+
+    return OK;
+}
+
+size_t WAVExtractor::countTracks() {
+    return mInitCheck == OK ? 1 : 0;
+}
+
+MediaTrack *WAVExtractor::getTrack(size_t index) {
+    if (mInitCheck != OK || index > 0) {
+        return NULL;
+    }
+
+    return new WAVSource(
+            mDataSource, mTrackMeta,
+            mWaveFormat, mBitsPerSample, mDataOffset, mDataSize);
+}
+
+status_t WAVExtractor::getTrackMetaData(
+        MetaDataBase &meta,
+        size_t index, uint32_t /* flags */) {
+    if (mInitCheck != OK || index > 0) {
+        return UNKNOWN_ERROR;
+    }
+
+    meta = mTrackMeta;
+    return OK;
+}
+
+status_t WAVExtractor::init() {
+    uint8_t header[12];
+    if (mDataSource->readAt(
+                0, header, sizeof(header)) < (ssize_t)sizeof(header)) {
+        return NO_INIT;
+    }
+
+    if (memcmp(header, "RIFF", 4) || memcmp(&header[8], "WAVE", 4)) {
+        return NO_INIT;
+    }
+
+    size_t totalSize = U32_LE_AT(&header[4]);
+
+    off64_t offset = 12;
+    size_t remainingSize = totalSize;
+    while (remainingSize >= 8) {
+        uint8_t chunkHeader[8];
+        if (mDataSource->readAt(offset, chunkHeader, 8) < 8) {
+            return NO_INIT;
+        }
+
+        remainingSize -= 8;
+        offset += 8;
+
+        uint32_t chunkSize = U32_LE_AT(&chunkHeader[4]);
+
+        if (chunkSize > remainingSize) {
+            return NO_INIT;
+        }
+
+        if (!memcmp(chunkHeader, "fmt ", 4)) {
+            if (chunkSize < 16) {
+                return NO_INIT;
+            }
+
+            uint8_t formatSpec[40];
+            if (mDataSource->readAt(offset, formatSpec, 2) < 2) {
+                return NO_INIT;
+            }
+
+            mWaveFormat = U16_LE_AT(formatSpec);
+            if (mWaveFormat != WAVE_FORMAT_PCM
+                    && mWaveFormat != WAVE_FORMAT_IEEE_FLOAT
+                    && mWaveFormat != WAVE_FORMAT_ALAW
+                    && mWaveFormat != WAVE_FORMAT_MULAW
+                    && mWaveFormat != WAVE_FORMAT_MSGSM
+                    && mWaveFormat != WAVE_FORMAT_EXTENSIBLE) {
+                return ERROR_UNSUPPORTED;
+            }
+
+            uint8_t fmtSize = 16;
+            if (mWaveFormat == WAVE_FORMAT_EXTENSIBLE) {
+                fmtSize = 40;
+            }
+            if (mDataSource->readAt(offset, formatSpec, fmtSize) < fmtSize) {
+                return NO_INIT;
+            }
+
+            mNumChannels = U16_LE_AT(&formatSpec[2]);
+
+            if (mNumChannels < 1 || mNumChannels > 8) {
+                ALOGE("Unsupported number of channels (%d)", mNumChannels);
+                return ERROR_UNSUPPORTED;
+            }
+
+            if (mWaveFormat != WAVE_FORMAT_EXTENSIBLE) {
+                if (mNumChannels != 1 && mNumChannels != 2) {
+                    ALOGW("More than 2 channels (%d) in non-WAVE_EXT, unknown channel mask",
+                            mNumChannels);
+                }
+            }
+
+            mSampleRate = U32_LE_AT(&formatSpec[4]);
+
+            if (mSampleRate == 0) {
+                return ERROR_MALFORMED;
+            }
+
+            mBitsPerSample = U16_LE_AT(&formatSpec[14]);
+
+            if (mWaveFormat == WAVE_FORMAT_EXTENSIBLE) {
+                uint16_t validBitsPerSample = U16_LE_AT(&formatSpec[18]);
+                if (validBitsPerSample != mBitsPerSample) {
+                    if (validBitsPerSample != 0) {
+                        ALOGE("validBits(%d) != bitsPerSample(%d) are not supported",
+                                validBitsPerSample, mBitsPerSample);
+                        return ERROR_UNSUPPORTED;
+                    } else {
+                        // we only support valitBitsPerSample == bitsPerSample but some WAV_EXT
+                        // writers don't correctly set the valid bits value, and leave it at 0.
+                        ALOGW("WAVE_EXT has 0 valid bits per sample, ignoring");
+                    }
+                }
+
+                mChannelMask = U32_LE_AT(&formatSpec[20]);
+                ALOGV("numChannels=%d channelMask=0x%x", mNumChannels, mChannelMask);
+                if ((mChannelMask >> 18) != 0) {
+                    ALOGE("invalid channel mask 0x%x", mChannelMask);
+                    return ERROR_MALFORMED;
+                }
+
+                if ((mChannelMask != CHANNEL_MASK_USE_CHANNEL_ORDER)
+                        && (popcount(mChannelMask) != mNumChannels)) {
+                    ALOGE("invalid number of channels (%d) in channel mask (0x%x)",
+                            popcount(mChannelMask), mChannelMask);
+                    return ERROR_MALFORMED;
+                }
+
+                // In a WAVE_EXT header, the first two bytes of the GUID stored at byte 24 contain
+                // the sample format, using the same definitions as a regular WAV header
+                mWaveFormat = U16_LE_AT(&formatSpec[24]);
+                if (memcmp(&formatSpec[26], WAVEEXT_SUBFORMAT, 14) &&
+                    memcmp(&formatSpec[26], AMBISONIC_SUBFORMAT, 14)) {
+                    ALOGE("unsupported GUID");
+                    return ERROR_UNSUPPORTED;
+                }
+            }
+
+            if (mWaveFormat == WAVE_FORMAT_PCM) {
+                if (mBitsPerSample != 8 && mBitsPerSample != 16
+                    && mBitsPerSample != 24 && mBitsPerSample != 32) {
+                    return ERROR_UNSUPPORTED;
+                }
+            } else if (mWaveFormat == WAVE_FORMAT_IEEE_FLOAT) {
+                if (mBitsPerSample != 32) {  // TODO we don't support double
+                    return ERROR_UNSUPPORTED;
+                }
+            }
+            else if (mWaveFormat == WAVE_FORMAT_MSGSM) {
+                if (mBitsPerSample != 0) {
+                    return ERROR_UNSUPPORTED;
+                }
+            } else if (mWaveFormat == WAVE_FORMAT_MULAW || mWaveFormat == WAVE_FORMAT_ALAW) {
+                if (mBitsPerSample != 8) {
+                    return ERROR_UNSUPPORTED;
+                }
+            } else {
+                return ERROR_UNSUPPORTED;
+            }
+
+            mValidFormat = true;
+        } else if (!memcmp(chunkHeader, "data", 4)) {
+            if (mValidFormat) {
+                mDataOffset = offset;
+                mDataSize = chunkSize;
+
+                mTrackMeta.clear();
+
+                switch (mWaveFormat) {
+                    case WAVE_FORMAT_PCM:
+                    case WAVE_FORMAT_IEEE_FLOAT:
+                        mTrackMeta.setCString(
+                                kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);
+                        break;
+                    case WAVE_FORMAT_ALAW:
+                        mTrackMeta.setCString(
+                                kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_G711_ALAW);
+                        break;
+                    case WAVE_FORMAT_MSGSM:
+                        mTrackMeta.setCString(
+                                kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MSGSM);
+                        break;
+                    default:
+                        CHECK_EQ(mWaveFormat, (uint16_t)WAVE_FORMAT_MULAW);
+                        mTrackMeta.setCString(
+                                kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_G711_MLAW);
+                        break;
+                }
+
+                mTrackMeta.setInt32(kKeyChannelCount, mNumChannels);
+                mTrackMeta.setInt32(kKeyChannelMask, mChannelMask);
+                mTrackMeta.setInt32(kKeySampleRate, mSampleRate);
+                mTrackMeta.setInt32(kKeyPcmEncoding, kAudioEncodingPcm16bit);
+
+                int64_t durationUs = 0;
+                if (mWaveFormat == WAVE_FORMAT_MSGSM) {
+                    // 65 bytes decode to 320 8kHz samples
+                    durationUs =
+                        1000000LL * (mDataSize / 65 * 320) / 8000;
+                } else {
+                    size_t bytesPerSample = mBitsPerSample >> 3;
+
+                    if (!bytesPerSample || !mNumChannels)
+                        return ERROR_MALFORMED;
+
+                    size_t num_samples = mDataSize / (mNumChannels * bytesPerSample);
+
+                    if (!mSampleRate)
+                        return ERROR_MALFORMED;
+
+                    durationUs =
+                        1000000LL * num_samples / mSampleRate;
+                }
+
+                mTrackMeta.setInt64(kKeyDuration, durationUs);
+
+                return OK;
+            }
+        }
+
+        offset += chunkSize;
+    }
+
+    return NO_INIT;
+}
+
+const size_t WAVSource::kMaxFrameSize = 32768;
+
+WAVSource::WAVSource(
+        DataSourceBase *dataSource,
+        MetaDataBase &meta,
+        uint16_t waveFormat,
+        int32_t bitsPerSample,
+        off64_t offset, size_t size)
+    : mDataSource(dataSource),
+      mMeta(meta),
+      mWaveFormat(waveFormat),
+      mSampleRate(0),
+      mNumChannels(0),
+      mBitsPerSample(bitsPerSample),
+      mOffset(offset),
+      mSize(size),
+      mStarted(false),
+      mGroup(NULL) {
+    CHECK(mMeta.findInt32(kKeySampleRate, &mSampleRate));
+    CHECK(mMeta.findInt32(kKeyChannelCount, &mNumChannels));
+
+    mMeta.setInt32(kKeyMaxInputSize, kMaxFrameSize);
+}
+
+WAVSource::~WAVSource() {
+    if (mStarted) {
+        stop();
+    }
+}
+
+status_t WAVSource::start(MetaDataBase * /* params */) {
+    ALOGV("WAVSource::start");
+
+    CHECK(!mStarted);
+
+    // some WAV files may have large audio buffers that use shared memory transfer.
+    mGroup = new MediaBufferGroup(4 /* buffers */, kMaxFrameSize);
+
+    if (mBitsPerSample == 8) {
+        // As a temporary buffer for 8->16 bit conversion.
+        mGroup->add_buffer(MediaBufferBase::Create(kMaxFrameSize));
+    }
+
+    mCurrentPos = mOffset;
+
+    mStarted = true;
+
+    return OK;
+}
+
+status_t WAVSource::stop() {
+    ALOGV("WAVSource::stop");
+
+    CHECK(mStarted);
+
+    delete mGroup;
+    mGroup = NULL;
+
+    mStarted = false;
+
+    return OK;
+}
+
+status_t WAVSource::getFormat(MetaDataBase &meta) {
+    ALOGV("WAVSource::getFormat");
+
+    meta = mMeta;
+    return OK;
+}
+
+status_t WAVSource::read(
+        MediaBufferBase **out, const ReadOptions *options) {
+    *out = NULL;
+
+    if (options != nullptr && options->getNonBlocking() && !mGroup->has_buffers()) {
+        return WOULD_BLOCK;
+    }
+
+    int64_t seekTimeUs;
+    ReadOptions::SeekMode mode;
+    if (options != NULL && options->getSeekTo(&seekTimeUs, &mode)) {
+        int64_t pos = 0;
+
+        if (mWaveFormat == WAVE_FORMAT_MSGSM) {
+            // 65 bytes decode to 320 8kHz samples
+            int64_t samplenumber = (seekTimeUs * mSampleRate) / 1000000;
+            int64_t framenumber = samplenumber / 320;
+            pos = framenumber * 65;
+        } else {
+            pos = (seekTimeUs * mSampleRate) / 1000000 * mNumChannels * (mBitsPerSample >> 3);
+        }
+        if (pos > (off64_t)mSize) {
+            pos = mSize;
+        }
+        mCurrentPos = pos + mOffset;
+    }
+
+    MediaBufferBase *buffer;
+    status_t err = mGroup->acquire_buffer(&buffer);
+    if (err != OK) {
+        return err;
+    }
+
+    // make sure that maxBytesToRead is multiple of 3, in 24-bit case
+    size_t maxBytesToRead =
+        mBitsPerSample == 8 ? kMaxFrameSize / 2 : 
+        (mBitsPerSample == 24 ? 3*(kMaxFrameSize/3): kMaxFrameSize);
+
+    size_t maxBytesAvailable =
+        (mCurrentPos - mOffset >= (off64_t)mSize)
+            ? 0 : mSize - (mCurrentPos - mOffset);
+
+    if (maxBytesToRead > maxBytesAvailable) {
+        maxBytesToRead = maxBytesAvailable;
+    }
+
+    if (mWaveFormat == WAVE_FORMAT_MSGSM) {
+        // Microsoft packs 2 frames into 65 bytes, rather than using separate 33-byte frames,
+        // so read multiples of 65, and use smaller buffers to account for ~10:1 expansion ratio
+        if (maxBytesToRead > 1024) {
+            maxBytesToRead = 1024;
+        }
+        maxBytesToRead = (maxBytesToRead / 65) * 65;
+    } else {
+        // read only integral amounts of audio unit frames.
+        const size_t inputUnitFrameSize = mNumChannels * mBitsPerSample / 8;
+        maxBytesToRead -= maxBytesToRead % inputUnitFrameSize;
+    }
+
+    ssize_t n = mDataSource->readAt(
+            mCurrentPos, buffer->data(),
+            maxBytesToRead);
+
+    if (n <= 0) {
+        buffer->release();
+        buffer = NULL;
+
+        return ERROR_END_OF_STREAM;
+    }
+
+    buffer->set_range(0, n);
+
+    // TODO: add capability to return data as float PCM instead of 16 bit PCM.
+    if (mWaveFormat == WAVE_FORMAT_PCM) {
+        if (mBitsPerSample == 8) {
+            // Convert 8-bit unsigned samples to 16-bit signed.
+
+            // Create new buffer with 2 byte wide samples
+            MediaBufferBase *tmp;
+            CHECK_EQ(mGroup->acquire_buffer(&tmp), (status_t)OK);
+            tmp->set_range(0, 2 * n);
+
+            memcpy_to_i16_from_u8((int16_t *)tmp->data(), (const uint8_t *)buffer->data(), n);
+            buffer->release();
+            buffer = tmp;
+        } else if (mBitsPerSample == 24) {
+            // Convert 24-bit signed samples to 16-bit signed in place
+            const size_t numSamples = n / 3;
+
+            memcpy_to_i16_from_p24((int16_t *)buffer->data(), (const uint8_t *)buffer->data(), numSamples);
+            buffer->set_range(0, 2 * numSamples);
+        }  else if (mBitsPerSample == 32) {
+            // Convert 32-bit signed samples to 16-bit signed in place
+            const size_t numSamples = n / 4;
+
+            memcpy_to_i16_from_i32((int16_t *)buffer->data(), (const int32_t *)buffer->data(), numSamples);
+            buffer->set_range(0, 2 * numSamples);
+        }
+    } else if (mWaveFormat == WAVE_FORMAT_IEEE_FLOAT) {
+        if (mBitsPerSample == 32) {
+            // Convert 32-bit float samples to 16-bit signed in place
+            const size_t numSamples = n / 4;
+
+            memcpy_to_i16_from_float((int16_t *)buffer->data(), (const float *)buffer->data(), numSamples);
+            buffer->set_range(0, 2 * numSamples);
+        }
+    }
+
+    int64_t timeStampUs = 0;
+
+    if (mWaveFormat == WAVE_FORMAT_MSGSM) {
+        timeStampUs = 1000000LL * (mCurrentPos - mOffset) * 320 / 65 / mSampleRate;
+    } else {
+        size_t bytesPerSample = mBitsPerSample >> 3;
+        timeStampUs = 1000000LL * (mCurrentPos - mOffset)
+                / (mNumChannels * bytesPerSample) / mSampleRate;
+    }
+
+    buffer->meta_data().setInt64(kKeyTime, timeStampUs);
+
+    buffer->meta_data().setInt32(kKeyIsSyncFrame, 1);
+    mCurrentPos += n;
+
+    *out = buffer;
+
+    return OK;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+static MediaExtractor* CreateExtractor(
+        DataSourceBase *source,
+        void *) {
+    return new WAVExtractor(source);
+}
+
+static MediaExtractor::CreatorFunc Sniff(
+        DataSourceBase *source,
+        float *confidence,
+        void **,
+        MediaExtractor::FreeMetaFunc *) {
+    char header[12];
+    if (source->readAt(0, header, sizeof(header)) < (ssize_t)sizeof(header)) {
+        return NULL;
+    }
+
+    if (memcmp(header, "RIFF", 4) || memcmp(&header[8], "WAVE", 4)) {
+        return NULL;
+    }
+
+    MediaExtractor *extractor = new WAVExtractor(source);
+    int numTracks = extractor->countTracks();
+    delete extractor;
+    if (numTracks == 0) {
+        return NULL;
+    }
+
+    *confidence = 0.3f;
+
+    return CreateExtractor;
+}
+
+extern "C" {
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
+MediaExtractor::ExtractorDef GETEXTRACTORDEF() {
+    return {
+        MediaExtractor::EXTRACTORDEF_VERSION,
+        UUID("7d613858-5837-4a38-84c5-332d1cddee27"),
+        1, // version
+        "WAV Extractor",
+        Sniff
+    };
+}
+
+} // extern "C"
+
+} // namespace android

diff --git a/media/extractors/wav/WAVExtractor.h b/media/extractors/wav/WAVExtractor.h
new file mode 100644
index 0000000..467d0b7
--- /dev/null
+++ b/media/extractors/wav/WAVExtractor.h

@@ -0,0 +1,66 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef WAV_EXTRACTOR_H_
+
+#define WAV_EXTRACTOR_H_
+
+#include <utils/Errors.h>
+#include <media/MediaExtractor.h>
+#include <media/stagefright/MetaDataBase.h>
+
+namespace android {
+
+struct AMessage;
+class DataSourceBase;
+class String8;
+
+class WAVExtractor : public MediaExtractor {
+public:
+    explicit WAVExtractor(DataSourceBase *source);
+
+    virtual size_t countTracks();
+    virtual MediaTrack *getTrack(size_t index);
+    virtual status_t getTrackMetaData(MetaDataBase& meta, size_t index, uint32_t flags);
+
+    virtual status_t getMetaData(MetaDataBase& meta);
+    virtual const char * name() { return "WAVExtractor"; }
+
+    virtual ~WAVExtractor();
+
+private:
+    DataSourceBase *mDataSource;
+    status_t mInitCheck;
+    bool mValidFormat;
+    uint16_t mWaveFormat;
+    uint16_t mNumChannels;
+    uint32_t mChannelMask;
+    uint32_t mSampleRate;
+    uint16_t mBitsPerSample;
+    off64_t mDataOffset;
+    size_t mDataSize;
+    MetaDataBase mTrackMeta;
+
+    status_t init();
+
+    WAVExtractor(const WAVExtractor &);
+    WAVExtractor &operator=(const WAVExtractor &);
+};
+
+}  // namespace android
+
+#endif  // WAV_EXTRACTOR_H_
+

diff --git a/media/extractors/wav/exports.lds b/media/extractors/wav/exports.lds
new file mode 100644
index 0000000..b1309ee
--- /dev/null
+++ b/media/extractors/wav/exports.lds

@@ -0,0 +1 @@
+{ global: GETEXTRACTORDEF; local: *; };

diff --git a/media/img_utils/include/img_utils/DngUtils.h b/media/img_utils/include/img_utils/DngUtils.h
index 1d8df9c..de8f120 100644
--- a/media/img_utils/include/img_utils/DngUtils.h
+++ b/media/img_utils/include/img_utils/DngUtils.h

@@ -39,11 +39,16 @@
  */
 class ANDROID_API OpcodeListBuilder : public LightRefBase<OpcodeListBuilder> {
     public:
+        // Note that the Adobe DNG 1.4 spec for Bayer phase (defined for the
+        // FixBadPixelsConstant and FixBadPixelsList opcodes) is incorrect. It's
+        // inconsistent with the DNG SDK (cf. dng_negative::SetBayerMosaic and
+        // dng_opcode_FixBadPixelsList::IsGreen), and Adobe confirms that the
+        // spec should be updated to match the SDK.
         enum CfaLayout {
-            CFA_RGGB = 0,
-            CFA_GRBG,
-            CFA_GBRG,
+            CFA_GRBG = 0,
+            CFA_RGGB,
             CFA_BGGR,
+            CFA_GBRG,
         };
 
         OpcodeListBuilder();

diff --git a/media/img_utils/src/DngUtils.cpp b/media/img_utils/src/DngUtils.cpp
index 9dc5f05..9ac7e2a 100644
--- a/media/img_utils/src/DngUtils.cpp
+++ b/media/img_utils/src/DngUtils.cpp

@@ -18,6 +18,7 @@
 
 #include <inttypes.h>
 
+#include <algorithm>
 #include <vector>
 #include <math.h>
 
@@ -61,8 +62,8 @@
                                                    const float* lensShadingMap) {
     uint32_t activeAreaWidth = activeAreaRight - activeAreaLeft;
     uint32_t activeAreaHeight = activeAreaBottom - activeAreaTop;
-    double spacingV = 1.0 / lsmHeight;
-    double spacingH = 1.0 / lsmWidth;
+    double spacingV = 1.0 / std::max(1u, lsmHeight - 1);
+    double spacingH = 1.0 / std::max(1u, lsmWidth - 1);
 
     std::vector<float> redMapVector(lsmWidth * lsmHeight);
     float *redMap = redMapVector.data();
@@ -301,29 +302,14 @@
     normalizedOCX = CLAMP(normalizedOCX, 0, 1);
     normalizedOCY = CLAMP(normalizedOCY, 0, 1);
 
-    // Conversion factors from Camera2 K factors to DNG spec. K factors:
-    //
-    //      Note: these are necessary because our unit system assumes a
-    //      normalized max radius of sqrt(2), whereas the DNG spec's
-    //      WarpRectilinear opcode assumes a normalized max radius of 1.
-    //      Thus, each K coefficient must include the domain scaling
-    //      factor (the DNG domain is scaled by sqrt(2) to emulate the
-    //      domain used by the Camera2 specification).
-
-    const double c_0 = sqrt(2);
-    const double c_1 = 2 * sqrt(2);
-    const double c_2 = 4 * sqrt(2);
-    const double c_3 = 8 * sqrt(2);
-    const double c_4 = 2;
-    const double c_5 = 2;
-
-    const double coeffs[] = { c_0 * kCoeffs[0],
-                              c_1 * kCoeffs[1],
-                              c_2 * kCoeffs[2],
-                              c_3 * kCoeffs[3],
-                              c_4 * kCoeffs[4],
-                              c_5 * kCoeffs[5] };
-
+    double coeffs[6] = {
+        kCoeffs[0],
+        kCoeffs[1],
+        kCoeffs[2],
+        kCoeffs[3],
+        kCoeffs[4],
+        kCoeffs[5]
+    };
 
     return addWarpRectilinear(/*numPlanes*/1,
                               /*opticalCenterX*/normalizedOCX,

diff --git a/media/img_utils/src/TiffWriter.cpp b/media/img_utils/src/TiffWriter.cpp
index 564474f..1711242 100644
--- a/media/img_utils/src/TiffWriter.cpp
+++ b/media/img_utils/src/TiffWriter.cpp

@@ -350,7 +350,7 @@
         if (nextIfd == NULL) {
             break;
         }
-        ifd = nextIfd;
+        ifd = std::move(nextIfd);
     }
     return ifd;
 }

diff --git a/media/libaaudio/Android.bp b/media/libaaudio/Android.bp
index 6e60f24..f00f7a8 100644
--- a/media/libaaudio/Android.bp
+++ b/media/libaaudio/Android.bp

@@ -28,3 +28,10 @@
     first_version: "26",
     unversioned_until: "current",
 }
+
+cc_library_headers {
+    name: "libaaudio_headers",
+    export_include_dirs: ["include"],
+}
+
+subdirs = ["*"]

diff --git a/media/libaaudio/Android.mk b/media/libaaudio/Android.mk
deleted file mode 100644
index 5053e7d..0000000
--- a/media/libaaudio/Android.mk
+++ /dev/null

@@ -1 +0,0 @@
-include $(call all-subdir-makefiles)

diff --git a/media/libaaudio/examples/Android.bp b/media/libaaudio/examples/Android.bp
new file mode 100644
index 0000000..639fab2
--- /dev/null
+++ b/media/libaaudio/examples/Android.bp

@@ -0,0 +1,6 @@
+subdirs = ["*"]
+
+cc_library_headers {
+    name: "libaaudio_example_utils",
+    export_include_dirs: ["utils"],
+}

diff --git a/media/libaaudio/examples/Android.mk b/media/libaaudio/examples/Android.mk
deleted file mode 100644
index 5053e7d..0000000
--- a/media/libaaudio/examples/Android.mk
+++ /dev/null

@@ -1 +0,0 @@
-include $(call all-subdir-makefiles)

diff --git a/media/libaaudio/examples/input_monitor/Android.bp b/media/libaaudio/examples/input_monitor/Android.bp
new file mode 100644
index 0000000..d8c5843
--- /dev/null
+++ b/media/libaaudio/examples/input_monitor/Android.bp

@@ -0,0 +1,17 @@
+cc_test {
+    name: "input_monitor",
+    gtest: false,
+    srcs: ["src/input_monitor.cpp"],
+    cflags: ["-Wall", "-Werror"],
+    shared_libs: ["libaaudio"],
+    header_libs: ["libaaudio_example_utils"],
+}
+
+cc_test {
+    name: "input_monitor_callback",
+    gtest: false,
+    srcs: ["src/input_monitor_callback.cpp"],
+    cflags: ["-Wall", "-Werror"],
+    shared_libs: ["libaaudio"],
+    header_libs: ["libaaudio_example_utils"],
+}

diff --git a/media/libaaudio/examples/input_monitor/Android.mk b/media/libaaudio/examples/input_monitor/Android.mk
deleted file mode 100644
index 5053e7d..0000000
--- a/media/libaaudio/examples/input_monitor/Android.mk
+++ /dev/null

@@ -1 +0,0 @@
-include $(call all-subdir-makefiles)

diff --git a/media/libaaudio/examples/input_monitor/jni/Android.mk b/media/libaaudio/examples/input_monitor/jni/Android.mk
index 9b1ce2c..a0b981c 100644
--- a/media/libaaudio/examples/input_monitor/jni/Android.mk
+++ b/media/libaaudio/examples/input_monitor/jni/Android.mk

@@ -10,6 +10,7 @@
 
 # NDK recommends using this kind of relative path instead of an absolute path.
 LOCAL_SRC_FILES:= ../src/input_monitor.cpp
+LOCAL_CFLAGS := -Wall -Werror
 LOCAL_SHARED_LIBRARIES := libaaudio
 LOCAL_MODULE := input_monitor
 include $(BUILD_EXECUTABLE)
@@ -22,6 +23,7 @@
     frameworks/av/media/libaaudio/examples/utils
 
 LOCAL_SRC_FILES:= ../src/input_monitor_callback.cpp
+LOCAL_CFLAGS := -Wall -Werror
 LOCAL_SHARED_LIBRARIES := libaaudio
 LOCAL_MODULE := input_monitor_callback
 include $(BUILD_EXECUTABLE)

diff --git a/media/libaaudio/examples/input_monitor/src/input_monitor.cpp b/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
index 9feb118..c1ff34b 100644
--- a/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
+++ b/media/libaaudio/examples/input_monitor/src/input_monitor.cpp

@@ -26,24 +26,22 @@
 #include "AAudioExampleUtils.h"
 #include "AAudioSimpleRecorder.h"
 
-// TODO support FLOAT
-#define REQUIRED_FORMAT    AAUDIO_FORMAT_PCM_I16
 #define MIN_FRAMES_TO_READ 48  /* arbitrary, 1 msec at 48000 Hz */
 
 static const int FRAMES_PER_LINE = 20000;
 
 int main(int argc, const char **argv)
 {
-    AAudioArgsParser   argParser;
-    aaudio_result_t result;
-    AAudioSimpleRecorder recorder;
-    int actualSamplesPerFrame;
-    int actualSampleRate;
-    aaudio_format_t       actualDataFormat;
-    aaudio_sharing_mode_t actualSharingMode;
+    AAudioArgsParser      argParser;
+    AAudioSimpleRecorder  recorder;
+    AAudioStream         *aaudioStream = nullptr;
 
-    AAudioStream *aaudioStream = nullptr;
+    aaudio_result_t       result;
+    aaudio_format_t       actualDataFormat;
     aaudio_stream_state_t state;
+
+    int32_t actualSamplesPerFrame;
+    int32_t actualSampleRate;
     int32_t framesPerBurst = 0;
     int32_t framesPerRead = 0;
     int32_t framesToRecord = 0;
@@ -51,19 +49,18 @@
     int32_t nextFrameCount = 0;
     int32_t frameCount = 0;
     int32_t xRunCount = 0;
-    int64_t previousFramePosition = -1;
-    int16_t *data = nullptr;
-    float peakLevel = 0.0;
-    int loopCounter = 0;
     int32_t deviceId;
 
+    int16_t *shortData = nullptr;
+    float   *floatData = nullptr;
+    float    peakLevel = 0.0;
+
     // Make printf print immediately so that debug info is not stuck
     // in a buffer if we hang or crash.
     setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
 
-    printf("%s - Monitor input level using AAudio read, V0.1.2\n", argv[0]);
+    printf("%s - Monitor input level using AAudio read, V0.1.3\n", argv[0]);
 
-    argParser.setFormat(REQUIRED_FORMAT);
     if (argParser.parseArgs(argc, argv)) {
         return EXIT_FAILURE;
     }
@@ -71,6 +68,7 @@
     result = recorder.open(argParser);
     if (result != AAUDIO_OK) {
         fprintf(stderr, "ERROR -  recorder.open() returned %d\n", result);
+        printf("IMPORTANT - Did you remember to enter:   adb root\n");
         goto finish;
     }
     aaudioStream = recorder.getStream();
@@ -98,17 +96,18 @@
     printf("DataFormat: framesPerRead  = %d\n",framesPerRead);
 
     actualDataFormat = AAudioStream_getFormat(aaudioStream);
-    printf("DataFormat: requested      = %d, actual = %d\n",
-           REQUIRED_FORMAT, actualDataFormat);
-    // TODO handle other data formats
-    assert(actualDataFormat == REQUIRED_FORMAT);
 
     // Allocate a buffer for the PCM_16 audio data.
-    data = new(std::nothrow) int16_t[framesPerRead * actualSamplesPerFrame];
-    if (data == nullptr) {
-        fprintf(stderr, "ERROR - could not allocate data buffer\n");
-        result = AAUDIO_ERROR_NO_MEMORY;
-        goto finish;
+    switch (actualDataFormat) {
+        case AAUDIO_FORMAT_PCM_I16:
+            shortData = new int16_t[framesPerRead * actualSamplesPerFrame];
+            break;
+        case AAUDIO_FORMAT_PCM_FLOAT:
+            floatData = new float[framesPerRead * actualSamplesPerFrame];
+            break;
+        default:
+            fprintf(stderr, "UNEXPECTED FORMAT! %d", actualDataFormat);
+            goto finish;
     }
 
     // Start the stream.
@@ -128,7 +127,12 @@
         // Read audio data from the stream.
         const int64_t timeoutNanos = 1000 * NANOS_PER_MILLISECOND;
         int minFrames = (framesToRecord < framesPerRead) ? framesToRecord : framesPerRead;
-        int actual = AAudioStream_read(aaudioStream, data, minFrames, timeoutNanos);
+        int actual = 0;
+        if (actualDataFormat == AAUDIO_FORMAT_PCM_I16) {
+            actual = AAudioStream_read(aaudioStream, shortData, minFrames, timeoutNanos);
+        } else if (actualDataFormat == AAUDIO_FORMAT_PCM_FLOAT) {
+            actual = AAudioStream_read(aaudioStream, floatData, minFrames, timeoutNanos);
+        }
         if (actual < 0) {
             fprintf(stderr, "ERROR - AAudioStream_read() returned %d\n", actual);
             result = actual;
@@ -142,7 +146,12 @@
 
         // Peak finder.
         for (int frameIndex = 0; frameIndex < actual; frameIndex++) {
-            float sample = data[frameIndex * actualSamplesPerFrame] * (1.0/32768);
+            float sample = 0.0f;
+            if (actualDataFormat == AAUDIO_FORMAT_PCM_I16) {
+                sample = shortData[frameIndex * actualSamplesPerFrame] * (1.0/32768);
+            } else if (actualDataFormat == AAUDIO_FORMAT_PCM_FLOAT) {
+                sample = floatData[frameIndex * actualSamplesPerFrame];
+            }
             if (sample > peakLevel) {
                 peakLevel = sample;
             }
@@ -153,17 +162,15 @@
             displayPeakLevel(peakLevel);
             peakLevel = 0.0;
             nextFrameCount += FRAMES_PER_LINE;
-        }
 
-        // Print timestamps.
-        int64_t framePosition = 0;
-        int64_t frameTime = 0;
-        aaudio_result_t timeResult;
-        timeResult = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC,
-                                               &framePosition, &frameTime);
+            // Print timestamps.
+            int64_t framePosition = 0;
+            int64_t frameTime = 0;
+            aaudio_result_t timeResult;
+            timeResult = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC,
+                                                   &framePosition, &frameTime);
 
-        if (timeResult == AAUDIO_OK) {
-            if (framePosition > (previousFramePosition + FRAMES_PER_LINE)) {
+            if (timeResult == AAUDIO_OK) {
                 int64_t realTime = getNanoseconds();
                 int64_t framesRead = AAudioStream_getFramesRead(aaudioStream);
 
@@ -177,11 +184,15 @@
                        (long long) framePosition,
                        (long long) frameTime,
                        latencyMillis);
-                previousFramePosition = framePosition;
+            } else {
+                printf("WARNING - AAudioStream_getTimestamp() returned %d\n", timeResult);
             }
         }
     }
 
+    state = AAudioStream_getState(aaudioStream);
+    printf("after loop, state = %s\n", AAudio_convertStreamStateToText(state));
+
     xRunCount = AAudioStream_getXRunCount(aaudioStream);
     printf("AAudioStream_getXRunCount %d\n", xRunCount);
 
@@ -194,7 +205,8 @@
 
 finish:
     recorder.close();
-    delete[] data;
+    delete[] shortData;
+    delete[] floatData;
     printf("exiting - AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
     return (result != AAUDIO_OK) ? EXIT_FAILURE : EXIT_SUCCESS;
 }

diff --git a/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp b/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp
index 893795b..d10f812 100644
--- a/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp
+++ b/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp

@@ -26,29 +26,39 @@
 #include "AAudioExampleUtils.h"
 #include "AAudioSimpleRecorder.h"
 
-#define NUM_SECONDS           5
-
-int main(int argc, char **argv)
+int main(int argc, const char **argv)
 {
-    (void)argc; // unused
-    AAudioSimpleRecorder recorder;
-    PeakTrackerData_t myData = {0.0};
-    aaudio_result_t result;
+    AAudioArgsParser      argParser;
+    AAudioSimpleRecorder  recorder;
+    PeakTrackerData_t     myData = {0.0};
+    AAudioStream         *aaudioStream = nullptr;
+    aaudio_result_t       result;
     aaudio_stream_state_t state;
+
+    int       loopsNeeded = 0;
     const int displayRateHz = 20; // arbitrary
-    const int loopsNeeded = NUM_SECONDS * displayRateHz;
 
     // Make printf print immediately so that debug info is not stuck
     // in a buffer if we hang or crash.
     setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
-    printf("%s - Display audio input using an AAudio callback, V0.1.2\n", argv[0]);
+    printf("%s - Display audio input using an AAudio callback, V0.1.3\n", argv[0]);
 
-    result = recorder.open(2, 48000, AAUDIO_FORMAT_PCM_I16,
-                       SimpleRecorderDataCallbackProc, SimpleRecorderErrorCallbackProc, &myData);
+    if (argParser.parseArgs(argc, argv)) {
+        return EXIT_FAILURE;
+    }
+
+    result = recorder.open(argParser,
+                           SimpleRecorderDataCallbackProc,
+                           SimpleRecorderErrorCallbackProc,
+                           &myData);
     if (result != AAUDIO_OK) {
         fprintf(stderr, "ERROR -  recorder.open() returned %d\n", result);
+        printf("IMPORTANT - Did you remember to enter:   adb root\n");
         goto error;
     }
+    aaudioStream = recorder.getStream();
+    argParser.compareWithStream(aaudioStream);
+
     printf("recorder.getFramesPerSecond() = %d\n", recorder.getFramesPerSecond());
     printf("recorder.getSamplesPerFrame() = %d\n", recorder.getSamplesPerFrame());
 
@@ -58,7 +68,9 @@
         goto error;
     }
 
-    printf("Sleep for %d seconds while audio record in a callback thread.\n", NUM_SECONDS);
+    printf("Sleep for %d seconds while audio record in a callback thread.\n",
+           argParser.getDurationSeconds());
+    loopsNeeded = argParser.getDurationSeconds() * displayRateHz;
     for (int i = 0; i < loopsNeeded; i++)
     {
         const struct timespec request = { .tv_sec = 0,
@@ -67,7 +79,7 @@
         printf("%08d: ", (int)recorder.getFramesRead());
         displayPeakLevel(myData.peakLevel);
 
-        result = AAudioStream_waitForStateChange(recorder.getStream(),
+        result = AAudioStream_waitForStateChange(aaudioStream,
                                                  AAUDIO_STREAM_STATE_CLOSED,
                                                  &state,
                                                  0);
@@ -93,7 +105,8 @@
         goto error;
     }
 
-    printf("Sleep for %d seconds while audio records in a callback thread.\n", NUM_SECONDS);
+    printf("Sleep for %d seconds while audio records in a callback thread.\n",
+           argParser.getDurationSeconds());
     for (int i = 0; i < loopsNeeded; i++)
     {
         const struct timespec request = { .tv_sec = 0,
@@ -102,13 +115,14 @@
         printf("%08d: ", (int)recorder.getFramesRead());
         displayPeakLevel(myData.peakLevel);
 
-        state = AAudioStream_getState(recorder.getStream());
+        state = AAudioStream_getState(aaudioStream);
         if (state != AAUDIO_STREAM_STATE_STARTING && state != AAUDIO_STREAM_STATE_STARTED) {
             printf("Stream state is %d %s!\n", state, AAudio_convertStreamStateToText(state));
             break;
         }
     }
     printf("Woke up now.\n");
+    argParser.compareWithStream(aaudioStream);
 
     result = recorder.stop();
     if (result != AAUDIO_OK) {

diff --git a/media/libaaudio/examples/loopback/Android.bp b/media/libaaudio/examples/loopback/Android.bp
new file mode 100644
index 0000000..5b7d956
--- /dev/null
+++ b/media/libaaudio/examples/loopback/Android.bp

@@ -0,0 +1,12 @@
+cc_test {
+    name: "aaudio_loopback",
+    gtest: false,
+    srcs: ["src/loopback.cpp"],
+    cflags: ["-Wall", "-Werror"],
+    static_libs: ["libsndfile"],
+    shared_libs: [
+        "libaaudio",
+        "libaudioutils",
+        ],
+    header_libs: ["libaaudio_example_utils"],
+}

diff --git a/media/libaaudio/examples/loopback/Android.mk b/media/libaaudio/examples/loopback/Android.mk
deleted file mode 100644
index 5053e7d..0000000
--- a/media/libaaudio/examples/loopback/Android.mk
+++ /dev/null

@@ -1 +0,0 @@
-include $(call all-subdir-makefiles)

diff --git a/media/libaaudio/examples/loopback/jni/Android.mk b/media/libaaudio/examples/loopback/jni/Android.mk
index d78f286..aebe877 100644
--- a/media/libaaudio/examples/loopback/jni/Android.mk
+++ b/media/libaaudio/examples/loopback/jni/Android.mk

@@ -9,6 +9,8 @@
 
 # NDK recommends using this kind of relative path instead of an absolute path.
 LOCAL_SRC_FILES:= ../src/loopback.cpp
-LOCAL_SHARED_LIBRARIES := libaaudio
+LOCAL_CFLAGS := -Wall -Werror
+LOCAL_STATIC_LIBRARIES := libsndfile
+LOCAL_SHARED_LIBRARIES := libaaudio libaudioutils
 LOCAL_MODULE := aaudio_loopback
 include $(BUILD_EXECUTABLE)

diff --git a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
index 21cf341..ef9a753 100644
--- a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
+++ b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h

@@ -30,6 +30,8 @@
 #include <stdlib.h>
 #include <unistd.h>
 
+#include <audio_utils/sndfile.h>
+
 // Tag for machine readable results as property = value pairs
 #define LOOPBACK_RESULT_TAG      "RESULT: "
 #define LOOPBACK_SAMPLE_RATE     48000
@@ -37,13 +39,18 @@
 #define MILLIS_PER_SECOND        1000
 
 #define MAX_ZEROTH_PARTIAL_BINS  40
+constexpr double MAX_ECHO_GAIN = 10.0; // based on experiments, otherwise autocorrelation too noisy
 
+// A narrow impulse seems to have better immunity against over estimating the
+// latency due to detecting subharmonics by the auto-correlator.
 static const float s_Impulse[] = {
-        0.0f, 0.0f, 0.0f, 0.0f, 0.2f, // silence on each side of the impulse
-        0.5f, 0.9999f, 0.0f, -0.9999, -0.5f, // bipolar
-        -0.2f, 0.0f, 0.0f, 0.0f, 0.0f
+        0.0f, 0.0f, 0.0f, 0.0f, 0.3f, // silence on each side of the impulse
+        0.99f, 0.0f, -0.99f, // bipolar with one zero crossing in middle
+        -0.3f, 0.0f, 0.0f, 0.0f, 0.0f
 };
 
+constexpr int32_t kImpulseSizeInFrames = (int32_t)(sizeof(s_Impulse) / sizeof(s_Impulse[0]));
+
 class PseudoRandom {
 public:
     PseudoRandom() {}
@@ -156,6 +163,8 @@
                             const float *needle, int needleSize,
                             LatencyReport *report) {
     const double threshold = 0.1;
+    printf("measureLatencyFromEchos: haystackSize = %d, needleSize = %d\n",
+           haystackSize, needleSize);
 
     // Find first peak
     int first = (int) (findFirstMatch(haystack,
@@ -173,7 +182,7 @@
                                       needleSize,
                                       threshold) + 0.5);
 
-    printf("first = %d, again at %d\n", first, again);
+    printf("measureLatencyFromEchos: first = %d, again at %d\n", first, again);
     first = again;
 
     // Allocate results array
@@ -270,37 +279,60 @@
         return mData;
     }
 
+    void setSampleRate(int32_t sampleRate) {
+        mSampleRate = sampleRate;
+    }
+
+    int32_t getSampleRate() {
+        return mSampleRate;
+    }
+
     int save(const char *fileName, bool writeShorts = true) {
+        SNDFILE *sndFile = nullptr;
         int written = 0;
-        const int chunkSize = 64;
-        FILE *fid = fopen(fileName, "wb");
-        if (fid == NULL) {
+        SF_INFO info = {
+                .frames = mFrameCounter,
+                .samplerate = mSampleRate,
+                .channels = 1,
+                .format = SF_FORMAT_WAV | (writeShorts ? SF_FORMAT_PCM_16 : SF_FORMAT_FLOAT)
+        };
+
+        sndFile = sf_open(fileName, SFM_WRITE, &info);
+        if (sndFile == nullptr) {
+            printf("AudioRecording::save(%s) failed to open file\n", fileName);
             return -errno;
         }
 
-        if (writeShorts) {
-            int16_t buffer[chunkSize];
-            int32_t framesLeft = mFrameCounter;
-            int32_t cursor = 0;
-            while (framesLeft) {
-                int32_t framesToWrite = framesLeft < chunkSize ? framesLeft : chunkSize;
-                for (int i = 0; i < framesToWrite; i++) {
-                    buffer[i] = (int16_t) (mData[cursor++] * 32767);
-                }
-                written += fwrite(buffer, sizeof(int16_t), framesToWrite, fid);
-                framesLeft -= framesToWrite;
-            }
-        } else {
-            written = (int) fwrite(mData, sizeof(float), mFrameCounter, fid);
-        }
-        fclose(fid);
+        written = sf_writef_float(sndFile, mData, mFrameCounter);
+
+        sf_close(sndFile);
         return written;
     }
 
+    int load(const char *fileName) {
+        SNDFILE *sndFile = nullptr;
+        SF_INFO info;
+
+        sndFile = sf_open(fileName, SFM_READ, &info);
+        if (sndFile == nullptr) {
+            printf("AudioRecording::load(%s) failed to open file\n", fileName);
+            return -errno;
+        }
+
+        assert(info.channels == 1);
+
+        allocate(info.frames);
+        mFrameCounter = sf_readf_float(sndFile, mData, info.frames);
+
+        sf_close(sndFile);
+        return mFrameCounter;
+    }
+
 private:
     float  *mData = nullptr;
     int32_t mFrameCounter = 0;
     int32_t mMaxFrames = 0;
+    int32_t mSampleRate = 48000; // common default
 };
 
 // ====================================================================================
@@ -320,11 +352,25 @@
 
     virtual void printStatus() {};
 
+    virtual int getResult() {
+        return -1;
+    }
+
     virtual bool isDone() {
         return false;
     }
 
-    void setSampleRate(int32_t sampleRate) {
+    virtual int save(const char *fileName) {
+        (void) fileName;
+        return AAUDIO_ERROR_UNIMPLEMENTED;
+    }
+
+    virtual int load(const char *fileName) {
+        (void) fileName;
+        return AAUDIO_ERROR_UNIMPLEMENTED;
+    }
+
+    virtual void setSampleRate(int32_t sampleRate) {
         mSampleRate = sampleRate;
     }
 
@@ -368,8 +414,7 @@
 
 
 static void printAudioScope(float sample) {
-    const int maxStars = 80
-    ; // arbitrary, fits on one line
+    const int maxStars = 80; // arbitrary, fits on one line
     char c = '*';
     if (sample < -1.0) {
         sample = -1.0;
@@ -395,7 +440,13 @@
 public:
 
     EchoAnalyzer() : LoopbackProcessor() {
-        audioRecorder.allocate(2 * LOOPBACK_SAMPLE_RATE);
+        mAudioRecording.allocate(2 * getSampleRate());
+        mAudioRecording.setSampleRate(getSampleRate());
+    }
+
+    void setSampleRate(int32_t sampleRate) override {
+        LoopbackProcessor::setSampleRate(sampleRate);
+        mAudioRecording.setSampleRate(sampleRate);
     }
 
     void reset() override {
@@ -406,8 +457,12 @@
         mState = STATE_INITIAL_SILENCE;
     }
 
+    virtual int getResult() {
+        return mState == STATE_DONE ? 0 : -1;
+    }
+
     virtual bool isDone() {
-        return mState == STATE_DONE;
+        return mState == STATE_DONE || mState == STATE_FAILED;
     }
 
     void setGain(float gain) {
@@ -423,46 +478,47 @@
         printf("EchoAnalyzer ---------------\n");
         printf(LOOPBACK_RESULT_TAG "measured.gain          = %f\n", mMeasuredLoopGain);
         printf(LOOPBACK_RESULT_TAG "echo.gain              = %f\n", mEchoGain);
-        printf(LOOPBACK_RESULT_TAG "frame.count            = %d\n", mFrameCounter);
         printf(LOOPBACK_RESULT_TAG "test.state             = %d\n", mState);
         if (mMeasuredLoopGain >= 0.9999) {
             printf("   ERROR - clipping, turn down volume slightly\n");
         } else {
             const float *needle = s_Impulse;
             int needleSize = (int) (sizeof(s_Impulse) / sizeof(float));
-            float *haystack = audioRecorder.getData();
-            int haystackSize = audioRecorder.size();
-            int result = measureLatencyFromEchos(haystack, haystackSize,
-                                                 needle, needleSize,
-                                                 &latencyReport);
-            if (latencyReport.confidence < 0.01) {
-                printf("   ERROR - confidence too low = %f\n", latencyReport.confidence);
+            float *haystack = mAudioRecording.getData();
+            int haystackSize = mAudioRecording.size();
+            measureLatencyFromEchos(haystack, haystackSize, needle, needleSize, &mLatencyReport);
+            if (mLatencyReport.confidence < 0.01) {
+                printf("   ERROR - confidence too low = %f\n", mLatencyReport.confidence);
             } else {
-                double latencyMillis = 1000.0 * latencyReport.latencyInFrames / getSampleRate();
-                printf(LOOPBACK_RESULT_TAG "latency.frames        = %8.2f\n", latencyReport.latencyInFrames);
+                double latencyMillis = 1000.0 * mLatencyReport.latencyInFrames / getSampleRate();
+                printf(LOOPBACK_RESULT_TAG "latency.frames        = %8.2f\n", mLatencyReport.latencyInFrames);
                 printf(LOOPBACK_RESULT_TAG "latency.msec          = %8.2f\n", latencyMillis);
-                printf(LOOPBACK_RESULT_TAG "latency.confidence    = %8.6f\n", latencyReport.confidence);
+                printf(LOOPBACK_RESULT_TAG "latency.confidence    = %8.6f\n", mLatencyReport.confidence);
             }
         }
-
-        {
-#define ECHO_FILENAME "/data/oboe_echo.raw"
-            int written = audioRecorder.save(ECHO_FILENAME);
-            printf("Echo wrote %d mono samples to %s on Android device\n", written, ECHO_FILENAME);
-        }
     }
 
     void printStatus() override {
-        printf("state = %d, echo gain = %f ", mState, mEchoGain);
+        printf("st = %d, echo gain = %f ", mState, mEchoGain);
     }
 
-    static void sendImpulse(float *outputData, int outputChannelCount) {
-        for (float sample : s_Impulse) {
+    void sendImpulses(float *outputData, int outputChannelCount, int numFrames) {
+        while (numFrames-- > 0) {
+            float sample = s_Impulse[mSampleIndex++];
+            if (mSampleIndex >= kImpulseSizeInFrames) {
+                mSampleIndex = 0;
+            }
+
             *outputData = sample;
             outputData += outputChannelCount;
         }
     }
 
+    void sendOneImpulse(float *outputData, int outputChannelCount) {
+        mSampleIndex = 0;
+        sendImpulses(outputData, outputChannelCount, kImpulseSizeInFrames);
+    }
+
     void process(float *inputData, int inputChannelCount,
                  float *outputData, int outputChannelCount,
                  int numFrames) override {
@@ -488,26 +544,31 @@
                 break;
 
             case STATE_MEASURING_GAIN:
-                sendImpulse(outputData, outputChannelCount);
+                sendImpulses(outputData, outputChannelCount, numFrames);
                 peak = measurePeakAmplitude(inputData, inputChannelCount, numFrames);
                 // If we get several in a row then go to next state.
                 if (peak > mPulseThreshold) {
                     if (mDownCounter-- <= 0) {
-                        nextState = STATE_WAITING_FOR_SILENCE;
                         //printf("%5d: switch to STATE_WAITING_FOR_SILENCE, measured peak = %f\n",
                         //       mLoopCounter, peak);
                         mDownCounter = 8;
                         mMeasuredLoopGain = peak;  // assumes original pulse amplitude is one
                         // Calculate gain that will give us a nice decaying echo.
                         mEchoGain = mDesiredEchoGain / mMeasuredLoopGain;
+                        if (mEchoGain > MAX_ECHO_GAIN) {
+                            printf("ERROR - loop gain too low. Increase the volume.\n");
+                            nextState = STATE_FAILED;
+                        } else {
+                            nextState = STATE_WAITING_FOR_SILENCE;
+                        }
                     }
-                } else {
+                } else if (numFrames > kImpulseSizeInFrames){ // ignore short callbacks
                     mDownCounter = 8;
                 }
                 break;
 
             case STATE_WAITING_FOR_SILENCE:
-                // Output silence.
+                // Output silence and wait for the echos to die down.
                 numSamples = numFrames * outputChannelCount;
                 for (int i = 0; i < numSamples; i++) {
                     outputData[i] = 0;
@@ -526,14 +587,14 @@
                 break;
 
             case STATE_SENDING_PULSE:
-                audioRecorder.write(inputData, inputChannelCount, numFrames);
-                sendImpulse(outputData, outputChannelCount);
+                mAudioRecording.write(inputData, inputChannelCount, numFrames);
+                sendOneImpulse(outputData, outputChannelCount);
                 nextState = STATE_GATHERING_ECHOS;
                 //printf("%5d: switch to STATE_GATHERING_ECHOS\n", mLoopCounter);
                 break;
 
             case STATE_GATHERING_ECHOS:
-                numWritten = audioRecorder.write(inputData, inputChannelCount, numFrames);
+                numWritten = mAudioRecording.write(inputData, inputChannelCount, numFrames);
                 peak = measurePeakAmplitude(inputData, inputChannelCount, numFrames);
                 if (peak > mMeasuredLoopGain) {
                     mMeasuredLoopGain = peak;  // AGC might be raising gain so adjust it on the fly.
@@ -567,6 +628,14 @@
         mLoopCounter++;
     }
 
+    int save(const char *fileName) override {
+        return mAudioRecording.save(fileName);
+    }
+
+    int load(const char *fileName) override {
+        return mAudioRecording.load(fileName);
+    }
+
 private:
 
     enum echo_state_t {
@@ -575,23 +644,23 @@
         STATE_WAITING_FOR_SILENCE,
         STATE_SENDING_PULSE,
         STATE_GATHERING_ECHOS,
-        STATE_DONE
+        STATE_DONE,
+        STATE_FAILED
     };
 
-    int           mDownCounter = 500;
-    int           mLoopCounter = 0;
-    int           mLoopStart = 1000;
-    float         mPulseThreshold = 0.02f;
-    float         mSilenceThreshold = 0.002f;
-    float         mMeasuredLoopGain = 0.0f;
-    float         mDesiredEchoGain = 0.95f;
-    float         mEchoGain = 1.0f;
-    echo_state_t  mState = STATE_INITIAL_SILENCE;
-    int32_t       mFrameCounter = 0;
+    int32_t         mDownCounter = 500;
+    int32_t         mLoopCounter = 0;
+    int32_t         mSampleIndex = 0;
+    float           mPulseThreshold = 0.02f;
+    float           mSilenceThreshold = 0.002f;
+    float           mMeasuredLoopGain = 0.0f;
+    float           mDesiredEchoGain = 0.95f;
+    float           mEchoGain = 1.0f;
+    echo_state_t    mState = STATE_INITIAL_SILENCE;
 
-    AudioRecording     audioRecorder;
-    LatencyReport      latencyReport;
-    PeakDetector       mPeakDetector;
+    AudioRecording  mAudioRecording; // contains only the input after the gain detection burst
+    LatencyReport   mLatencyReport;
+    // PeakDetector    mPeakDetector;
 };
 
 
@@ -605,6 +674,10 @@
 class SineAnalyzer : public LoopbackProcessor {
 public:
 
+    virtual int getResult() {
+        return mState == STATE_LOCKED ? 0 : -1;
+    }
+
     void report() override {
         printf("SineAnalyzer ------------------\n");
         printf(LOOPBACK_RESULT_TAG "peak.amplitude     = %7.5f\n", mPeakAmplitude);
@@ -612,7 +685,7 @@
         printf(LOOPBACK_RESULT_TAG "phase.offset       = %7.5f\n", mPhaseOffset);
         printf(LOOPBACK_RESULT_TAG "ref.phase          = %7.5f\n", mPhase);
         printf(LOOPBACK_RESULT_TAG "frames.accumulated = %6d\n", mFramesAccumulated);
-        printf(LOOPBACK_RESULT_TAG "sine.period        = %6d\n", mPeriod);
+        printf(LOOPBACK_RESULT_TAG "sine.period        = %6d\n", mSinePeriod);
         printf(LOOPBACK_RESULT_TAG "test.state         = %6d\n", mState);
         printf(LOOPBACK_RESULT_TAG "frame.count        = %6d\n", mFrameCounter);
         // Did we ever get a lock?
@@ -626,7 +699,7 @@
     }
 
     void printStatus() override {
-        printf("  state = %d, glitches = %d,", mState, mGlitchCount);
+        printf("st = %d, #gl = %3d,", mState, mGlitchCount);
     }
 
     double calculateMagnitude(double *phasePtr = NULL) {
@@ -651,7 +724,8 @@
     void process(float *inputData, int inputChannelCount,
                  float *outputData, int outputChannelCount,
                  int numFrames) override {
-        float sample;
+        mProcessCount++;
+
         float peak = measurePeakAmplitude(inputData, inputChannelCount, numFrames);
         if (peak > mPeakAmplitude) {
             mPeakAmplitude = peak;
@@ -663,6 +737,7 @@
             float sinOut = sinf(mPhase);
 
             switch (mState) {
+                case STATE_IDLE:
                 case STATE_IMMUNE:
                 case STATE_WAITING_FOR_SIGNAL:
                     break;
@@ -671,7 +746,7 @@
                     mCosAccumulator += sample * cosf(mPhase);
                     mFramesAccumulated++;
                     // Must be a multiple of the period or the calculation will not be accurate.
-                    if (mFramesAccumulated == mPeriod * 4) {
+                    if (mFramesAccumulated == mSinePeriod * PERIODS_NEEDED_FOR_LOCK) {
                         mPhaseOffset = 0.0;
                         mMagnitude = calculateMagnitude(&mPhaseOffset);
                         if (mMagnitude > mThreshold) {
@@ -697,7 +772,22 @@
                         //       mFrameCounter, mGlitchCount, predicted, sample);
                         mState = STATE_IMMUNE;
                         //printf("%5d: switch to STATE_IMMUNE\n", mFrameCounter);
-                        mDownCounter = mPeriod;  // Set duration of IMMUNE state.
+                        mDownCounter = mSinePeriod;  // Set duration of IMMUNE state.
+                    }
+
+                    // Track incoming signal and slowly adjust magnitude to account
+                    // for drift in the DRC or AGC.
+                    mSinAccumulator += sample * sinOut;
+                    mCosAccumulator += sample * cosf(mPhase);
+                    mFramesAccumulated++;
+                    // Must be a multiple of the period or the calculation will not be accurate.
+                    if (mFramesAccumulated == mSinePeriod) {
+                        const double coefficient = 0.1;
+                        double phaseOffset = 0.0;
+                        double magnitude = calculateMagnitude(&phaseOffset);
+                        // One pole averaging filter.
+                        mMagnitude = (mMagnitude * (1.0 - coefficient)) + (magnitude * coefficient);
+                        resetAccumulator();
                     }
                 } break;
             }
@@ -718,6 +808,9 @@
 
         // Do these once per buffer.
         switch (mState) {
+            case STATE_IDLE:
+                mState = STATE_IMMUNE; // so we can tell when
+                break;
             case STATE_IMMUNE:
                 mDownCounter -= numFrames;
                 if (mDownCounter <= 0) {
@@ -748,21 +841,29 @@
     void reset() override {
         mGlitchCount = 0;
         mState = STATE_IMMUNE;
-        mPhaseIncrement = 2.0 * M_PI / mPeriod;
-        printf("phaseInc = %f for period %d\n", mPhaseIncrement, mPeriod);
+        mDownCounter = IMMUNE_FRAME_COUNT;
+        mPhaseIncrement = 2.0 * M_PI / mSinePeriod;
+        printf("phaseInc = %f for period %d\n", mPhaseIncrement, mSinePeriod);
         resetAccumulator();
+        mProcessCount = 0;
     }
 
 private:
 
     enum sine_state_t {
+        STATE_IDLE,
         STATE_IMMUNE,
         STATE_WAITING_FOR_SIGNAL,
         STATE_WAITING_FOR_LOCK,
         STATE_LOCKED
     };
 
-    int     mPeriod = 79;
+    enum constants {
+        IMMUNE_FRAME_COUNT = 48 * 500,
+        PERIODS_NEEDED_FOR_LOCK = 8
+    };
+
+    int     mSinePeriod = 79;
     double  mPhaseIncrement = 0.0;
     double  mPhase = 0.0;
     double  mPhaseOffset = 0.0;
@@ -771,20 +872,19 @@
     double  mThreshold = 0.005;
     double  mTolerance = 0.01;
     int32_t mFramesAccumulated = 0;
+    int32_t mProcessCount = 0;
     double  mSinAccumulator = 0.0;
     double  mCosAccumulator = 0.0;
     int32_t mGlitchCount = 0;
     double  mPeakAmplitude = 0.0;
-    int     mDownCounter = 4000;
+    int     mDownCounter = IMMUNE_FRAME_COUNT;
     int32_t mFrameCounter = 0;
     float   mOutputAmplitude = 0.75;
 
-    int32_t mZeroCrossings = 0;
-
     PseudoRandom  mWhiteNoise;
     float   mNoiseAmplitude = 0.00; // Used to experiment with warbling caused by DRC.
 
-    sine_state_t  mState = STATE_IMMUNE;
+    sine_state_t  mState = STATE_IDLE;
 };
 
 

diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index df0df04..91ebf73 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp

@@ -19,9 +19,12 @@
 #include <algorithm>
 #include <assert.h>
 #include <cctype>
+#include <errno.h>
 #include <math.h>
 #include <stdio.h>
 #include <stdlib.h>
+#include <stdlib.h>
+#include <string.h>
 #include <unistd.h>
 
 #include <aaudio/AAudio.h>
@@ -34,23 +37,32 @@
 
 // Tag for machine readable results as property = value pairs
 #define RESULT_TAG              "RESULT: "
-#define SAMPLE_RATE             48000
 #define NUM_SECONDS             5
+#define PERIOD_MILLIS           1000
 #define NUM_INPUT_CHANNELS      1
-#define FILENAME                "/data/oboe_input.raw"
-#define APP_VERSION             "0.1.22"
+#define FILENAME_ALL            "/data/loopback_all.wav"
+#define FILENAME_ECHOS          "/data/loopback_echos.wav"
+#define APP_VERSION             "0.2.04"
+
+constexpr int kNumCallbacksToDrain   = 20;
+constexpr int kNumCallbacksToDiscard = 20;
 
 struct LoopbackData {
     AAudioStream      *inputStream = nullptr;
     int32_t            inputFramesMaximum = 0;
-    int16_t           *inputData = nullptr;
-    int16_t            peakShort = 0;
-    float             *conversionBuffer = nullptr;
+    int16_t           *inputShortData = nullptr;
+    float             *inputFloatData = nullptr;
+    aaudio_format_t    actualInputFormat = AAUDIO_FORMAT_INVALID;
     int32_t            actualInputChannelCount = 0;
     int32_t            actualOutputChannelCount = 0;
-    int32_t            inputBuffersToDiscard = 10;
+    int32_t            numCallbacksToDrain = kNumCallbacksToDrain;
+    int32_t            numCallbacksToDiscard = kNumCallbacksToDiscard;
     int32_t            minNumFrames = INT32_MAX;
     int32_t            maxNumFrames = 0;
+    int32_t            insufficientReadCount = 0;
+    int32_t            insufficientReadFrames = 0;
+    int32_t            framesReadTotal = 0;
+    int32_t            framesWrittenTotal = 0;
     bool               isDone = false;
 
     aaudio_result_t    inputError = AAUDIO_OK;
@@ -58,14 +70,14 @@
 
     SineAnalyzer       sineAnalyzer;
     EchoAnalyzer       echoAnalyzer;
-    AudioRecording     audioRecorder;
+    AudioRecording     audioRecording;
     LoopbackProcessor *loopbackProcessor;
 };
 
 static void convertPcm16ToFloat(const int16_t *source,
                                 float *destination,
                                 int32_t numSamples) {
-    const float scaler = 1.0f / 32768.0f;
+    constexpr float scaler = 1.0f / 32768.0f;
     for (int i = 0; i < numSamples; i++) {
         destination[i] = source[i] * scaler;
     }
@@ -75,6 +87,31 @@
 // ========================= CALLBACK =================================================
 // ====================================================================================
 // Callback function that fills the audio output buffer.
+
+static int32_t readFormattedData(LoopbackData *myData, int32_t numFrames) {
+    int32_t framesRead = AAUDIO_ERROR_INVALID_FORMAT;
+    if (myData->actualInputFormat == AAUDIO_FORMAT_PCM_I16) {
+        framesRead = AAudioStream_read(myData->inputStream, myData->inputShortData,
+                                       numFrames,
+                                       0 /* timeoutNanoseconds */);
+    } else if (myData->actualInputFormat == AAUDIO_FORMAT_PCM_FLOAT) {
+        framesRead = AAudioStream_read(myData->inputStream, myData->inputFloatData,
+                                       numFrames,
+                                       0 /* timeoutNanoseconds */);
+    } else {
+        printf("ERROR actualInputFormat = %d\n", myData->actualInputFormat);
+        assert(false);
+    }
+    if (framesRead < 0) {
+        myData->inputError = framesRead;
+        printf("ERROR in read = %d = %s\n", framesRead,
+               AAudio_convertResultToText(framesRead));
+    } else {
+        myData->framesReadTotal += framesRead;
+    }
+    return framesRead;
+}
+
 static aaudio_data_callback_result_t MyDataCallbackProc(
         AAudioStream *outputStream,
         void *userData,
@@ -87,7 +124,7 @@
     float  *outputData = (float  *) audioData;
 
     // Read audio data from the input stream.
-    int32_t framesRead;
+    int32_t actualFramesRead;
 
     if (numFrames > myData->inputFramesMaximum) {
         myData->inputError = AAUDIO_ERROR_OUT_OF_RANGE;
@@ -101,46 +138,86 @@
         myData->minNumFrames = numFrames;
     }
 
-    if (myData->inputBuffersToDiscard > 0) {
+    // Silence the output.
+    int32_t numBytes = numFrames * myData->actualOutputChannelCount * sizeof(float);
+    memset(audioData, 0 /* value */, numBytes);
+
+    if (myData->numCallbacksToDrain > 0) {
         // Drain the input.
+        int32_t totalFramesRead = 0;
         do {
-            framesRead = AAudioStream_read(myData->inputStream, myData->inputData,
-                                       numFrames, 0);
-            if (framesRead < 0) {
-                myData->inputError = framesRead;
-                printf("ERROR in read = %d", framesRead);
-                result = AAUDIO_CALLBACK_RESULT_STOP;
-            } else if (framesRead > 0) {
-                myData->inputBuffersToDiscard--;
+            actualFramesRead = readFormattedData(myData, numFrames);
+            if (actualFramesRead) {
+                totalFramesRead += actualFramesRead;
             }
-        } while(framesRead > 0);
-    } else {
-        framesRead = AAudioStream_read(myData->inputStream, myData->inputData,
-                                       numFrames, 0);
-        if (framesRead < 0) {
-            myData->inputError = framesRead;
-            printf("ERROR in read = %d", framesRead);
+            // Ignore errors because input stream may not be started yet.
+        } while (actualFramesRead > 0);
+        // Only counts if we actually got some data.
+        if (totalFramesRead > 0) {
+            myData->numCallbacksToDrain--;
+        }
+
+    } else if (myData->numCallbacksToDiscard > 0) {
+        // Ignore. Allow the input to fill back up to equilibrium with the output.
+        actualFramesRead = readFormattedData(myData, numFrames);
+        if (actualFramesRead < 0) {
             result = AAUDIO_CALLBACK_RESULT_STOP;
-        } else if (framesRead > 0) {
+        }
+        myData->numCallbacksToDiscard--;
 
-            myData->audioRecorder.write(myData->inputData,
-                                        myData->actualInputChannelCount,
-                                        numFrames);
+    } else {
 
-            int32_t numSamples = framesRead * myData->actualInputChannelCount;
-            convertPcm16ToFloat(myData->inputData, myData->conversionBuffer, numSamples);
+        int32_t numInputBytes = numFrames * myData->actualInputChannelCount * sizeof(float);
+        memset(myData->inputFloatData, 0 /* value */, numInputBytes);
 
-            myData->loopbackProcessor->process(myData->conversionBuffer,
-                                              myData->actualInputChannelCount,
-                                              outputData,
-                                              myData->actualOutputChannelCount,
-                                              framesRead);
+        // Process data after equilibrium.
+        int64_t inputFramesWritten = AAudioStream_getFramesWritten(myData->inputStream);
+        int64_t inputFramesRead = AAudioStream_getFramesRead(myData->inputStream);
+        int64_t framesAvailable = inputFramesWritten - inputFramesRead;
+        actualFramesRead = readFormattedData(myData, numFrames);
+        if (actualFramesRead < 0) {
+            result = AAUDIO_CALLBACK_RESULT_STOP;
+        } else {
+
+            if (actualFramesRead < numFrames) {
+                if(actualFramesRead < (int32_t) framesAvailable) {
+                    printf("insufficient but numFrames = %d"
+                                   ", actualFramesRead = %d"
+                                   ", inputFramesWritten = %d"
+                                   ", inputFramesRead = %d"
+                                   ", available = %d\n",
+                           numFrames,
+                           actualFramesRead,
+                           (int) inputFramesWritten,
+                           (int) inputFramesRead,
+                           (int) framesAvailable);
+                }
+                myData->insufficientReadCount++;
+                myData->insufficientReadFrames += numFrames - actualFramesRead; // deficit
+            }
+
+            int32_t numSamples = actualFramesRead * myData->actualInputChannelCount;
+
+            if (myData->actualInputFormat == AAUDIO_FORMAT_PCM_I16) {
+                convertPcm16ToFloat(myData->inputShortData, myData->inputFloatData, numSamples);
+            }
+            // Save for later.
+            myData->audioRecording.write(myData->inputFloatData,
+                                         myData->actualInputChannelCount,
+                                         numFrames);
+            // Analyze the data.
+            myData->loopbackProcessor->process(myData->inputFloatData,
+                                               myData->actualInputChannelCount,
+                                               outputData,
+                                               myData->actualOutputChannelCount,
+                                               numFrames);
             myData->isDone = myData->loopbackProcessor->isDone();
             if (myData->isDone) {
                 result = AAUDIO_CALLBACK_RESULT_STOP;
             }
         }
     }
+    myData->framesWrittenTotal += numFrames;
 
     return result;
 }
@@ -148,29 +225,29 @@
 static void MyErrorCallbackProc(
         AAudioStream *stream __unused,
         void *userData __unused,
-        aaudio_result_t error)
-{
+        aaudio_result_t error) {
     printf("Error Callback, error: %d\n",(int)error);
     LoopbackData *myData = (LoopbackData *) userData;
     myData->outputError = error;
 }
 
 static void usage() {
-    printf("loopback: -n{numBursts} -p{outPerf} -P{inPerf} -t{test} -g{gain} -f{freq}\n");
-    printf("          -c{inputChannels}\n");
-    printf("          -f{freq}  sine frequency\n");
-    printf("          -g{gain}  recirculating loopback gain\n");
-    printf("          -m enable MMAP mode\n");
-    printf("          -n{numBursts} buffer size, for example 2 for double buffered\n");
-    printf("          -p{outPerf}  set output AAUDIO_PERFORMANCE_MODE*\n");
-    printf("          -P{inPerf}   set input AAUDIO_PERFORMANCE_MODE*\n");
-    printf("              n for _NONE\n");
-    printf("              l for _LATENCY\n");
-    printf("              p for _POWER_SAVING;\n");
-    printf("          -t{test}   select test mode\n");
-    printf("              m for sine magnitude\n");
-    printf("              e for echo latency (default)\n");
-    printf("For example:  loopback -b2 -pl -Pn\n");
+    printf("Usage: aaudio_loopback [OPTION]...\n\n");
+    AAudioArgsParser::usage();
+    printf("      -B{frames}        input capacity in frames\n");
+    printf("      -C{channels}      number of input channels\n");
+    printf("      -F{0,1,2}         input format, 1=I16, 2=FLOAT\n");
+    printf("      -g{gain}          recirculating loopback gain\n");
+    printf("      -P{inPerf}        set input AAUDIO_PERFORMANCE_MODE*\n");
+    printf("          n for _NONE\n");
+    printf("          l for _LATENCY\n");
+    printf("          p for _POWER_SAVING\n");
+    printf("      -t{test}          select test mode\n");
+    printf("          m for sine magnitude\n");
+    printf("          e for echo latency (default)\n");
+    printf("          f for file latency, analyzes %s\n\n", FILENAME_ECHOS);
+    printf("      -X  use EXCLUSIVE mode for input\n");
+    printf("Example:  aaudio_loopback -n2 -pl -Pl -x\n");
 }
 
 static aaudio_performance_mode_t parsePerformanceMode(char c) {
@@ -196,6 +273,7 @@
 enum {
     TEST_SINE_MAGNITUDE = 0,
     TEST_ECHO_LATENCY,
+    TEST_FILE_LATENCY,
 };
 
 static int parseTestMode(char c) {
@@ -208,6 +286,9 @@
         case 'e':
             testMode = TEST_ECHO_LATENCY;
             break;
+        case 'f':
+            testMode = TEST_FILE_LATENCY;
+            break;
         default:
             printf("ERROR in value test mode %c\n", c);
             break;
@@ -231,9 +312,10 @@
         }
     }
     float gain = 0.98f / maxSample;
+
     for (int32_t i = start; i < end; i++) {
         float sample = data[i];
-        printf("%5.3f ", sample); // actual value
+        printf("%6d: %7.4f ", i, sample); // actual value
         sample *= gain;
         printAudioScope(sample);
     }
@@ -245,33 +327,28 @@
 int main(int argc, const char **argv)
 {
 
-    AAudioArgsParser     argParser;
-    AAudioSimplePlayer   player;
-    AAudioSimpleRecorder recorder;
-    LoopbackData         loopbackData;
-    AAudioStream        *outputStream = nullptr;
+    AAudioArgsParser      argParser;
+    AAudioSimplePlayer    player;
+    AAudioSimpleRecorder  recorder;
+    LoopbackData          loopbackData;
+    AAudioStream         *inputStream                = nullptr;
+    AAudioStream         *outputStream               = nullptr;
 
-    aaudio_result_t      result = AAUDIO_OK;
-    aaudio_sharing_mode_t requestedInputSharingMode     = AAUDIO_SHARING_MODE_SHARED;
+    aaudio_result_t       result = AAUDIO_OK;
+    aaudio_sharing_mode_t requestedInputSharingMode  = AAUDIO_SHARING_MODE_SHARED;
     int                   requestedInputChannelCount = NUM_INPUT_CHANNELS;
-    const int             requestedOutputChannelCount = AAUDIO_UNSPECIFIED;
-    int                   actualSampleRate = 0;
-    const aaudio_format_t requestedInputFormat = AAUDIO_FORMAT_PCM_I16;
-    const aaudio_format_t requestedOutputFormat = AAUDIO_FORMAT_PCM_FLOAT;
-    aaudio_format_t       actualInputFormat;
-    aaudio_format_t       actualOutputFormat;
-    aaudio_performance_mode_t outputPerformanceLevel = AAUDIO_PERFORMANCE_MODE_LOW_LATENCY;
-    aaudio_performance_mode_t inputPerformanceLevel = AAUDIO_PERFORMANCE_MODE_LOW_LATENCY;
+    aaudio_format_t       requestedInputFormat       = AAUDIO_FORMAT_UNSPECIFIED;
+    int32_t               requestedInputCapacity     = -1;
+    aaudio_performance_mode_t inputPerformanceLevel  = AAUDIO_PERFORMANCE_MODE_LOW_LATENCY;
 
-    int testMode = TEST_ECHO_LATENCY;
-    double gain = 1.0;
+    int32_t               outputFramesPerBurst = 0;
 
-    aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNINITIALIZED;
-    int32_t framesPerBurst = 0;
-    float *outputData = NULL;
-    double deviation;
-    double latency;
-    int32_t burstsPerBuffer = 1; // single buffered
+    aaudio_format_t       actualOutputFormat         = AAUDIO_FORMAT_INVALID;
+    int32_t               actualSampleRate           = 0;
+    int                   written                    = 0;
+
+    int                   testMode                   = TEST_ECHO_LATENCY;
+    double                gain                       = 1.0;
 
     // Make printf print immediately so that debug info is not stuck
     // in a buffer if we hang or crash.
@@ -286,9 +363,15 @@
             if (arg[0] == '-') {
                 char option = arg[1];
                 switch (option) {
+                    case 'B':
+                        requestedInputCapacity = atoi(&arg[2]);
+                        break;
                     case 'C':
                         requestedInputChannelCount = atoi(&arg[2]);
                         break;
+                    case 'F':
+                        requestedInputFormat = atoi(&arg[2]);
+                        break;
                     case 'g':
                         gain = atof(&arg[2]);
                         break;
@@ -321,8 +404,9 @@
     }
 
     int32_t requestedDuration = argParser.getDurationSeconds();
-    int32_t recordingDuration = std::min(60, requestedDuration);
-    loopbackData.audioRecorder.allocate(recordingDuration * SAMPLE_RATE);
+    int32_t requestedDurationMillis = requestedDuration * MILLIS_PER_SECOND;
+    int32_t timeMillis = 0;
+    int32_t recordingDuration = std::min(60 * 5, requestedDuration);
 
     switch(testMode) {
         case TEST_SINE_MAGNITUDE:
@@ -332,61 +416,112 @@
             loopbackData.echoAnalyzer.setGain(gain);
             loopbackData.loopbackProcessor = &loopbackData.echoAnalyzer;
             break;
+        case TEST_FILE_LATENCY: {
+            loopbackData.echoAnalyzer.setGain(gain);
+
+            loopbackData.loopbackProcessor = &loopbackData.echoAnalyzer;
+            int read = loopbackData.loopbackProcessor->load(FILENAME_ECHOS);
+            printf("main() read %d mono samples from %s on Android device\n", read, FILENAME_ECHOS);
+            loopbackData.loopbackProcessor->report();
+            return 0;
+        }
+            break;
         default:
             exit(1);
             break;
     }
 
     printf("OUTPUT stream ----------------------------------------\n");
-    argParser.setFormat(requestedOutputFormat);
     result = player.open(argParser, MyDataCallbackProc, MyErrorCallbackProc, &loopbackData);
     if (result != AAUDIO_OK) {
         fprintf(stderr, "ERROR -  player.open() returned %d\n", result);
-        goto finish;
+        exit(1);
     }
     outputStream = player.getStream();
-    argParser.compareWithStream(outputStream);
 
     actualOutputFormat = AAudioStream_getFormat(outputStream);
-    assert(actualOutputFormat == AAUDIO_FORMAT_PCM_FLOAT);
+    if (actualOutputFormat != AAUDIO_FORMAT_PCM_FLOAT) {
+        fprintf(stderr, "ERROR - only AAUDIO_FORMAT_PCM_FLOAT supported\n");
+        exit(1);
+    }
 
-    printf("INPUT stream ----------------------------------------\n");
+    actualSampleRate = AAudioStream_getSampleRate(outputStream);
+    loopbackData.audioRecording.allocate(recordingDuration * actualSampleRate);
+    loopbackData.audioRecording.setSampleRate(actualSampleRate);
+    outputFramesPerBurst = AAudioStream_getFramesPerBurst(outputStream);
+
+    argParser.compareWithStream(outputStream);
+
+    printf("INPUT  stream ----------------------------------------\n");
     // Use different parameters for the input.
     argParser.setNumberOfBursts(AAUDIO_UNSPECIFIED);
     argParser.setFormat(requestedInputFormat);
     argParser.setPerformanceMode(inputPerformanceLevel);
     argParser.setChannelCount(requestedInputChannelCount);
     argParser.setSharingMode(requestedInputSharingMode);
+
+    // Make sure the input buffer has plenty of capacity.
+    // Extra capacity on input should not increase latency if we keep it drained.
+    int32_t inputBufferCapacity = requestedInputCapacity;
+    if (inputBufferCapacity < 0) {
+        int32_t outputBufferCapacity = AAudioStream_getBufferCapacityInFrames(outputStream);
+        inputBufferCapacity = 2 * outputBufferCapacity;
+    }
+    argParser.setBufferCapacity(inputBufferCapacity);
+
     result = recorder.open(argParser);
     if (result != AAUDIO_OK) {
         fprintf(stderr, "ERROR -  recorder.open() returned %d\n", result);
         goto finish;
     }
-    loopbackData.inputStream = recorder.getStream();
-    argParser.compareWithStream(loopbackData.inputStream);
+    inputStream = loopbackData.inputStream = recorder.getStream();
 
-    // This is the number of frames that are read in one chunk by a DMA controller
-    // or a DSP or a mixer.
-    framesPerBurst = AAudioStream_getFramesPerBurst(outputStream);
+    {
+        int32_t actualCapacity = AAudioStream_getBufferCapacityInFrames(inputStream);
+        result = AAudioStream_setBufferSizeInFrames(inputStream, actualCapacity);
+        if (result < 0) {
+            fprintf(stderr, "ERROR -  AAudioStream_setBufferSizeInFrames() returned %d\n", result);
+            goto finish;
+        } else {}
+    }
 
-    actualInputFormat = AAudioStream_getFormat(outputStream);
-    assert(actualInputFormat == AAUDIO_FORMAT_PCM_I16);
+    argParser.compareWithStream(inputStream);
 
+    // If the input stream is too small then we cannot satisfy the output callback.
+    {
+        int32_t actualCapacity = AAudioStream_getBufferCapacityInFrames(inputStream);
+        if (actualCapacity < 2 * outputFramesPerBurst) {
+            fprintf(stderr, "ERROR - input capacity < 2 * outputFramesPerBurst\n");
+            goto finish;
+        }
+    }
+
+    // ------- Setup loopbackData -----------------------------
+    loopbackData.actualInputFormat = AAudioStream_getFormat(inputStream);
 
     loopbackData.actualInputChannelCount = recorder.getChannelCount();
     loopbackData.actualOutputChannelCount = player.getChannelCount();
 
     // Allocate a buffer for the audio data.
-    loopbackData.inputFramesMaximum = 32 * framesPerBurst;
-    loopbackData.inputBuffersToDiscard = 100;
+    loopbackData.inputFramesMaximum = 32 * AAudioStream_getFramesPerBurst(inputStream);
 
-    loopbackData.inputData = new int16_t[loopbackData.inputFramesMaximum
-                                         * loopbackData.actualInputChannelCount];
-    loopbackData.conversionBuffer = new float[loopbackData.inputFramesMaximum *
-                                              loopbackData.actualInputChannelCount];
+    if (loopbackData.actualInputFormat == AAUDIO_FORMAT_PCM_I16) {
+        loopbackData.inputShortData = new int16_t[loopbackData.inputFramesMaximum
+                                                  * loopbackData.actualInputChannelCount]{};
+    }
+    loopbackData.inputFloatData = new float[loopbackData.inputFramesMaximum *
+                                              loopbackData.actualInputChannelCount]{};
 
     loopbackData.loopbackProcessor->reset();
 
+    // Start OUTPUT first so INPUT does not overflow.
+    result = player.start();
+    if (result != AAUDIO_OK) {
+        printf("ERROR - AAudioStream_requestStart(output) returned %d = %s\n",
+               result, AAudio_convertResultToText(result));
+        goto finish;
+    }
+
     result = recorder.start();
     if (result != AAUDIO_OK) {
         printf("ERROR - AAudioStream_requestStart(input) returned %d = %s\n",
@@ -394,16 +529,8 @@
         goto finish;
     }
 
-    result = player.start();
-    if (result != AAUDIO_OK) {
-        printf("ERROR - AAudioStream_requestStart(output) returned %d = %s\n",
-               result, AAudio_convertResultToText(result));
-        goto finish;
-    }
-
-    printf("------- sleep while the callback runs --------------\n");
-    fflush(stdout);
-    for (int i = requestedDuration; i > 0 ; i--) {
+    printf("------- sleep and log while the callback runs --------------\n");
+    while (timeMillis <= requestedDurationMillis) {
         if (loopbackData.inputError != AAUDIO_OK) {
             printf("  ERROR on input stream\n");
             break;
@@ -411,60 +538,128 @@
                 printf("  ERROR on output stream\n");
                 break;
         } else if (loopbackData.isDone) {
-                printf("  test says it is done!\n");
+                printf("  Test says it is DONE!\n");
                 break;
         } else {
-            sleep(1);
-            printf("%4d: ", i);
+            // Log a line of stream data.
+            printf("%7.3f: ", 0.001 * timeMillis); // display in seconds
             loopbackData.loopbackProcessor->printStatus();
+            printf(" insf %3d,", (int) loopbackData.insufficientReadCount);
 
-            int64_t inputFramesWritten = AAudioStream_getFramesWritten(loopbackData.inputStream);
-            int64_t inputFramesRead = AAudioStream_getFramesRead(loopbackData.inputStream);
+            int64_t inputFramesWritten = AAudioStream_getFramesWritten(inputStream);
+            int64_t inputFramesRead = AAudioStream_getFramesRead(inputStream);
             int64_t outputFramesWritten = AAudioStream_getFramesWritten(outputStream);
             int64_t outputFramesRead = AAudioStream_getFramesRead(outputStream);
-            printf(" INPUT: wr %lld rd %lld state %s, OUTPUT: wr %lld rd %lld state %s, xruns %d\n",
+            static const int textOffset = strlen("AAUDIO_STREAM_STATE_"); // strip this off
+            printf(" | INPUT: wr %7lld - rd %7lld = %5lld, st %8s, oruns %3d",
                    (long long) inputFramesWritten,
                    (long long) inputFramesRead,
-                   AAudio_convertStreamStateToText(AAudioStream_getState(loopbackData.inputStream)),
+                   (long long) (inputFramesWritten - inputFramesRead),
+                   &AAudio_convertStreamStateToText(
+                           AAudioStream_getState(inputStream))[textOffset],
+                   AAudioStream_getXRunCount(inputStream));
+
+            printf(" | OUTPUT: wr %7lld - rd %7lld = %5lld, st %8s, uruns %3d\n",
                    (long long) outputFramesWritten,
                    (long long) outputFramesRead,
-                   AAudio_convertStreamStateToText(AAudioStream_getState(outputStream)),
+                    (long long) (outputFramesWritten - outputFramesRead),
+                   &AAudio_convertStreamStateToText(
+                           AAudioStream_getState(outputStream))[textOffset],
                    AAudioStream_getXRunCount(outputStream)
             );
         }
+        int32_t periodMillis = (timeMillis < 2000) ? PERIOD_MILLIS / 4 : PERIOD_MILLIS;
+        usleep(periodMillis * 1000);
+        timeMillis += periodMillis;
+    }
+
+    result = player.stop();
+    if (result != AAUDIO_OK) {
+        printf("ERROR - player.stop() returned %d = %s\n",
+               result, AAudio_convertResultToText(result));
+        goto finish;
+    }
+
+    result = recorder.stop();
+    if (result != AAUDIO_OK) {
+        printf("ERROR - recorder.stop() returned %d = %s\n",
+               result, AAudio_convertResultToText(result));
+        goto finish;
     }
 
     printf("input error = %d = %s\n",
-                loopbackData.inputError, AAudio_convertResultToText(loopbackData.inputError));
-
-    printf("AAudioStream_getXRunCount %d\n", AAudioStream_getXRunCount(outputStream));
-    printf("framesRead    = %8d\n", (int) AAudioStream_getFramesRead(outputStream));
-    printf("framesWritten = %8d\n", (int) AAudioStream_getFramesWritten(outputStream));
-    printf("min numFrames = %8d\n", (int) loopbackData.minNumFrames);
-    printf("max numFrames = %8d\n", (int) loopbackData.maxNumFrames);
+           loopbackData.inputError, AAudio_convertResultToText(loopbackData.inputError));
 
     if (loopbackData.inputError == AAUDIO_OK) {
         if (testMode == TEST_SINE_MAGNITUDE) {
-            printAudioGraph(loopbackData.audioRecorder, 200);
+            printAudioGraph(loopbackData.audioRecording, 200);
         }
+        // Print again so we don't have to scroll past waveform.
+        printf("OUTPUT Stream ----------------------------------------\n");
+        argParser.compareWithStream(outputStream);
+        printf("INPUT  Stream ----------------------------------------\n");
+        argParser.compareWithStream(inputStream);
+
         loopbackData.loopbackProcessor->report();
     }
 
     {
-        int written = loopbackData.audioRecorder.save(FILENAME);
-        printf("main() wrote %d mono samples to %s on Android device\n", written, FILENAME);
+        int32_t framesRead = AAudioStream_getFramesRead(inputStream);
+        int32_t framesWritten = AAudioStream_getFramesWritten(inputStream);
+        printf("Callback Results ---------------------------------------- INPUT\n");
+        printf("  input overruns   = %d\n", AAudioStream_getXRunCount(inputStream));
+        printf("  framesWritten    = %8d\n", framesWritten);
+        printf("  framesRead       = %8d\n", framesRead);
+        printf("  myFramesRead     = %8d\n", (int) loopbackData.framesReadTotal);
+        printf("  written - read   = %8d\n", (int) (framesWritten - framesRead));
+        printf("  insufficient #   = %8d\n", (int) loopbackData.insufficientReadCount);
+        if (loopbackData.insufficientReadCount > 0) {
+            printf("  insufficient frames = %8d\n", (int) loopbackData.insufficientReadFrames);
+        }
+    }
+    {
+        int32_t framesRead = AAudioStream_getFramesRead(outputStream);
+        int32_t framesWritten = AAudioStream_getFramesWritten(outputStream);
+        printf("Callback Results ---------------------------------------- OUTPUT\n");
+        printf("  output underruns = %d\n", AAudioStream_getXRunCount(outputStream));
+        printf("  myFramesWritten  = %8d\n", (int) loopbackData.framesWrittenTotal);
+        printf("  framesWritten    = %8d\n", framesWritten);
+        printf("  framesRead       = %8d\n", framesRead);
+        printf("  min numFrames    = %8d\n", (int) loopbackData.minNumFrames);
+        printf("  max numFrames    = %8d\n", (int) loopbackData.maxNumFrames);
+    }
+
+    written = loopbackData.loopbackProcessor->save(FILENAME_ECHOS);
+    if (written > 0) {
+        printf("main() wrote %8d mono samples to \"%s\" on Android device\n",
+               written, FILENAME_ECHOS);
+    }
+
+    written = loopbackData.audioRecording.save(FILENAME_ALL);
+    if (written > 0) {
+        printf("main() wrote %8d mono samples to \"%s\" on Android device\n",
+               written, FILENAME_ALL);
+    }
+
+    if (loopbackData.loopbackProcessor->getResult() < 0) {
+        printf("ERROR: LOOPBACK PROCESSING FAILED. Maybe because the volume was too low.\n");
+        result = loopbackData.loopbackProcessor->getResult();
+    }
+    if (loopbackData.insufficientReadCount > 3) {
+        printf("ERROR: LOOPBACK PROCESSING FAILED. insufficientReadCount too high\n");
+        result = AAUDIO_ERROR_UNAVAILABLE;
     }
 
 finish:
     player.close();
     recorder.close();
-    delete[] loopbackData.conversionBuffer;
-    delete[] loopbackData.inputData;
-    delete[] outputData;
+    delete[] loopbackData.inputFloatData;
+    delete[] loopbackData.inputShortData;
 
-    printf(RESULT_TAG "error = %d = %s\n", result, AAudio_convertResultToText(result));
-    if ((result != AAUDIO_OK)) {
-        printf("error %d = %s\n", result, AAudio_convertResultToText(result));
+    printf(RESULT_TAG "result = %d \n", result); // machine readable
+    printf("result is %s\n", AAudio_convertResultToText(result)); // human readable
+    if (result != AAUDIO_OK) {
+        printf("FAILURE\n");
         return EXIT_FAILURE;
     } else {
         printf("SUCCESS\n");

diff --git a/media/libaaudio/examples/loopback/src/loopback.sh b/media/libaaudio/examples/loopback/src/loopback.sh
index bc63125..a5712b8 100644
--- a/media/libaaudio/examples/loopback/src/loopback.sh
+++ b/media/libaaudio/examples/loopback/src/loopback.sh

@@ -1,10 +1,30 @@
 #!/system/bin/sh
 # Run a loopback test in the background after a delay.
-# To run the script enter:
+# To run the script, enter these commands once:
+#    adb disable-verity
+#    adb reboot
+#    adb remount
+#    adb sync
+#    adb push loopback.sh /data/
+# For each test run:
 #    adb shell "nohup sh /data/loopback.sh &"
+# Quickly connect USB audio if needed, either manually or via Tigertail switch.
+# Wait until the test completes, restore USB to host if needed, and then:
+#    adb pull /data/loopreport.txt
+#    adb pull /data/loopback_all.wav
+#    adb pull /data/loopback_echos.wav
 
 SLEEP_TIME=10
-TEST_COMMAND="aaudio_loopback -pl -Pl -C1 -n2 -m2 -tm -d5"
+TEST_COMMAND="/data/nativetest/aaudio_loopback/aaudio_loopback -pl -Pl -C1 -n2 -m2 -te -d5"
+# Partial list of options:
+#   -pl (output) performance mode: low latency
+#   -Pl input performance mode: low latency
+#   -C1 input channel count: 1
+#   -n2 number of bursts: 2
+#   -m2 mmap policy: 2
+#   -t? test mode: -tm for sine magnitude, -te for echo latency, -tf for file latency
+#   -d5 device ID
+# For full list of available options, see AAudioArgsParser.h and loopback.cpp
 
 echo "Plug in USB Mir and Fun Plug."
 echo "Test will start in ${SLEEP_TIME} seconds: ${TEST_COMMAND}"

diff --git a/media/libaaudio/examples/utils/AAudioArgsParser.h b/media/libaaudio/examples/utils/AAudioArgsParser.h
index ada37e2..88d7401 100644
--- a/media/libaaudio/examples/utils/AAudioArgsParser.h
+++ b/media/libaaudio/examples/utils/AAudioArgsParser.h

@@ -17,7 +17,10 @@
 #ifndef AAUDIO_EXAMPLE_ARGS_PARSER_H
 #define AAUDIO_EXAMPLE_ARGS_PARSER_H
 
-#include <cctype>
+#define MAX_CHANNELS                     8
+
+//#include <cctype>
+#include <dlfcn.h>
 #include <unistd.h>
 #include <stdio.h>
 #include <stdlib.h>
@@ -27,7 +30,63 @@
 
 #include "AAudioExampleUtils.h"
 
-// TODO use this as a base class within AAudio
+
+static void (*s_setUsage)(AAudioStreamBuilder* builder, aaudio_usage_t usage) = nullptr;
+static void (*s_setContentType)(AAudioStreamBuilder* builder,
+                                aaudio_content_type_t contentType) = nullptr;
+static void (*s_setInputPreset)(AAudioStreamBuilder* builder,
+                                aaudio_input_preset_t inputPreset) = nullptr;
+
+static bool s_loadAttempted = false;
+static aaudio_usage_t (*s_getUsage)(AAudioStream *stream) = nullptr;
+static aaudio_content_type_t (*s_getContentType)(AAudioStream *stream) = nullptr;
+static aaudio_input_preset_t (*s_getInputPreset)(AAudioStream *stream) = nullptr;
+
+// Link to test functions in shared library.
+static void loadFutureFunctions() {
+    if (s_loadAttempted)  return; // only try once
+    s_loadAttempted = true;
+
+    void *handle = dlopen("libaaudio.so", RTLD_NOW);
+    if (handle != nullptr) {
+        s_setUsage = (void (*)(AAudioStreamBuilder *, aaudio_usage_t))
+                dlsym(handle, "AAudioStreamBuilder_setUsage");
+        if (s_setUsage == nullptr) goto error;
+
+        s_setContentType = (void (*)(AAudioStreamBuilder *, aaudio_content_type_t))
+                dlsym(handle, "AAudioStreamBuilder_setContentType");
+        if (s_setContentType == nullptr) goto error;
+
+        s_setInputPreset = (void (*)(AAudioStreamBuilder *, aaudio_input_preset_t))
+                dlsym(handle, "AAudioStreamBuilder_setInputPreset");
+        if (s_setInputPreset == nullptr) goto error;
+
+        s_getUsage = (aaudio_usage_t (*)(AAudioStream *))
+                dlsym(handle, "AAudioStream_getUsage");
+        if (s_getUsage == nullptr) goto error;
+
+        s_getContentType = (aaudio_content_type_t (*)(AAudioStream *))
+                dlsym(handle, "AAudioStream_getContentType");
+        if (s_getContentType == nullptr) goto error;
+
+        s_getInputPreset = (aaudio_input_preset_t (*)(AAudioStream *))
+                dlsym(handle, "AAudioStream_getInputPreset");
+        if (s_getInputPreset == nullptr) goto error;
+    }
+    return;
+
+error:
+    // prevent any calls to these functions
+    s_setUsage = nullptr;
+    s_setContentType = nullptr;
+    s_setInputPreset = nullptr;
+    s_getUsage = nullptr;
+    s_getContentType = nullptr;
+    s_getInputPreset = nullptr;
+    dlclose(handle);
+    return;
+}
+
 class AAudioParameters {
 public:
 
@@ -39,6 +98,10 @@
     }
 
     void setChannelCount(int32_t channelCount) {
+        if (channelCount > MAX_CHANNELS) {
+            printf("Sorry, MAX of %d channels!\n", MAX_CHANNELS);
+            channelCount = MAX_CHANNELS;
+        }
         mChannelCount = channelCount;
     }
 
@@ -82,6 +145,30 @@
         mPerformanceMode = performanceMode;
     }
 
+    aaudio_usage_t getUsage() const {
+        return mUsage;
+    }
+
+    void setUsage(aaudio_usage_t usage) {
+        mUsage = usage;
+    }
+
+    aaudio_content_type_t getContentType() const {
+        return mContentType;
+    }
+
+    void setContentType(aaudio_content_type_t contentType) {
+        mContentType = contentType;
+    }
+
+    aaudio_input_preset_t getInputPreset() const {
+        return mInputPreset;
+    }
+
+    void setInputPreset(aaudio_input_preset_t inputPreset) {
+        mInputPreset = inputPreset;
+    }
+
     int32_t getDeviceId() const {
         return mDeviceId;
     }
@@ -110,6 +197,24 @@
         AAudioStreamBuilder_setDeviceId(builder, mDeviceId);
         AAudioStreamBuilder_setSharingMode(builder, mSharingMode);
         AAudioStreamBuilder_setPerformanceMode(builder, mPerformanceMode);
+
+        // Call P functions if supported.
+        loadFutureFunctions();
+        if (s_setUsage != nullptr) {
+            s_setUsage(builder, mUsage);
+        } else if (mUsage != AAUDIO_UNSPECIFIED){
+            printf("WARNING: setUsage not supported");
+        }
+        if (s_setContentType != nullptr) {
+            s_setContentType(builder, mContentType);
+        } else if (mUsage != AAUDIO_UNSPECIFIED){
+            printf("WARNING: setContentType not supported");
+        }
+        if (s_setInputPreset != nullptr) {
+            s_setInputPreset(builder, mInputPreset);
+        } else if (mUsage != AAUDIO_UNSPECIFIED){
+            printf("WARNING: setInputPreset not supported");
+        }
     }
 
 private:
@@ -122,6 +227,10 @@
     aaudio_sharing_mode_t      mSharingMode     = AAUDIO_SHARING_MODE_SHARED;
     aaudio_performance_mode_t  mPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
 
+    aaudio_usage_t             mUsage           = AAUDIO_UNSPECIFIED;
+    aaudio_content_type_t      mContentType     = AAUDIO_UNSPECIFIED;
+    aaudio_input_preset_t      mInputPreset     = AAUDIO_UNSPECIFIED;
+
     int32_t                    mNumberOfBursts  = AAUDIO_UNSPECIFIED;
 };
 
@@ -152,8 +261,11 @@
                 case 'd':
                     setDeviceId(atoi(&arg[2]));
                     break;
-                case 's':
-                    mDurationSeconds = atoi(&arg[2]);
+                case 'f':
+                    setFormat(atoi(&arg[2]));
+                    break;
+                case 'i':
+                    setInputPreset(atoi(&arg[2]));
                     break;
                 case 'm': {
                     aaudio_policy_t policy = AAUDIO_POLICY_AUTO;
@@ -171,9 +283,18 @@
                 case 'r':
                     setSampleRate(atoi(&arg[2]));
                     break;
+                case 's':
+                    mDurationSeconds = atoi(&arg[2]);
+                    break;
+                case 'u':
+                    setUsage(atoi(&arg[2]));
+                    break;
                 case 'x':
                     setSharingMode(AAUDIO_SHARING_MODE_EXCLUSIVE);
                     break;
+                case 'y':
+                    setContentType(atoi(&arg[2]));
+                    break;
                 default:
                     unrecognized = true;
                     break;
@@ -201,24 +322,32 @@
     }
 
     static void usage() {
-        printf("-c{channels} -d{duration} -m -n{burstsPerBuffer} -p{perfMode} -r{rate} -x\n");
+        printf("-c{channels} -d{deviceId} -m{mmapPolicy} -n{burstsPerBuffer} -p{perfMode}");
+        printf(" -r{rate} -s{seconds} -x\n");
         printf("      Default values are UNSPECIFIED unless otherwise stated.\n");
         printf("      -b{bufferCapacity} frames\n");
         printf("      -c{channels} for example 2 for stereo\n");
         printf("      -d{deviceId} default is %d\n", AAUDIO_UNSPECIFIED);
-        printf("      -s{duration} in seconds, default is %d\n", DEFAULT_DURATION_SECONDS);
+        printf("      -f{0|1|2} set format\n");
+        printf("          0 = UNSPECIFIED\n");
+        printf("          1 = PCM_I16\n");
+        printf("          2 = FLOAT\n");
+        printf("      -i{inputPreset} eg. 5 for AAUDIO_INPUT_PRESET_CAMCORDER\n");
         printf("      -m{0|1|2|3} set MMAP policy\n");
-        printf("          0 = _UNSPECIFIED, default\n");
-        printf("          1 = _NEVER\n");
-        printf("          2 = _AUTO, also if -m is used with no number\n");
-        printf("          3 = _ALWAYS\n");
+        printf("          0 = _UNSPECIFIED, use aaudio.mmap_policy system property, default\n");
+        printf("          1 = _NEVER, never use MMAP\n");
+        printf("          2 = _AUTO, use MMAP if available, default for -m with no number\n");
+        printf("          3 = _ALWAYS, use MMAP or fail\n");
         printf("      -n{numberOfBursts} for setBufferSize\n");
         printf("      -p{performanceMode} set output AAUDIO_PERFORMANCE_MODE*, default NONE\n");
         printf("          n for _NONE\n");
         printf("          l for _LATENCY\n");
         printf("          p for _POWER_SAVING;\n");
         printf("      -r{sampleRate} for example 44100\n");
+        printf("      -s{duration} in seconds, default is %d\n", DEFAULT_DURATION_SECONDS);
+        printf("      -u{usage} eg. 14 for AAUDIO_USAGE_GAME\n");
         printf("      -x to use EXCLUSIVE mode\n");
+        printf("      -y{contentType} eg. 1 for AAUDIO_CONTENT_TYPE_SPEECH\n");
     }
 
     static aaudio_performance_mode_t parsePerformanceMode(char c) {
@@ -281,6 +410,24 @@
 
         printf("  PerformanceMode: requested = %d, actual = %d\n",
                getPerformanceMode(), AAudioStream_getPerformanceMode(stream));
+
+        loadFutureFunctions();
+
+        if (s_setUsage != nullptr) {
+            printf("  Usage:        requested = %d, actual = %d\n",
+                   getUsage(), s_getUsage(stream));
+        }
+        if (s_getContentType != nullptr) {
+            printf("  ContentType:  requested = %d, actual = %d\n",
+                   getContentType(), s_getContentType(stream));
+        }
+
+        if (AAudioStream_getDirection(stream) == AAUDIO_DIRECTION_INPUT
+            && s_getInputPreset != nullptr) {
+                printf("  InputPreset:  requested = %d, actual = %d\n",
+                       getInputPreset(), s_getInputPreset(stream));
+        }
+
         printf("  Is MMAP used? %s\n", AAudioStream_isMMapUsed(stream)
                ? "yes" : "no");
 

diff --git a/media/libaaudio/examples/utils/AAudioExampleUtils.h b/media/libaaudio/examples/utils/AAudioExampleUtils.h
index c179ce6..46b8895 100644
--- a/media/libaaudio/examples/utils/AAudioExampleUtils.h
+++ b/media/libaaudio/examples/utils/AAudioExampleUtils.h

@@ -18,8 +18,10 @@
 #define AAUDIO_EXAMPLE_UTILS_H
 
 #include <atomic>
+#include <errno.h>
 #include <linux/futex.h>
 #include <sched.h>
+#include <string.h>
 #include <sys/syscall.h>
 #include <unistd.h>
 
@@ -30,6 +32,7 @@
 #define NANOS_PER_MILLISECOND (NANOS_PER_MICROSECOND * 1000)
 #define NANOS_PER_SECOND      (NANOS_PER_MILLISECOND * 1000)
 
+template <class T = aaudio_sharing_mode_t>
 const char *getSharingModeText(aaudio_sharing_mode_t mode) {
     const char *text = "unknown";
     switch (mode) {
@@ -78,13 +81,15 @@
     return text;
 }
 
-static void convertNanosecondsToTimespec(int64_t nanoseconds, struct timespec *time) {
+template <class T = int64_t>
+void convertNanosecondsToTimespec(int64_t nanoseconds, struct timespec *time) {
     time->tv_sec = nanoseconds / NANOS_PER_SECOND;
     // Calculate the fractional nanoseconds. Avoids expensive % operation.
     time->tv_nsec = nanoseconds - (time->tv_sec * NANOS_PER_SECOND);
 }
 
-static int64_t getNanoseconds(clockid_t clockId = CLOCK_MONOTONIC) {
+template <class T = clockid_t>
+int64_t getNanoseconds(clockid_t clockId = CLOCK_MONOTONIC) {
     struct timespec time;
     int result = clock_gettime(clockId, &time);
     if (result < 0) {
@@ -93,7 +98,8 @@
     return (time.tv_sec * NANOS_PER_SECOND) + time.tv_nsec;
 }
 
-static void displayPeakLevel(float peakLevel) {
+template <class T = float>
+void displayPeakLevel(float peakLevel) {
     printf("%5.3f ", peakLevel);
     const int maxStars = 50; // arbitrary, fits on one line
     int numStars = (int) (peakLevel * maxStars);
@@ -111,7 +117,8 @@
  * @param sampleRate
  * @return latency in milliseconds
  */
-static double calculateLatencyMillis(int64_t position1, int64_t nanoseconds1,
+template <class T = int64_t>
+double calculateLatencyMillis(int64_t position1, int64_t nanoseconds1,
                               int64_t position2, int64_t nanoseconds2,
                               int64_t sampleRate) {
     int64_t deltaFrames = position2 - position1;
@@ -125,7 +132,8 @@
 
 // ================================================================================
 // These Futex calls are common online examples.
-static android::status_t sys_futex(void *addr1, int op, int val1,
+template <class T = int>
+android::status_t sys_futex(void *addr1, int op, int val1,
                       struct timespec *timeout, void *addr2, int val3) {
     android::status_t result = (android::status_t) syscall(SYS_futex, addr1,
                                                            op, val1, timeout,
@@ -133,12 +141,14 @@
     return (result == 0) ? 0 : -errno;
 }
 
-static android::status_t futex_wake(void *addr, int numWake) {
+template <class T = int>
+android::status_t futex_wake(void *addr, int numWake) {
     // Use _PRIVATE because we are just using the futex in one process.
     return sys_futex(addr, FUTEX_WAKE_PRIVATE, numWake, NULL, NULL, 0);
 }
 
-static android::status_t futex_wait(void *addr, int current, struct timespec *time) {
+template <class T = int>
+android::status_t futex_wait(void *addr, int current, struct timespec *time) {
     // Use _PRIVATE because we are just using the futex in one process.
     return sys_futex(addr, FUTEX_WAIT_PRIVATE, current, time, NULL, 0);
 }

diff --git a/media/libaaudio/examples/utils/AAudioSimplePlayer.h b/media/libaaudio/examples/utils/AAudioSimplePlayer.h
index 606c4ba..54b77ba 100644
--- a/media/libaaudio/examples/utils/AAudioSimplePlayer.h
+++ b/media/libaaudio/examples/utils/AAudioSimplePlayer.h

@@ -19,11 +19,10 @@
 #ifndef AAUDIO_SIMPLE_PLAYER_H
 #define AAUDIO_SIMPLE_PLAYER_H
 
-#include <unistd.h>
 #include <sched.h>
+#include <unistd.h>
 
 #include <aaudio/AAudio.h>
-#include <atomic>
 #include "AAudioArgsParser.h"
 #include "SineGenerator.h"
 
@@ -31,12 +30,12 @@
 #define SHARING_MODE  AAUDIO_SHARING_MODE_SHARED
 #define PERFORMANCE_MODE AAUDIO_PERFORMANCE_MODE_NONE
 
-// Arbitrary period for glitches, once per second at 48000 Hz.
-#define FORCED_UNDERRUN_PERIOD_FRAMES    48000
+// Arbitrary period for glitches
+#define FORCED_UNDERRUN_PERIOD_FRAMES    (2 * 48000)
 // How long to sleep in a callback to cause an intentional glitch. For testing.
 #define FORCED_UNDERRUN_SLEEP_MICROS     (10 * 1000)
 
-#define MAX_TIMESTAMPS   16
+#define MAX_TIMESTAMPS                   16
 
 typedef struct Timestamp {
     int64_t position;
@@ -70,13 +69,6 @@
     }
 
     // TODO Extract a common base class for record and playback.
-    /**
-     * Also known as "sample rate"
-     * Only call this after open() has been called.
-     */
-    int32_t getFramesPerSecond() const {
-        return getSampleRate(); // alias
-    }
 
     /**
      * Only call this after open() has been called.
@@ -172,12 +164,12 @@
         result = AAudioStreamBuilder_openStream(builder, &mStream);
 
         AAudioStreamBuilder_delete(builder);
+
         return result;
     }
 
     aaudio_result_t close() {
         if (mStream != nullptr) {
-            printf("call AAudioStream_close(%p)\n", mStream);  fflush(stdout);
             AAudioStream_close(mStream);
             mStream = nullptr;
         }
@@ -212,13 +204,35 @@
         aaudio_result_t result = AAudioStream_requestStop(mStream);
         if (result != AAUDIO_OK) {
             printf("ERROR - AAudioStream_requestStop() returned %d %s\n",
-                    result, AAudio_convertResultToText(result));
+                   result, AAudio_convertResultToText(result));
         }
         int32_t xRunCount = AAudioStream_getXRunCount(mStream);
         printf("AAudioStream_getXRunCount %d\n", xRunCount);
         return result;
     }
 
+    // Pause the stream. AAudio will stop calling your callback function.
+    aaudio_result_t pause() {
+        aaudio_result_t result = AAudioStream_requestPause(mStream);
+        if (result != AAUDIO_OK) {
+            printf("ERROR - AAudioStream_requestPause() returned %d %s\n",
+                   result, AAudio_convertResultToText(result));
+        }
+        int32_t xRunCount = AAudioStream_getXRunCount(mStream);
+        printf("AAudioStream_getXRunCount %d\n", xRunCount);
+        return result;
+    }
+
+    // Flush the stream. AAudio will stop calling your callback function.
+    aaudio_result_t flush() {
+        aaudio_result_t result = AAudioStream_requestFlush(mStream);
+        if (result != AAUDIO_OK) {
+            printf("ERROR - AAudioStream_requestFlush() returned %d %s\n",
+                   result, AAudio_convertResultToText(result));
+        }
+        return result;
+    }
+
     AAudioStream *getStream() const {
         return mStream;
     }
@@ -232,23 +246,49 @@
 
 typedef struct SineThreadedData_s {
 
-    SineGenerator  sineOsc1;
-    SineGenerator  sineOsc2;
-    Timestamp      timestamps[MAX_TIMESTAMPS];
-    int64_t        framesTotal = 0;
-    int64_t        nextFrameToGlitch = FORCED_UNDERRUN_PERIOD_FRAMES;
-    int32_t        minNumFrames = INT32_MAX;
-    int32_t        maxNumFrames = 0;
-    int32_t        timestampCount = 0; // in timestamps
+    SineGenerator      sineOscillators[MAX_CHANNELS];
+    Timestamp          timestamps[MAX_TIMESTAMPS];
+    int64_t            framesTotal = 0;
+    int64_t            nextFrameToGlitch = FORCED_UNDERRUN_PERIOD_FRAMES;
+    int32_t            minNumFrames = INT32_MAX;
+    int32_t            maxNumFrames = 0;
+    int32_t            timestampCount = 0; // in timestamps
+    int32_t            sampleRate = 48000;
+    int32_t            prefixToneFrames = 0;
+    bool               sweepSetup = false;
 
-    int            scheduler = 0;
-    bool           schedulerChecked = false;
-    bool           forceUnderruns = false;
+    int                scheduler = 0;
+    bool               schedulerChecked = false;
+    bool               forceUnderruns = false;
 
     AAudioSimplePlayer simplePlayer;
     int32_t            callbackCount = 0;
     WakeUp             waker{AAUDIO_OK};
 
+    /**
+     * Set sampleRate first.
+     */
+    void setupSineBlip() {
+        for (int i = 0; i < MAX_CHANNELS; ++i) {
+            double centerFrequency = 880.0 * (i + 2);
+            sineOscillators[i].setup(centerFrequency, sampleRate);
+            sineOscillators[i].setSweep(centerFrequency, centerFrequency, 0.0);
+        }
+    }
+
+    void setupSineSweeps() {
+        for (int i = 0; i < MAX_CHANNELS; ++i) {
+            double centerFrequency = 220.0 * (i + 2);
+            sineOscillators[i].setup(centerFrequency, sampleRate);
+            double minFrequency = centerFrequency * 2.0 / 3.0;
+            // Change range slightly so they will go out of phase.
+            double maxFrequency = centerFrequency * 3.0 / 2.0;
+            double sweepSeconds = 5.0 + i;
+            sineOscillators[i].setSweep(minFrequency, maxFrequency, sweepSeconds);
+        }
+        sweepSetup = true;
+    }
+
 } SineThreadedData_t;
 
 // Callback function that fills the audio output buffer.
@@ -265,9 +305,11 @@
         return AAUDIO_CALLBACK_RESULT_STOP;
     }
     SineThreadedData_t *sineData = (SineThreadedData_t *) userData;
-    sineData->callbackCount++;
 
-    sineData->framesTotal += numFrames;
+    // Play an initial high tone so we can tell whether the beginning was truncated.
+    if (!sineData->sweepSetup && sineData->framesTotal >= sineData->prefixToneFrames) {
+        sineData->setupSineSweeps();
+    }
 
     if (sineData->forceUnderruns) {
         if (sineData->framesTotal > sineData->nextFrameToGlitch) {
@@ -301,33 +343,32 @@
     }
 
     int32_t samplesPerFrame = AAudioStream_getChannelCount(stream);
-    // This code only plays on the first one or two channels.
-    // TODO Support arbitrary number of channels.
+
+
+    int numActiveOscilators = (samplesPerFrame > MAX_CHANNELS) ? MAX_CHANNELS : samplesPerFrame;
     switch (AAudioStream_getFormat(stream)) {
         case AAUDIO_FORMAT_PCM_I16: {
             int16_t *audioBuffer = (int16_t *) audioData;
-            // Render sine waves as shorts to first channel.
-            sineData->sineOsc1.render(&audioBuffer[0], samplesPerFrame, numFrames);
-            // Render sine waves to second channel if there is one.
-            if (samplesPerFrame > 1) {
-                sineData->sineOsc2.render(&audioBuffer[1], samplesPerFrame, numFrames);
+            for (int i = 0; i < numActiveOscilators; ++i) {
+                sineData->sineOscillators[i].render(&audioBuffer[i], samplesPerFrame,
+                                                    numFrames);
             }
         }
-        break;
+            break;
         case AAUDIO_FORMAT_PCM_FLOAT: {
             float *audioBuffer = (float *) audioData;
-            // Render sine waves as floats to first channel.
-            sineData->sineOsc1.render(&audioBuffer[0], samplesPerFrame, numFrames);
-            // Render sine waves to second channel if there is one.
-            if (samplesPerFrame > 1) {
-                sineData->sineOsc2.render(&audioBuffer[1], samplesPerFrame, numFrames);
+            for (int i = 0; i < numActiveOscilators; ++i) {
+                sineData->sineOscillators[i].render(&audioBuffer[i], samplesPerFrame,
+                                                    numFrames);
             }
         }
-        break;
+            break;
         default:
             return AAUDIO_CALLBACK_RESULT_STOP;
     }
 
+    sineData->callbackCount++;
+    sineData->framesTotal += numFrames;
     return AAUDIO_CALLBACK_RESULT_CONTINUE;
 }
 

diff --git a/media/libaaudio/examples/utils/AAudioSimpleRecorder.h b/media/libaaudio/examples/utils/AAudioSimpleRecorder.h
index 1344273..869fad0 100644
--- a/media/libaaudio/examples/utils/AAudioSimpleRecorder.h
+++ b/media/libaaudio/examples/utils/AAudioSimpleRecorder.h

@@ -178,7 +178,6 @@
 
     aaudio_result_t close() {
         if (mStream != nullptr) {
-            printf("call AAudioStream_close(%p)\n", mStream);  fflush(stdout);
             AAudioStream_close(mStream);
             mStream = nullptr;
         }

diff --git a/media/libaaudio/examples/utils/SineGenerator.h b/media/libaaudio/examples/utils/SineGenerator.h
index a755582..9e6d46d 100644
--- a/media/libaaudio/examples/utils/SineGenerator.h
+++ b/media/libaaudio/examples/utils/SineGenerator.h

@@ -31,20 +31,20 @@
     }
 
     void setSweep(double frequencyLow, double frequencyHigh, double seconds) {
-        mPhaseIncrementLow = frequencyLow * M_PI * 2 / mFrameRate;
-        mPhaseIncrementHigh = frequencyHigh * M_PI * 2 / mFrameRate;
-
-        double numFrames = seconds * mFrameRate;
-        mUpScaler = pow((frequencyHigh / frequencyLow), (1.0 / numFrames));
-        mDownScaler = 1.0 / mUpScaler;
-        mGoingUp = true;
-        mSweeping = true;
+        mSweeping = seconds > 0.0;
+        if (mSweeping) {
+            mPhaseIncrementLow = frequencyLow * M_PI * 2 / mFrameRate;
+            mPhaseIncrementHigh = frequencyHigh * M_PI * 2 / mFrameRate;
+            double numFrames = seconds * mFrameRate;
+            mUpScaler = pow((frequencyHigh / frequencyLow), (1.0 / numFrames));
+            mDownScaler = 1.0 / mUpScaler;
+        }
     }
 
     void render(int16_t *buffer, int32_t channelStride, int32_t numFrames) {
         int sampleIndex = 0;
         for (int i = 0; i < numFrames; i++) {
-            buffer[sampleIndex] = (int16_t) (32767 * sin(mPhase) * mAmplitude);
+            buffer[sampleIndex] = (int16_t) (INT16_MAX * sin(mPhase) * mAmplitude);
             sampleIndex += channelStride;
             advancePhase();
         }
@@ -61,6 +61,7 @@
     void setAmplitude(double amplitude) {
         mAmplitude = amplitude;
     }
+
     double getAmplitude() const {
         return mAmplitude;
     }

diff --git a/media/libaaudio/examples/write_sine/Android.bp b/media/libaaudio/examples/write_sine/Android.bp
new file mode 100644
index 0000000..aa25e67
--- /dev/null
+++ b/media/libaaudio/examples/write_sine/Android.bp

@@ -0,0 +1,15 @@
+cc_test {
+    name: "write_sine",
+    srcs: ["src/write_sine.cpp"],
+    cflags: ["-Wall", "-Werror"],
+    shared_libs: ["libaaudio"],
+    header_libs: ["libaaudio_example_utils"],
+}
+
+cc_test {
+    name: "write_sine_callback",
+    srcs: ["src/write_sine_callback.cpp"],
+    cflags: ["-Wall", "-Werror"],
+    shared_libs: ["libaaudio"],
+    header_libs: ["libaaudio_example_utils"],
+}

diff --git a/media/libaaudio/examples/write_sine/Android.mk b/media/libaaudio/examples/write_sine/Android.mk
deleted file mode 100644
index 5053e7d..0000000
--- a/media/libaaudio/examples/write_sine/Android.mk
+++ /dev/null

@@ -1 +0,0 @@
-include $(call all-subdir-makefiles)

diff --git a/media/libaaudio/examples/write_sine/jni/Android.mk b/media/libaaudio/examples/write_sine/jni/Android.mk
index d630e76..1a1bd43 100644
--- a/media/libaaudio/examples/write_sine/jni/Android.mk
+++ b/media/libaaudio/examples/write_sine/jni/Android.mk

@@ -10,6 +10,7 @@
 
 # NDK recommends using this kind of relative path instead of an absolute path.
 LOCAL_SRC_FILES:= ../src/write_sine.cpp
+LOCAL_CFLAGS := -Wall -Werror
 LOCAL_SHARED_LIBRARIES := libaaudio
 LOCAL_MODULE := write_sine
 include $(BUILD_EXECUTABLE)
@@ -22,6 +23,7 @@
     frameworks/av/media/libaaudio/examples/utils
 
 LOCAL_SRC_FILES:= ../src/write_sine_callback.cpp
+LOCAL_CFLAGS := -Wall -Werror
 LOCAL_SHARED_LIBRARIES := libaaudio
 LOCAL_MODULE := write_sine_callback
 include $(BUILD_EXECUTABLE)

diff --git a/media/libaaudio/examples/write_sine/src/write_sine.cpp b/media/libaaudio/examples/write_sine/src/write_sine.cpp
index 677fb6c..8e33a31 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine.cpp

@@ -44,10 +44,10 @@
     AAudioStream *aaudioStream = nullptr;
     int32_t  framesPerBurst = 0;
     int32_t  framesPerWrite = 0;
-    int32_t  bufferCapacity = 0;
     int32_t  framesToPlay = 0;
     int32_t  framesLeft = 0;
     int32_t  xRunCount = 0;
+    int      numActiveOscilators = 0;
     float   *floatData = nullptr;
     int16_t *shortData = nullptr;
 
@@ -57,7 +57,7 @@
     // in a buffer if we hang or crash.
     setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
 
-    printf("%s - Play a sine wave using AAudio V0.1.2\n", argv[0]);
+    printf("%s - Play a sine wave using AAudio V0.1.3\n", argv[0]);
 
     if (argParser.parseArgs(argc, argv)) {
         return EXIT_FAILURE;
@@ -77,8 +77,8 @@
     actualSampleRate = AAudioStream_getSampleRate(aaudioStream);
     actualDataFormat = AAudioStream_getFormat(aaudioStream);
 
-    myData.sineOsc1.setup(440.0, actualSampleRate);
-    myData.sineOsc2.setup(660.0, actualSampleRate);
+    myData.sampleRate = actualSampleRate;
+    myData.setupSineSweeps();
 
     // Some DMA might use very short bursts of 16 frames. We don't need to write such small
     // buffers. But it helps to use a multiple of the burst size for predictable scheduling.
@@ -117,19 +117,18 @@
     // Play for a while.
     framesToPlay = actualSampleRate * argParser.getDurationSeconds();
     framesLeft = framesToPlay;
+    numActiveOscilators = (actualChannelCount > MAX_CHANNELS) ? MAX_CHANNELS : actualChannelCount;
     while (framesLeft > 0) {
-
+        // Render as FLOAT or PCM
         if (actualDataFormat == AAUDIO_FORMAT_PCM_FLOAT) {
-            // Render sine waves to left and right channels.
-            myData.sineOsc1.render(&floatData[0], actualChannelCount, framesPerWrite);
-            if (actualChannelCount > 1) {
-                myData.sineOsc2.render(&floatData[1], actualChannelCount, framesPerWrite);
+            for (int i = 0; i < numActiveOscilators; ++i) {
+                myData.sineOscillators[i].render(&floatData[i], actualChannelCount,
+                                                  framesPerWrite);
             }
         } else if (actualDataFormat == AAUDIO_FORMAT_PCM_I16) {
-            // Render sine waves to left and right channels.
-            myData.sineOsc1.render(&shortData[0], actualChannelCount, framesPerWrite);
-            if (actualChannelCount > 1) {
-                myData.sineOsc2.render(&shortData[1], actualChannelCount, framesPerWrite);
+            for (int i = 0; i < numActiveOscilators; ++i) {
+                myData.sineOscillators[i].render(&shortData[i], actualChannelCount,
+                                                  framesPerWrite);
             }
         }
 

diff --git a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
index 4f9cde6..e33e9f8 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp

@@ -28,7 +28,7 @@
 #include <aaudio/AAudio.h>
 #include "AAudioExampleUtils.h"
 #include "AAudioSimplePlayer.h"
-#include "../../utils/AAudioSimplePlayer.h"
+#include "AAudioArgsParser.h"
 
 /**
  * Open stream, play some sine waves, then close the stream.
@@ -36,37 +36,39 @@
  * @param argParser
  * @return AAUDIO_OK or negative error code
  */
-static aaudio_result_t testOpenPlayClose(AAudioArgsParser &argParser)
+static aaudio_result_t testOpenPlayClose(AAudioArgsParser &argParser,
+                                         int32_t loopCount,
+                                         int32_t prefixToneMsec,
+                                         bool forceUnderruns)
 {
     SineThreadedData_t myData;
     AAudioSimplePlayer &player = myData.simplePlayer;
     aaudio_result_t    result = AAUDIO_OK;
     bool               disconnected = false;
+    bool               bailOut = false;
     int64_t            startedAtNanos;
 
     printf("----------------------- run complete test --------------------------\n");
     myData.schedulerChecked = false;
     myData.callbackCount = 0;
-    myData.forceUnderruns = false; // set true to test AAudioStream_getXRunCount()
+    myData.forceUnderruns = forceUnderruns; // test AAudioStream_getXRunCount()
 
     result = player.open(argParser,
                          SimplePlayerDataCallbackProc, SimplePlayerErrorCallbackProc, &myData);
     if (result != AAUDIO_OK) {
-        fprintf(stderr, "ERROR -  player.open() returned %d\n", result);
+        fprintf(stderr, "ERROR -  player.open() returned %s\n",
+                AAudio_convertResultToText(result));
         goto error;
     }
 
     argParser.compareWithStream(player.getStream());
 
-    // Setup sine wave generators.
-    {
-        int32_t actualSampleRate = player.getSampleRate();
-        myData.sineOsc1.setup(440.0, actualSampleRate);
-        myData.sineOsc1.setSweep(300.0, 600.0, 5.0);
-        myData.sineOsc1.setAmplitude(0.2);
-        myData.sineOsc2.setup(660.0, actualSampleRate);
-        myData.sineOsc2.setSweep(350.0, 900.0, 7.0);
-        myData.sineOsc2.setAmplitude(0.2);
+    myData.sampleRate = player.getSampleRate();
+    myData.prefixToneFrames = prefixToneMsec * myData.sampleRate / 1000;
+    if (myData.prefixToneFrames > 0) {
+        myData.setupSineBlip();
+    } else {
+        myData.setupSineSweeps();
     }
 
 #if 0
@@ -78,42 +80,93 @@
     }
 #endif
 
-    result = player.start();
-    if (result != AAUDIO_OK) {
-        fprintf(stderr, "ERROR - player.start() returned %d\n", result);
-        goto error;
-    }
+    for (int loopIndex = 0; loopIndex < loopCount; loopIndex++) {
+        // Only play data on every other loop so we can hear if there is stale data.
+        double amplitude;
+        int32_t durationSeconds;
+        if ((loopIndex & 1) == 0) {
+            printf("--------------- SINE ------\n");
+            amplitude = 0.2;
+            durationSeconds = argParser.getDurationSeconds();
+        } else {
+            printf("--------------- QUIET -----\n");
+            amplitude = 0.0;
+            durationSeconds = 2; // just wait briefly when quiet
+        }
+        for (int i = 0; i < MAX_CHANNELS; ++i) {
+            myData.sineOscillators[i].setAmplitude(amplitude);
+        }
 
-    // Play a sine wave in the background.
-    printf("Sleep for %d seconds while audio plays in a callback thread.\n",
-           argParser.getDurationSeconds());
-    startedAtNanos = getNanoseconds(CLOCK_MONOTONIC);
-    for (int second = 0; second < argParser.getDurationSeconds(); second++)
-    {
-        // Sleep a while. Wake up early if there is an error, for example a DISCONNECT.
-        long ret = myData.waker.wait(AAUDIO_OK, NANOS_PER_SECOND);
-        int64_t millis = (getNanoseconds(CLOCK_MONOTONIC) - startedAtNanos) / NANOS_PER_MILLISECOND;
-        result = myData.waker.get();
-        printf("wait() returns %ld, aaudio_result = %d, at %6d millis"
-               ", second = %d, framesWritten = %8d, underruns = %d\n",
-               ret, result, (int) millis,
-               second,
-               (int) AAudioStream_getFramesWritten(player.getStream()),
-               (int) AAudioStream_getXRunCount(player.getStream()));
+        result = player.start();
         if (result != AAUDIO_OK) {
-            if (result == AAUDIO_ERROR_DISCONNECTED) {
-                disconnected = true;
+            fprintf(stderr, "ERROR - player.start() returned %d\n", result);
+            goto error;
+        }
+
+        // Play a sine wave in the background.
+        printf("Sleep for %d seconds while audio plays in a callback thread. %d of %d\n",
+               argParser.getDurationSeconds(), (loopIndex + 1), loopCount);
+        startedAtNanos = getNanoseconds(CLOCK_MONOTONIC);
+        for (int second = 0; second < durationSeconds; second++) {
+            // Sleep a while. Wake up early if there is an error, for example a DISCONNECT.
+            long ret = myData.waker.wait(AAUDIO_OK, NANOS_PER_SECOND);
+            int64_t millis =
+                    (getNanoseconds(CLOCK_MONOTONIC) - startedAtNanos) / NANOS_PER_MILLISECOND;
+            result = myData.waker.get();
+            printf("wait() returns %ld, aaudio_result = %d, at %6d millis"
+                           ", second = %3d, framesWritten = %8d, underruns = %d\n",
+                   ret, result, (int) millis,
+                   second,
+                   (int) AAudioStream_getFramesWritten(player.getStream()),
+                   (int) AAudioStream_getXRunCount(player.getStream()));
+            if (result != AAUDIO_OK) {
+                disconnected = (result == AAUDIO_ERROR_DISCONNECTED);
+                bailOut = true;
+                break;
             }
+        }
+        printf("AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
+
+        // Alternate between using stop or pause for each sine/quiet pair.
+        // Repeat this pattern: {sine-stop-quiet-stop-sine-pause-quiet-pause}
+        if ((loopIndex & 2) == 0) {
+            printf("STOP, callback # = %d\n", myData.callbackCount);
+            result = player.stop();
+        } else {
+            printf("PAUSE/FLUSH, callback # = %d\n", myData.callbackCount);
+            result = player.pause();
+            if (result != AAUDIO_OK) {
+                goto error;
+            }
+            result = player.flush();
+        }
+        if (result != AAUDIO_OK) {
+            goto error;
+        }
+
+        if (bailOut) {
             break;
         }
-    }
-    printf("AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
 
-    printf("call stop() callback # = %d\n", myData.callbackCount);
-    result = player.stop();
-    if (result != AAUDIO_OK) {
-        goto error;
+        {
+            aaudio_stream_state_t state = AAudioStream_getState(player.getStream());
+            aaudio_stream_state_t finalState = AAUDIO_STREAM_STATE_UNINITIALIZED;
+            int64_t timeoutNanos = 2000 * NANOS_PER_MILLISECOND;
+            result = AAudioStream_waitForStateChange(player.getStream(), state,
+                                                     &finalState, timeoutNanos);
+            printf("waitForStateChange returns %s, state = %s\n",
+                   AAudio_convertResultToText(result),
+                   AAudio_convertStreamStateToText(finalState));
+            int64_t written = AAudioStream_getFramesWritten(player.getStream());
+            int64_t read = AAudioStream_getFramesRead(player.getStream());
+            printf("   framesWritten = %lld, framesRead = %lld, diff = %d\n",
+                   (long long) written,
+                   (long long) read,
+                   (int) (written - read));
+        }
+
     }
+
     printf("call close()\n");
     result = player.close();
     if (result != AAUDIO_OK) {
@@ -147,23 +200,59 @@
     return disconnected ? AAUDIO_ERROR_DISCONNECTED : result;
 }
 
+static void usage() {
+    AAudioArgsParser::usage();
+    printf("      -l{count} loopCount start/stop, every other one is silent\n");
+    printf("      -t{msec}  play a high pitched tone at the beginning\n");
+    printf("      -z        force periodic underruns by sleeping in callback\n");
+}
+
 int main(int argc, const char **argv)
 {
     AAudioArgsParser   argParser;
     aaudio_result_t    result;
+    int32_t            loopCount = 1;
+    int32_t            prefixToneMsec = 0;
+    bool               forceUnderruns = false;
 
     // Make printf print immediately so that debug info is not stuck
     // in a buffer if we hang or crash.
     setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
 
-    printf("%s - Play a sine sweep using an AAudio callback V0.1.2\n", argv[0]);
+    printf("%s - Play a sine sweep using an AAudio callback V0.1.4\n", argv[0]);
 
-    if (argParser.parseArgs(argc, argv)) {
-        return EXIT_FAILURE;
+    for (int i = 1; i < argc; i++) {
+        const char *arg = argv[i];
+        if (argParser.parseArg(arg)) {
+            // Handle options that are not handled by the ArgParser
+            if (arg[0] == '-') {
+                char option = arg[1];
+                switch (option) {
+                    case 'l':
+                        loopCount = atoi(&arg[2]);
+                        break;
+                    case 't':
+                        prefixToneMsec = atoi(&arg[2]);
+                        break;
+                    case 'z':
+                        forceUnderruns = true;  // Zzzzzzz
+                        break;
+                    default:
+                        usage();
+                        exit(EXIT_FAILURE);
+                        break;
+                }
+            } else {
+                usage();
+                exit(EXIT_FAILURE);
+                break;
+            }
+        }
     }
 
     // Keep looping until we can complete the test without disconnecting.
-    while((result = testOpenPlayClose(argParser)) == AAUDIO_ERROR_DISCONNECTED);
+    while((result = testOpenPlayClose(argParser, loopCount, prefixToneMsec, forceUnderruns))
+            == AAUDIO_ERROR_DISCONNECTED);
 
     return (result) ? EXIT_FAILURE : EXIT_SUCCESS;
 }

diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index 3c23736..5b29419 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h

@@ -44,7 +44,15 @@
 #define AAUDIO_UNSPECIFIED           0
 
 enum {
+    /**
+     * Audio data will travel out of the device, for example through a speaker.
+     */
     AAUDIO_DIRECTION_OUTPUT,
+
+
+    /**
+     * Audio data will travel into the device, for example from a microphone.
+     */
     AAUDIO_DIRECTION_INPUT
 };
 typedef int32_t aaudio_direction_t;
@@ -52,33 +60,112 @@
 enum {
     AAUDIO_FORMAT_INVALID = -1,
     AAUDIO_FORMAT_UNSPECIFIED = 0,
+
+    /**
+     * This format uses the int16_t data type.
+     * The maximum range of the data is -32768 to 32767.
+     */
     AAUDIO_FORMAT_PCM_I16,
+
+    /**
+     * This format uses the float data type.
+     * The nominal range of the data is [-1.0f, 1.0f).
+     * Values outside that range may be clipped.
+     *
+     * See also 'floatData' at
+     * https://developer.android.com/reference/android/media/AudioTrack#write(float[],%20int,%20int,%20int)
+     */
     AAUDIO_FORMAT_PCM_FLOAT
 };
 typedef int32_t aaudio_format_t;
 
+/**
+ * These result codes are returned from AAudio functions to indicate success or failure.
+ * Note that error return codes may change in the future so applications should generally
+ * not rely on specific return codes.
+ */
 enum {
+    /**
+     * The call was successful.
+     */
     AAUDIO_OK,
     AAUDIO_ERROR_BASE = -900, // TODO review
+
+    /**
+     * The audio device was disconnected. This could occur, for example, when headphones
+     * are plugged in or unplugged. The stream cannot be used after the device is disconnected.
+     * Applications should stop and close the stream.
+     * If this error is received in an error callback then another thread should be
+     * used to stop and close the stream.
+     */
     AAUDIO_ERROR_DISCONNECTED,
+
+    /**
+     * An invalid parameter was passed to AAudio.
+     */
     AAUDIO_ERROR_ILLEGAL_ARGUMENT,
     // reserved
     AAUDIO_ERROR_INTERNAL = AAUDIO_ERROR_ILLEGAL_ARGUMENT + 2,
+
+    /**
+     * The requested operation is not appropriate for the current state of AAudio.
+     */
     AAUDIO_ERROR_INVALID_STATE,
     // reserved
     // reserved
+    /* The server rejected the handle used to identify the stream.
+     */
     AAUDIO_ERROR_INVALID_HANDLE = AAUDIO_ERROR_INVALID_STATE + 3,
     // reserved
+
+    /**
+     * The function is not implemented for this stream.
+     */
     AAUDIO_ERROR_UNIMPLEMENTED = AAUDIO_ERROR_INVALID_HANDLE + 2,
+
+    /**
+     * A resource or information is unavailable.
+     * This could occur when an application tries to open too many streams,
+     * or a timestamp is not available.
+     */
     AAUDIO_ERROR_UNAVAILABLE,
     AAUDIO_ERROR_NO_FREE_HANDLES,
+
+    /**
+     * Memory could not be allocated.
+     */
     AAUDIO_ERROR_NO_MEMORY,
+
+    /**
+     * A NULL pointer was passed to AAudio.
+     * Or a NULL pointer was detected internally.
+     */
     AAUDIO_ERROR_NULL,
+
+    /**
+     * An operation took longer than expected.
+     */
     AAUDIO_ERROR_TIMEOUT,
     AAUDIO_ERROR_WOULD_BLOCK,
+
+    /**
+     * The requested data format is not supported.
+     */
     AAUDIO_ERROR_INVALID_FORMAT,
+
+    /**
+     * A requested was out of range.
+     */
     AAUDIO_ERROR_OUT_OF_RANGE,
+
+    /**
+     * The audio service was not available.
+     */
     AAUDIO_ERROR_NO_SERVICE,
+
+    /**
+     * The requested sample rate was not supported.
+     */
     AAUDIO_ERROR_INVALID_RATE
 };
 typedef int32_t  aaudio_result_t;
@@ -123,20 +210,200 @@
     /**
      * No particular performance needs. Default.
      */
-            AAUDIO_PERFORMANCE_MODE_NONE = 10,
+    AAUDIO_PERFORMANCE_MODE_NONE = 10,
 
     /**
-     * Extending battery life is most important.
+     * Extending battery life is more important than low latency.
+     *
+     * This mode is not supported in input streams.
+     * For input, mode NONE will be used if this is requested.
      */
-            AAUDIO_PERFORMANCE_MODE_POWER_SAVING,
+    AAUDIO_PERFORMANCE_MODE_POWER_SAVING,
 
     /**
-     * Reducing latency is most important.
+     * Reducing latency is more important than battery life.
      */
-            AAUDIO_PERFORMANCE_MODE_LOW_LATENCY
+    AAUDIO_PERFORMANCE_MODE_LOW_LATENCY
 };
 typedef int32_t aaudio_performance_mode_t;
 
+/**
+ * The USAGE attribute expresses "why" you are playing a sound, what is this sound used for.
+ * This information is used by certain platforms or routing policies
+ * to make more refined volume or routing decisions.
+ *
+ * Note that these match the equivalent values in AudioAttributes in the Android Java API.
+ *
+ * Added in API level 28.
+ */
+enum {
+    /**
+     * Use this for streaming media, music performance, video, podcasts, etcetera.
+     */
+    AAUDIO_USAGE_MEDIA = 1,
+
+    /**
+     * Use this for voice over IP, telephony, etcetera.
+     */
+    AAUDIO_USAGE_VOICE_COMMUNICATION = 2,
+
+    /**
+     * Use this for sounds associated with telephony such as busy tones, DTMF, etcetera.
+     */
+    AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING = 3,
+
+    /**
+     * Use this to demand the users attention.
+     */
+    AAUDIO_USAGE_ALARM = 4,
+
+    /**
+     * Use this for notifying the user when a message has arrived or some
+     * other background event has occured.
+     */
+    AAUDIO_USAGE_NOTIFICATION = 5,
+
+    /**
+     * Use this when the phone rings.
+     */
+    AAUDIO_USAGE_NOTIFICATION_RINGTONE = 6,
+
+    /**
+     * Use this to attract the users attention when, for example, the battery is low.
+     */
+    AAUDIO_USAGE_NOTIFICATION_EVENT = 10,
+
+    /**
+     * Use this for screen readers, etcetera.
+     */
+    AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY = 11,
+
+    /**
+     * Use this for driving or navigation directions.
+     */
+    AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE = 12,
+
+    /**
+     * Use this for user interface sounds, beeps, etcetera.
+     */
+    AAUDIO_USAGE_ASSISTANCE_SONIFICATION = 13,
+
+    /**
+     * Use this for game audio and sound effects.
+     */
+    AAUDIO_USAGE_GAME = 14,
+
+    /**
+     * Use this for audio responses to user queries, audio instructions or help utterances.
+     */
+    AAUDIO_USAGE_ASSISTANT = 16
+};
+typedef int32_t aaudio_usage_t;
+
+/**
+ * The CONTENT_TYPE attribute describes "what" you are playing.
+ * It expresses the general category of the content. This information is optional.
+ * But in case it is known (for instance {@link #AAUDIO_CONTENT_TYPE_MOVIE} for a
+ * movie streaming service or {@link #AAUDIO_CONTENT_TYPE_SPEECH} for
+ * an audio book application) this information might be used by the audio framework to
+ * enforce audio focus.
+ *
+ * Note that these match the equivalent values in AudioAttributes in the Android Java API.
+ *
+ * Added in API level 28.
+ */
+enum {
+
+    /**
+     * Use this for spoken voice, audio books, etcetera.
+     */
+    AAUDIO_CONTENT_TYPE_SPEECH = 1,
+
+    /**
+     * Use this for pre-recorded or live music.
+     */
+    AAUDIO_CONTENT_TYPE_MUSIC = 2,
+
+    /**
+     * Use this for a movie or video soundtrack.
+     */
+    AAUDIO_CONTENT_TYPE_MOVIE = 3,
+
+    /**
+     * Use this for sound is designed to accompany a user action,
+     * such as a click or beep sound made when the user presses a button.
+     */
+    AAUDIO_CONTENT_TYPE_SONIFICATION = 4
+};
+typedef int32_t aaudio_content_type_t;
+
+/**
+ * Defines the audio source.
+ * An audio source defines both a default physical source of audio signal, and a recording
+ * configuration.
+ *
+ * Note that these match the equivalent values in MediaRecorder.AudioSource in the Android Java API.
+ *
+ * Added in API level 28.
+ */
+enum {
+    /**
+     * Use this preset when other presets do not apply.
+     */
+    AAUDIO_INPUT_PRESET_GENERIC = 1,
+
+    /**
+     * Use this preset when recording video.
+     */
+    AAUDIO_INPUT_PRESET_CAMCORDER = 5,
+
+    /**
+     * Use this preset when doing speech recognition.
+     */
+    AAUDIO_INPUT_PRESET_VOICE_RECOGNITION = 6,
+
+    /**
+     * Use this preset when doing telephony or voice messaging.
+     */
+    AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION = 7,
+
+    /**
+     * Use this preset to obtain an input with no effects.
+     * Note that this input will not have automatic gain control
+     * so the recorded volume may be very low.
+     */
+    AAUDIO_INPUT_PRESET_UNPROCESSED = 9,
+};
+typedef int32_t aaudio_input_preset_t;
+
+/**
+ * These may be used with AAudioStreamBuilder_setSessionId().
+ *
+ * Added in API level 28.
+ */
+enum {
+    /**
+     * Do not allocate a session ID.
+     * Effects cannot be used with this stream.
+     * Default.
+     *
+     * Added in API level 28.
+     */
+    AAUDIO_SESSION_ID_NONE = -1,
+
+    /**
+     * Allocate a session ID that can be used to attach and control
+     * effects using the Java AudioEffects API.
+     * Note that using this may result in higher latency.
+     *
+     * Note that this matches the value of AudioManager.AUDIO_SESSION_ID_GENERATE.
+     *
+     * Added in API level 28.
+     */
+    AAUDIO_SESSION_ID_ALLOCATE = 0,
+};
+typedef int32_t aaudio_session_id_t;
+
 typedef struct AAudioStreamStruct         AAudioStream;
 typedef struct AAudioStreamBuilderStruct  AAudioStreamBuilder;
 
@@ -299,8 +566,14 @@
 /**
  * Set the requested performance mode.
  *
+ * Supported modes are AAUDIO_PERFORMANCE_MODE_NONE, AAUDIO_PERFORMANCE_MODE_POWER_SAVING
+ * and AAUDIO_PERFORMANCE_MODE_LOW_LATENCY.
+ *
  * The default, if you do not call this function, is AAUDIO_PERFORMANCE_MODE_NONE.
  *
+ * You may not get the mode you requested.
+ * You can call AAudioStream_getPerformanceMode() to find out the final mode for the stream.
+ *
  * @param builder reference provided by AAudio_createStreamBuilder()
  * @param mode the desired performance mode, eg. AAUDIO_PERFORMANCE_MODE_LOW_LATENCY
  */
@@ -308,6 +581,90 @@
                                                 aaudio_performance_mode_t mode);
 
 /**
+ * Set the intended use case for the stream.
+ *
+ * The AAudio system will use this information to optimize the
+ * behavior of the stream.
+ * This could, for example, affect how volume and focus is handled for the stream.
+ *
+ * The default, if you do not call this function, is AAUDIO_USAGE_MEDIA.
+ *
+ * Added in API level 28.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param usage the desired usage, eg. AAUDIO_USAGE_GAME
+ */
+AAUDIO_API void AAudioStreamBuilder_setUsage(AAudioStreamBuilder* builder,
+                                                       aaudio_usage_t usage);
+
+/**
+ * Set the type of audio data that the stream will carry.
+ *
+ * The AAudio system will use this information to optimize the
+ * behavior of the stream.
+ * This could, for example, affect whether a stream is paused when a notification occurs.
+ *
+ * The default, if you do not call this function, is AAUDIO_CONTENT_TYPE_MUSIC.
+ *
+ * Added in API level 28.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param contentType the type of audio data, eg. AAUDIO_CONTENT_TYPE_SPEECH
+ */
+AAUDIO_API void AAudioStreamBuilder_setContentType(AAudioStreamBuilder* builder,
+                                             aaudio_content_type_t contentType);
+
+/**
+ * Set the input (capture) preset for the stream.
+ *
+ * The AAudio system will use this information to optimize the
+ * behavior of the stream.
+ * This could, for example, affect which microphones are used and how the
+ * recorded data is processed.
+ *
+ * The default, if you do not call this function, is AAUDIO_INPUT_PRESET_VOICE_RECOGNITION.
+ * That is because VOICE_RECOGNITION is the preset with the lowest latency
+ * on many platforms.
+ *
+ * Added in API level 28.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param inputPreset the desired configuration for recording
+ */
+AAUDIO_API void AAudioStreamBuilder_setInputPreset(AAudioStreamBuilder* builder,
+                                                   aaudio_input_preset_t inputPreset);
+
+/** Set the requested session ID.
+ *
+ * The session ID can be used to associate a stream with effects processors.
+ * The effects are controlled using the Android AudioEffect Java API.
+ *
+ * The default, if you do not call this function, is AAUDIO_SESSION_ID_NONE.
+ *
+ * If set to AAUDIO_SESSION_ID_ALLOCATE then a session ID will be allocated
+ * when the stream is opened.
+ *
+ * The allocated session ID can be obtained by calling AAudioStream_getSessionId()
+ * and then used with this function when opening another stream.
+ * This allows effects to be shared between streams.
+ *
+ * Session IDs from AAudio can be used with the Android Java APIs and vice versa.
+ * So a session ID from an AAudio stream can be passed to Java
+ * and effects applied using the Java AudioEffect API.
+ *
+ * Note that allocating or setting a session ID may result in a stream with higher latency.
+ *
+ * Allocated session IDs will always be positive and nonzero.
+ *
+ * Added in API level 28.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param sessionId an allocated sessionID or AAUDIO_SESSION_ID_ALLOCATE
+ */
+AAUDIO_API void AAudioStreamBuilder_setSessionId(AAudioStreamBuilder* builder,
+                                                aaudio_session_id_t sessionId);
+
+/**
  * Return one of these values from the data callback function.
  */
 enum {
@@ -337,7 +694,13 @@
  * For an input stream, this function should read and process numFrames of data
  * from the audioData buffer.
  *
- * Note that this callback function should be considered a "real-time" function.
+ * The audio data is passed through the buffer. So do NOT call AAudioStream_read() or
+ * AAudioStream_write() on the stream that is making the callback.
+ *
+ * Note that numFrames can vary unless AAudioStreamBuilder_setFramesPerDataCallback()
+ * is called.
+ *
+ * Also note that this callback function should be considered a "real-time" function.
  * It must not do anything that could cause an unbounded delay because that can cause the
  * audio to glitch or pop.
  *
@@ -348,6 +711,15 @@
  * <li>any network operations such as streaming</li>
  * <li>use any mutexes or other synchronization primitives</li>
  * <li>sleep</li>
+ * <li>stop or close the stream</li>
+ * <li>AAudioStream_read()</li>
+ * <li>AAudioStream_write()</li>
+ * </ul>
+ *
+ * The following are OK to call from the data callback:
+ * <ul>
+ * <li>AAudioStream_get*()</li>
+ * <li>AAudio_convertResultToText()</li>
  * </ul>
  *
  * If you need to move data, eg. MIDI commands, in or out of the callback function then
@@ -356,7 +728,7 @@
  * @param stream reference provided by AAudioStreamBuilder_openStream()
  * @param userData the same address that was passed to AAudioStreamBuilder_setCallback()
  * @param audioData a pointer to the audio data
- * @param numFrames the number of frames to be processed
+ * @param numFrames the number of frames to be processed, which can vary
  * @return AAUDIO_CALLBACK_RESULT_*
  */
 typedef aaudio_data_callback_result_t (*AAudioStream_dataCallback)(
@@ -421,6 +793,22 @@
  * Prototype for the callback function that is passed to
  * AAudioStreamBuilder_setErrorCallback().
  *
+ * The following may NOT be called from the error callback:
+ * <ul>
+ * <li>AAudioStream_requestStop()</li>
+ * <li>AAudioStream_requestPause()</li>
+ * <li>AAudioStream_close()</li>
+ * <li>AAudioStream_waitForStateChange()</li>
+ * <li>AAudioStream_read()</li>
+ * <li>AAudioStream_write()</li>
+ * </ul>
+ *
+ * The following are OK to call from the error callback:
+ * <ul>
+ * <li>AAudioStream_get*()</li>
+ * <li>AAudio_convertResultToText()</li>
+ * </ul>
+ *
  * @param stream reference provided by AAudioStreamBuilder_openStream()
  * @param userData the same address that was passed to AAudioStreamBuilder_setErrorCallback()
  * @param error an AAUDIO_ERROR_* value.
@@ -431,18 +819,19 @@
         aaudio_result_t error);
 
 /**
- * Request that AAudio call this functions if any error occurs on a callback thread.
+ * Request that AAudio call this function if any error occurs or the stream is disconnected.
  *
  * It will be called, for example, if a headset or a USB device is unplugged causing the stream's
- * device to be unavailable.
- * In response, this function could signal or launch another thread to reopen a
- * stream on another device. Do not reopen the stream in this callback.
- *
- * This will not be called because of actions by the application, such as stopping
- * or closing a stream.
- *
+ * device to be unavailable or "disconnected".
  * Another possible cause of error would be a timeout or an unanticipated internal error.
  *
+ * In response, this function should signal or create another thread to stop
+ * and close this stream. The other thread could then reopen a stream on another device.
+ * Do not stop or close the stream, or reopen the new stream, directly from this callback.
+ *
+ * This callback will not be called because of actions by the application, such as stopping
+ * or closing a stream.
+ *
  * Note that the AAudio callbacks will never be called simultaneously from multiple threads.
  *
  * @param builder reference provided by AAudio_createStreamBuilder()
@@ -554,11 +943,13 @@
  * This will update the current client state.
  *
  * <pre><code>
- * aaudio_stream_state_t currentState;
- * aaudio_result_t result = AAudioStream_getState(stream, &currentState);
- * while (result == AAUDIO_OK && currentState != AAUDIO_STREAM_STATE_PAUSING) {
+ * aaudio_result_t result = AAUDIO_OK;
+ * aaudio_stream_state_t currentState = AAudioStream_getState(stream);
+ * aaudio_stream_state_t inputState = currentState;
+ * while (result == AAUDIO_OK && currentState != AAUDIO_STREAM_STATE_PAUSED) {
  *     result = AAudioStream_waitForStateChange(
- *                                   stream, currentState, &currentState, MY_TIMEOUT_NANOS);
+ *                                   stream, inputState, &currentState, MY_TIMEOUT_NANOS);
+ *     inputState = currentState;
  * }
  * </code></pre>
  *
@@ -589,6 +980,8 @@
  *
  * This call is "strong non-blocking" unless it has to wait for data.
  *
+ * If the call times out then zero or a partial frame count will be returned.
+ *
  * @param stream A stream created using AAudioStreamBuilder_openStream().
  * @param buffer The address of the first sample.
  * @param numFrames Number of frames to read. Only complete frames will be written.
@@ -612,6 +1005,8 @@
  *
  * This call is "strong non-blocking" unless it has to wait for room in the buffer.
  *
+ * If the call times out then zero or a partial frame count will be returned.
+ *
  * @param stream A stream created using AAudioStreamBuilder_openStream().
  * @param buffer The address of the first sample.
  * @param numFrames Number of frames to write. Only complete frames will be written.
@@ -636,7 +1031,8 @@
  * This cannot be set higher than AAudioStream_getBufferCapacityInFrames().
  *
  * Note that you will probably not get the exact size you request.
- * Call AAudioStream_getBufferSizeInFrames() to see what the actual final size is.
+ * You can check the return value or call AAudioStream_getBufferSizeInFrames()
+ * to see what the actual final size is.
  *
  * @param stream reference provided by AAudioStreamBuilder_openStream()
  * @param numFrames requested number of frames that can be filled without blocking
@@ -683,10 +1079,10 @@
  * This call can be used if the application needs to know the value of numFrames before
  * the stream is started. This is not normally necessary.
  *
- * If a specific size was requested by calling AAudioStreamBuilder_setCallbackSizeInFrames()
+ * If a specific size was requested by calling AAudioStreamBuilder_setFramesPerDataCallback()
  * then this will be the same size.
  *
- * If AAudioStreamBuilder_setCallbackSizeInFrames() was not called then this will
+ * If AAudioStreamBuilder_setFramesPerDataCallback() was not called then this will
  * return the size chosen by AAudio, or AAUDIO_UNSPECIFIED.
  *
  * AAUDIO_UNSPECIFIED indicates that the callback buffer size for this stream
@@ -771,7 +1167,8 @@
 
 /**
  * Passes back the number of frames that have been written since the stream was created.
- * For an output stream, this will be advanced by the application calling write().
+ * For an output stream, this will be advanced by the application calling write()
+ * or by a data callback.
  * For an input stream, this will be advanced by the endpoint.
  *
  * The frame position is monotonically increasing.
@@ -784,7 +1181,8 @@
 /**
  * Passes back the number of frames that have been read since the stream was created.
  * For an output stream, this will be advanced by the endpoint.
- * For an input stream, this will be advanced by the application calling read().
+ * For an input stream, this will be advanced by the application calling read()
+ * or by a data callback.
  *
  * The frame position is monotonically increasing.
  *
@@ -794,6 +1192,30 @@
 AAUDIO_API int64_t AAudioStream_getFramesRead(AAudioStream* stream);
 
 /**
+ * Passes back the session ID associated with this stream.
+ *
+ * The session ID can be used to associate a stream with effects processors.
+ * The effects are controlled using the Android AudioEffect Java API.
+ *
+ * If AAudioStreamBuilder_setSessionId() was called with AAUDIO_SESSION_ID_ALLOCATE
+ * then a new session ID should be allocated once when the stream is opened.
+ *
+ * If AAudioStreamBuilder_setSessionId() was called with a previously allocated
+ * session ID then that value should be returned.
+ *
+ * If AAudioStreamBuilder_setSessionId() was not called then this function should
+ * return AAUDIO_SESSION_ID_NONE.
+ *
+ * The sessionID for a stream should not change once the stream has been opened.
+ *
+ * Added in API level 28.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @return session ID or AAUDIO_SESSION_ID_NONE
+ */
+AAUDIO_API aaudio_session_id_t AAudioStream_getSessionId(AAudioStream* stream);
+
+/**
  * Passes back the time at which a particular frame was presented.
  * This can be used to synchronize audio with video or MIDI.
  * It can also be used to align a recorded stream with a playback stream.
@@ -820,6 +1242,36 @@
                                       int64_t *framePosition,
                                       int64_t *timeNanoseconds);
 
+/**
+ * Return the use case for the stream.
+ *
+ * Added in API level 28.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @return frames read
+ */
+AAUDIO_API aaudio_usage_t AAudioStream_getUsage(AAudioStream* stream);
+
+/**
+ * Return the content type for the stream.
+ *
+ * Added in API level 28.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @return content type, for example AAUDIO_CONTENT_TYPE_MUSIC
+ */
+AAUDIO_API aaudio_content_type_t AAudioStream_getContentType(AAudioStream* stream);
+
+/**
+ * Return the input preset for the stream.
+ *
+ * Added in API level 28.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @return input preset, for example AAUDIO_INPUT_PRESET_CAMCORDER
+ */
+AAUDIO_API aaudio_input_preset_t AAudioStream_getInputPreset(AAudioStream* stream);
+
 #ifdef __cplusplus
 }
 #endif

diff --git a/media/libaaudio/libaaudio.map.txt b/media/libaaudio/libaaudio.map.txt
index 2ba5250..cbf5921 100644
--- a/media/libaaudio/libaaudio.map.txt
+++ b/media/libaaudio/libaaudio.map.txt

@@ -17,6 +17,10 @@
     AAudioStreamBuilder_setSharingMode;
     AAudioStreamBuilder_setDirection;
     AAudioStreamBuilder_setBufferCapacityInFrames;
+    AAudioStreamBuilder_setUsage;       # introduced=28
+    AAudioStreamBuilder_setContentType; # introduced=28
+    AAudioStreamBuilder_setInputPreset; # introduced=28
+    AAudioStreamBuilder_setSessionId;   # introduced=28
     AAudioStreamBuilder_openStream;
     AAudioStreamBuilder_delete;
     AAudioStream_close;
@@ -42,8 +46,12 @@
     AAudioStream_getFormat;
     AAudioStream_getSharingMode;
     AAudioStream_getDirection;
+    AAudioStream_getUsage;       # introduced=28
+    AAudioStream_getContentType; # introduced=28
+    AAudioStream_getInputPreset; # introduced=28
     AAudioStream_getFramesWritten;
     AAudioStream_getFramesRead;
+    AAudioStream_getSessionId;   # introduced=28
     AAudioStream_getTimestamp;
     AAudioStream_isMMapUsed;
   local:

diff --git a/media/libaaudio/src/Android.bp b/media/libaaudio/src/Android.bp
new file mode 100644
index 0000000..b9e28a0
--- /dev/null
+++ b/media/libaaudio/src/Android.bp

@@ -0,0 +1,67 @@
+cc_library {
+    name: "libaaudio",
+
+    local_include_dirs: [
+        "binding",
+        "client",
+        "core",
+        "fifo",
+        "legacy",
+        "utility",
+    ],
+    export_include_dirs: ["."],
+    header_libs: ["libaaudio_headers"],
+    export_header_lib_headers: ["libaaudio_headers"],
+
+    srcs: [
+        "core/AudioStream.cpp",
+        "core/AudioStreamBuilder.cpp",
+        "core/AAudioAudio.cpp",
+        "core/AAudioStreamParameters.cpp",
+        "legacy/AudioStreamLegacy.cpp",
+        "legacy/AudioStreamRecord.cpp",
+        "legacy/AudioStreamTrack.cpp",
+        "utility/AAudioUtilities.cpp",
+        "utility/FixedBlockAdapter.cpp",
+        "utility/FixedBlockReader.cpp",
+        "utility/FixedBlockWriter.cpp",
+        "utility/LinearRamp.cpp",
+        "fifo/FifoBuffer.cpp",
+        "fifo/FifoControllerBase.cpp",
+        "client/AudioEndpoint.cpp",
+        "client/AudioStreamInternal.cpp",
+        "client/AudioStreamInternalCapture.cpp",
+        "client/AudioStreamInternalPlay.cpp",
+        "client/IsochronousClockModel.cpp",
+        "binding/AudioEndpointParcelable.cpp",
+        "binding/AAudioBinderClient.cpp",
+        "binding/AAudioStreamRequest.cpp",
+        "binding/AAudioStreamConfiguration.cpp",
+        "binding/IAAudioClient.cpp",
+        "binding/IAAudioService.cpp",
+        "binding/RingBufferParcelable.cpp",
+        "binding/SharedMemoryParcelable.cpp",
+        "binding/SharedRegionParcelable.cpp",
+    ],
+
+    cflags: [
+        "-Wno-unused-parameter",
+        "-Wall",
+        "-Werror",
+
+        // By default, all symbols are hidden.
+        // "-fvisibility=hidden",
+        // AAUDIO_API is used to explicitly export a function or a variable as a visible symbol.
+        "-DAAUDIO_API=__attribute__((visibility(\"default\")))",
+    ],
+
+    shared_libs: [
+        "libaudioclient",
+        "libaudioutils",
+        "liblog",
+        "libcutils",
+        "libutils",
+        "libbinder",
+        "libaudiomanager",
+    ],
+}

diff --git a/media/libaaudio/src/Android.mk b/media/libaaudio/src/Android.mk
deleted file mode 100644
index 6861248..0000000
--- a/media/libaaudio/src/Android.mk
+++ /dev/null

@@ -1,128 +0,0 @@
-LOCAL_PATH:= $(call my-dir)
-
-# ======================= STATIC LIBRARY ==========================
-# This is being built because it make AAudio testing very easy with a complete executable.
-# TODO Remove this target later, when not needed.
-include $(CLEAR_VARS)
-
-LOCAL_MODULE := libaaudio
-LOCAL_MODULE_TAGS := optional
-
-LIBAAUDIO_DIR := $(TOP)/frameworks/av/media/libaaudio
-LIBAAUDIO_SRC_DIR := $(LIBAAUDIO_DIR)/src
-
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/native/include \
-    system/core/base/include \
-    frameworks/native/media/libaaudio/include/include \
-    frameworks/av/media/libaaudio/include \
-    frameworks/native/include \
-    frameworks/av/media/libaudioclient/include \
-    $(LOCAL_PATH) \
-    $(LOCAL_PATH)/binding \
-    $(LOCAL_PATH)/client \
-    $(LOCAL_PATH)/core \
-    $(LOCAL_PATH)/fifo \
-    $(LOCAL_PATH)/legacy \
-    $(LOCAL_PATH)/utility
-
-# If you add a file here then also add it below in the SHARED target
-LOCAL_SRC_FILES = \
-    core/AudioStream.cpp \
-    core/AudioStreamBuilder.cpp \
-    core/AAudioAudio.cpp \
-    core/AAudioStreamParameters.cpp \
-    legacy/AudioStreamLegacy.cpp \
-    legacy/AudioStreamRecord.cpp \
-    legacy/AudioStreamTrack.cpp \
-    utility/AAudioUtilities.cpp \
-    utility/FixedBlockAdapter.cpp \
-    utility/FixedBlockReader.cpp \
-    utility/FixedBlockWriter.cpp \
-    utility/LinearRamp.cpp \
-    fifo/FifoBuffer.cpp \
-    fifo/FifoControllerBase.cpp \
-    client/AudioEndpoint.cpp \
-    client/AudioStreamInternal.cpp \
-    client/AudioStreamInternalCapture.cpp \
-    client/AudioStreamInternalPlay.cpp \
-    client/IsochronousClockModel.cpp \
-    binding/AudioEndpointParcelable.cpp \
-    binding/AAudioBinderClient.cpp \
-    binding/AAudioStreamRequest.cpp \
-    binding/AAudioStreamConfiguration.cpp \
-    binding/IAAudioClient.cpp \
-    binding/IAAudioService.cpp \
-    binding/RingBufferParcelable.cpp \
-    binding/SharedMemoryParcelable.cpp \
-    binding/SharedRegionParcelable.cpp
-
-LOCAL_CFLAGS += -Wno-unused-parameter -Wall -Werror
-
-# By default, all symbols are hidden.
-# LOCAL_CFLAGS += -fvisibility=hidden
-# AAUDIO_API is used to explicitly export a function or a variable as a visible symbol.
-LOCAL_CFLAGS += -DAAUDIO_API='__attribute__((visibility("default")))'
-
-include $(BUILD_STATIC_LIBRARY)
-
-# ======================= SHARED LIBRARY ==========================
-include $(CLEAR_VARS)
-
-LOCAL_MODULE := libaaudio
-LOCAL_MODULE_TAGS := optional
-
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/native/include \
-    system/core/base/include \
-    frameworks/native/media/libaaudio/include/include \
-    frameworks/av/media/libaaudio/include \
-    $(LOCAL_PATH) \
-    $(LOCAL_PATH)/binding \
-    $(LOCAL_PATH)/client \
-    $(LOCAL_PATH)/core \
-    $(LOCAL_PATH)/fifo \
-    $(LOCAL_PATH)/legacy \
-    $(LOCAL_PATH)/utility
-
-LOCAL_SRC_FILES = core/AudioStream.cpp \
-    core/AudioStreamBuilder.cpp \
-    core/AAudioAudio.cpp \
-    core/AAudioStreamParameters.cpp \
-    legacy/AudioStreamLegacy.cpp \
-    legacy/AudioStreamRecord.cpp \
-    legacy/AudioStreamTrack.cpp \
-    utility/AAudioUtilities.cpp \
-    utility/FixedBlockAdapter.cpp \
-    utility/FixedBlockReader.cpp \
-    utility/FixedBlockWriter.cpp \
-    utility/LinearRamp.cpp \
-    fifo/FifoBuffer.cpp \
-    fifo/FifoControllerBase.cpp \
-    client/AudioEndpoint.cpp \
-    client/AudioStreamInternal.cpp \
-    client/AudioStreamInternalCapture.cpp \
-    client/AudioStreamInternalPlay.cpp \
-    client/IsochronousClockModel.cpp \
-    binding/AudioEndpointParcelable.cpp \
-    binding/AAudioBinderClient.cpp \
-    binding/AAudioStreamRequest.cpp \
-    binding/AAudioStreamConfiguration.cpp \
-    binding/IAAudioClient.cpp \
-    binding/IAAudioService.cpp \
-    binding/RingBufferParcelable.cpp \
-    binding/SharedMemoryParcelable.cpp \
-    binding/SharedRegionParcelable.cpp
-
-LOCAL_CFLAGS += -Wno-unused-parameter -Wall -Werror
-
-# By default, all symbols are hidden.
-# LOCAL_CFLAGS += -fvisibility=hidden
-# AAUDIO_API is used to explicitly export a function or a variable as a visible symbol.
-LOCAL_CFLAGS += -DAAUDIO_API='__attribute__((visibility("default")))'
-
-LOCAL_SHARED_LIBRARIES := libaudioclient liblog libcutils libutils libbinder libaudiomanager
-
-include $(BUILD_SHARED_LIBRARY)

diff --git a/media/libaaudio/src/binding/AAudioBinderClient.cpp b/media/libaaudio/src/binding/AAudioBinderClient.cpp
index 07ee2de..dd620e3 100644
--- a/media/libaaudio/src/binding/AAudioBinderClient.cpp
+++ b/media/libaaudio/src/binding/AAudioBinderClient.cpp

@@ -15,7 +15,7 @@
  */
 
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG "AAudioBinderClient"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -61,11 +61,11 @@
         , Singleton<AAudioBinderClient>() {
     gKeepBinderClient = this; // so this singleton won't get deleted
     mAAudioClient = new AAudioClient(this);
-    ALOGV("AAudioBinderClient() this = %p, created mAAudioClient = %p", this, mAAudioClient.get());
+    ALOGV("%s - this = %p, created mAAudioClient = %p", __func__, this, mAAudioClient.get());
 }
 
 AAudioBinderClient::~AAudioBinderClient() {
-    ALOGV("AAudioBinderClient()::~AAudioBinderClient() destroying %p", this);
+    ALOGV("%s - destroying %p", __func__, this);
     Mutex::Autolock _l(mServiceLock);
     if (mAAudioService != 0) {
         IInterface::asBinder(mAAudioService)->unlinkToDeath(mAAudioClient);
@@ -137,7 +137,7 @@
         stream = service->openStream(request, configurationOutput);
 
         if (stream == AAUDIO_ERROR_NO_SERVICE) {
-            ALOGE("AAudioBinderClient::openStream lost connection to AAudioService.");
+            ALOGE("openStream lost connection to AAudioService.");
             dropAAudioService(); // force a reconnect
         } else {
             break;

diff --git a/media/libaaudio/src/binding/AAudioServiceMessage.h b/media/libaaudio/src/binding/AAudioServiceMessage.h
index 54e8001..3981454 100644
--- a/media/libaaudio/src/binding/AAudioServiceMessage.h
+++ b/media/libaaudio/src/binding/AAudioServiceMessage.h

@@ -36,15 +36,17 @@
     AAUDIO_SERVICE_EVENT_PAUSED,
     AAUDIO_SERVICE_EVENT_STOPPED,
     AAUDIO_SERVICE_EVENT_FLUSHED,
-    AAUDIO_SERVICE_EVENT_CLOSED,
     AAUDIO_SERVICE_EVENT_DISCONNECTED,
-    AAUDIO_SERVICE_EVENT_VOLUME
+    AAUDIO_SERVICE_EVENT_VOLUME,
+    AAUDIO_SERVICE_EVENT_XRUN
 } aaudio_service_event_t;
 
 struct AAudioMessageEvent {
     aaudio_service_event_t event;
-    double                 dataDouble;
-    int64_t                dataLong;
+    union {
+        double  dataDouble;
+        int64_t dataLong;
+    };
 };
 
 typedef struct AAudioServiceMessage_s {

diff --git a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
index 153fce3..959db61 100644
--- a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp

@@ -50,6 +50,14 @@
     if (status != NO_ERROR) goto error;
     status = parcel->writeInt32(getBufferCapacity());
     if (status != NO_ERROR) goto error;
+    status = parcel->writeInt32((int32_t) getUsage());
+    if (status != NO_ERROR) goto error;
+    status = parcel->writeInt32((int32_t) getContentType());
+    if (status != NO_ERROR) goto error;
+    status = parcel->writeInt32((int32_t) getInputPreset());
+    if (status != NO_ERROR) goto error;
+    status = parcel->writeInt32(getSessionId());
+    if (status != NO_ERROR) goto error;
     return NO_ERROR;
 error:
     ALOGE("AAudioStreamConfiguration.writeToParcel(): write failed = %d", status);
@@ -69,16 +77,28 @@
     setSamplesPerFrame(value);
     status = parcel->readInt32(&value);
     if (status != NO_ERROR) goto error;
-    setSharingMode(value);
+    setSharingMode((aaudio_sharing_mode_t) value);
     status = parcel->readInt32(&value);
     if (status != NO_ERROR) goto error;
-    setFormat(value);
+    setFormat((aaudio_format_t) value);
     status = parcel->readInt32(&value);
     if (status != NO_ERROR) goto error;
     setDirection((aaudio_direction_t) value);
     status = parcel->readInt32(&value);
     if (status != NO_ERROR) goto error;
     setBufferCapacity(value);
+    status = parcel->readInt32(&value);
+    if (status != NO_ERROR) goto error;
+    setUsage((aaudio_usage_t) value);
+    status = parcel->readInt32(&value);
+    if (status != NO_ERROR) goto error;
+    setContentType((aaudio_content_type_t) value);
+    status = parcel->readInt32(&value);
+    if (status != NO_ERROR) goto error;
+    setInputPreset((aaudio_input_preset_t) value);
+    status = parcel->readInt32(&value);
+    if (status != NO_ERROR) goto error;
+    setSessionId(value);
     return NO_ERROR;
 error:
     ALOGE("AAudioStreamConfiguration.readFromParcel(): read failed = %d", status);

diff --git a/media/libaaudio/src/binding/AAudioStreamRequest.cpp b/media/libaaudio/src/binding/AAudioStreamRequest.cpp
index 1200ab2..c30c5b9 100644
--- a/media/libaaudio/src/binding/AAudioStreamRequest.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamRequest.cpp

@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG "AAudioStreamRequest"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -58,7 +58,7 @@
     return NO_ERROR;
 
 error:
-    ALOGE("AAudioStreamRequest.writeToParcel(): write failed = %d", status);
+    ALOGE("writeToParcel(): write failed = %d", status);
     return status;
 }
 
@@ -80,7 +80,7 @@
     return NO_ERROR;
 
 error:
-    ALOGE("AAudioStreamRequest.readFromParcel(): read failed = %d", status);
+    ALOGE("readFromParcel(): read failed = %d", status);
     return status;
 }
 
@@ -89,9 +89,9 @@
 }
 
 void AAudioStreamRequest::dump() const {
-    ALOGD("AAudioStreamRequest mUserId    = %d", mUserId);
-    ALOGD("AAudioStreamRequest mProcessId = %d", mProcessId);
-    ALOGD("AAudioStreamRequest mSharingModeMatchRequired = %d", mSharingModeMatchRequired);
-    ALOGD("AAudioStreamRequest mInService = %d", mInService);
+    ALOGD("mUserId    = %d", mUserId);
+    ALOGD("mProcessId = %d", mProcessId);
+    ALOGD("mSharingModeMatchRequired = %d", mSharingModeMatchRequired);
+    ALOGD("mInService = %d", mInService);
     mConfiguration.dump();
 }

diff --git a/media/libaaudio/src/binding/AudioEndpointParcelable.cpp b/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
index 1a97555..61d7d27 100644
--- a/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
+++ b/media/libaaudio/src/binding/AudioEndpointParcelable.cpp

@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG "AudioEndpointParcelable"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -64,27 +64,54 @@
  * The read and write must be symmetric.
  */
 status_t AudioEndpointParcelable::writeToParcel(Parcel* parcel) const {
-    parcel->writeInt32(mNumSharedMemories);
+    status_t status = AAudioConvert_aaudioToAndroidStatus(validate());
+    if (status != NO_ERROR) goto error;
+
+    status = parcel->writeInt32(mNumSharedMemories);
+    if (status != NO_ERROR) goto error;
+
     for (int i = 0; i < mNumSharedMemories; i++) {
-        mSharedMemories[i].writeToParcel(parcel);
+        status = mSharedMemories[i].writeToParcel(parcel);
+        if (status != NO_ERROR) goto error;
     }
-    mUpMessageQueueParcelable.writeToParcel(parcel);
-    mDownMessageQueueParcelable.writeToParcel(parcel);
-    mUpDataQueueParcelable.writeToParcel(parcel);
-    mDownDataQueueParcelable.writeToParcel(parcel);
-    return NO_ERROR; // TODO check for errors above
+    status = mUpMessageQueueParcelable.writeToParcel(parcel);
+    if (status != NO_ERROR) goto error;
+    status = mDownMessageQueueParcelable.writeToParcel(parcel);
+    if (status != NO_ERROR) goto error;
+    status = mUpDataQueueParcelable.writeToParcel(parcel);
+    if (status != NO_ERROR) goto error;
+    status = mDownDataQueueParcelable.writeToParcel(parcel);
+    if (status != NO_ERROR) goto error;
+
+    return NO_ERROR;
+
+error:
+    ALOGE("%s returning %d", __func__, status);
+    return status;
 }
 
 status_t AudioEndpointParcelable::readFromParcel(const Parcel* parcel) {
-    parcel->readInt32(&mNumSharedMemories);
+    status_t status = parcel->readInt32(&mNumSharedMemories);
+    if (status != NO_ERROR) goto error;
+
     for (int i = 0; i < mNumSharedMemories; i++) {
         mSharedMemories[i].readFromParcel(parcel);
+        if (status != NO_ERROR) goto error;
     }
-    mUpMessageQueueParcelable.readFromParcel(parcel);
-    mDownMessageQueueParcelable.readFromParcel(parcel);
-    mUpDataQueueParcelable.readFromParcel(parcel);
-    mDownDataQueueParcelable.readFromParcel(parcel);
-    return NO_ERROR; // TODO check for errors above
+    status = mUpMessageQueueParcelable.readFromParcel(parcel);
+    if (status != NO_ERROR) goto error;
+    status = mDownMessageQueueParcelable.readFromParcel(parcel);
+    if (status != NO_ERROR) goto error;
+    status = mUpDataQueueParcelable.readFromParcel(parcel);
+    if (status != NO_ERROR) goto error;
+    status = mDownDataQueueParcelable.readFromParcel(parcel);
+    if (status != NO_ERROR) goto error;
+
+    return AAudioConvert_aaudioToAndroidStatus(validate());
+
+error:
+    ALOGE("%s returning %d", __func__, status);
+    return status;
 }
 
 aaudio_result_t AudioEndpointParcelable::resolve(EndpointDescriptor *descriptor) {
@@ -109,52 +136,28 @@
     return AAudioConvert_androidToAAudioResult(err);
 }
 
-aaudio_result_t AudioEndpointParcelable::validate() {
-    aaudio_result_t result;
+aaudio_result_t AudioEndpointParcelable::validate() const {
     if (mNumSharedMemories < 0 || mNumSharedMemories >= MAX_SHARED_MEMORIES) {
-        ALOGE("AudioEndpointParcelable invalid mNumSharedMemories = %d", mNumSharedMemories);
+        ALOGE("invalid mNumSharedMemories = %d", mNumSharedMemories);
         return AAUDIO_ERROR_INTERNAL;
     }
-    for (int i = 0; i < mNumSharedMemories; i++) {
-        result = mSharedMemories[i].validate();
-        if (result != AAUDIO_OK) {
-            ALOGE("AudioEndpointParcelable invalid mSharedMemories[%d] = %d", i, result);
-            return result;
-        }
-    }
-    if ((result = mUpMessageQueueParcelable.validate()) != AAUDIO_OK) {
-        ALOGE("AudioEndpointParcelable invalid mUpMessageQueueParcelable = %d", result);
-        return result;
-    }
-    if ((result = mDownMessageQueueParcelable.validate()) != AAUDIO_OK) {
-        ALOGE("AudioEndpointParcelable invalid mDownMessageQueueParcelable = %d", result);
-        return result;
-    }
-    if ((result = mUpDataQueueParcelable.validate()) != AAUDIO_OK) {
-        ALOGE("AudioEndpointParcelable invalid mUpDataQueueParcelable = %d", result);
-        return result;
-    }
-    if ((result = mDownDataQueueParcelable.validate()) != AAUDIO_OK) {
-        ALOGE("AudioEndpointParcelable invalid mDownDataQueueParcelable = %d", result);
-        return result;
-    }
     return AAUDIO_OK;
 }
 
 void AudioEndpointParcelable::dump() {
-    ALOGD("AudioEndpointParcelable ======================================= BEGIN");
-    ALOGD("AudioEndpointParcelable mNumSharedMemories = %d", mNumSharedMemories);
+    ALOGD("======================================= BEGIN");
+    ALOGD("mNumSharedMemories = %d", mNumSharedMemories);
     for (int i = 0; i < mNumSharedMemories; i++) {
         mSharedMemories[i].dump();
     }
-    ALOGD("AudioEndpointParcelable mUpMessageQueueParcelable =========");
+    ALOGD("mUpMessageQueueParcelable =========");
     mUpMessageQueueParcelable.dump();
-    ALOGD("AudioEndpointParcelable mDownMessageQueueParcelable =======");
+    ALOGD("mDownMessageQueueParcelable =======");
     mDownMessageQueueParcelable.dump();
-    ALOGD("AudioEndpointParcelable mUpDataQueueParcelable ============");
+    ALOGD("mUpDataQueueParcelable ============");
     mUpDataQueueParcelable.dump();
-    ALOGD("AudioEndpointParcelable mDownDataQueueParcelable ==========");
+    ALOGD("mDownDataQueueParcelable ==========");
     mDownDataQueueParcelable.dump();
-    ALOGD("AudioEndpointParcelable ======================================= END");
+    ALOGD("======================================= END");
 }
 

diff --git a/media/libaaudio/src/binding/AudioEndpointParcelable.h b/media/libaaudio/src/binding/AudioEndpointParcelable.h
index aa8573f..e4f8b9e 100644
--- a/media/libaaudio/src/binding/AudioEndpointParcelable.h
+++ b/media/libaaudio/src/binding/AudioEndpointParcelable.h

@@ -56,8 +56,6 @@
 
     aaudio_result_t resolve(EndpointDescriptor *descriptor);
 
-    aaudio_result_t validate();
-
     aaudio_result_t close();
 
     void dump();
@@ -70,6 +68,8 @@
     RingBufferParcelable    mDownDataQueueParcelable;    // eg. playback
 
 private:
+    aaudio_result_t         validate() const;
+
     int32_t                 mNumSharedMemories = 0;
     SharedMemoryParcelable  mSharedMemories[MAX_SHARED_MEMORIES];
 };

diff --git a/media/libaaudio/src/binding/IAAudioService.cpp b/media/libaaudio/src/binding/IAAudioService.cpp
index b3c4934..620edc7 100644
--- a/media/libaaudio/src/binding/IAAudioService.cpp
+++ b/media/libaaudio/src/binding/IAAudioService.cpp

@@ -121,17 +121,11 @@
             ALOGE("BpAAudioService::client GET_STREAM_DESCRIPTION passed result %d", result);
             return result;
         }
-        err = parcelable.readFromParcel(&reply);;
+        err = parcelable.readFromParcel(&reply);
         if (err != NO_ERROR) {
             ALOGE("BpAAudioService::client transact(GET_STREAM_DESCRIPTION) read endpoint %d", err);
             return AAudioConvert_androidToAAudioResult(err);
         }
-        //parcelable.dump();
-        result = parcelable.validate();
-        if (result != AAUDIO_OK) {
-            ALOGE("BpAAudioService::client GET_STREAM_DESCRIPTION validation fails %d", result);
-            return result;
-        }
         return result;
     }
 
@@ -250,6 +244,7 @@
     pid_t tid;
     int64_t nanoseconds;
     aaudio_result_t result;
+    status_t status = NO_ERROR;
     ALOGV("BnAAudioService::onTransact(%i) %i", code, flags);
 
     switch(code) {
@@ -294,21 +289,20 @@
 
         case GET_STREAM_DESCRIPTION: {
             CHECK_INTERFACE(IAAudioService, data, reply);
-            data.readInt32(&streamHandle);
+            status = data.readInt32(&streamHandle);
+            if (status != NO_ERROR) {
+                return status;
+            }
             aaudio::AudioEndpointParcelable parcelable;
             result = getStreamDescription(streamHandle, parcelable);
             if (result != AAUDIO_OK) {
                 return AAudioConvert_aaudioToAndroidStatus(result);
             }
-            result = parcelable.validate();
-            if (result != AAUDIO_OK) {
-                ALOGE("BnAAudioService::onTransact getStreamDescription() returns %d", result);
-                parcelable.dump();
-                return AAudioConvert_aaudioToAndroidStatus(result);
+            status = reply->writeInt32(result);
+            if (status != NO_ERROR) {
+                return status;
             }
-            reply->writeInt32(result);
-            parcelable.writeToParcel(reply);
-            return NO_ERROR;
+            return parcelable.writeToParcel(reply);
         } break;
 
         case START_STREAM: {

diff --git a/media/libaaudio/src/binding/RingBufferParcelable.cpp b/media/libaaudio/src/binding/RingBufferParcelable.cpp
index 6b74b21..4996b3f 100644
--- a/media/libaaudio/src/binding/RingBufferParcelable.cpp
+++ b/media/libaaudio/src/binding/RingBufferParcelable.cpp

@@ -14,13 +14,14 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG "RingBufferParcelable"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
 #include <stdint.h>
 
 #include <binder/Parcelable.h>
+#include <utility/AAudioUtilities.h>
 
 #include "binding/AAudioServiceDefinitions.h"
 #include "binding/SharedRegionParcelable.h"
@@ -79,7 +80,10 @@
  * The read and write must be symmetric.
  */
 status_t RingBufferParcelable::writeToParcel(Parcel* parcel) const {
-    status_t status = parcel->writeInt32(mCapacityInFrames);
+    status_t status = AAudioConvert_aaudioToAndroidStatus(validate());
+    if (status != NO_ERROR) goto error;
+
+    status = parcel->writeInt32(mCapacityInFrames);
     if (status != NO_ERROR) goto error;
     if (mCapacityInFrames > 0) {
         status = parcel->writeInt32(mBytesPerFrame);
@@ -97,7 +101,7 @@
     }
     return NO_ERROR;
 error:
-    ALOGE("RingBufferParcelable::writeToParcel() error = %d", status);
+    ALOGE("%s returning %d", __func__, status);
     return status;
 }
 
@@ -118,9 +122,9 @@
         status = mDataParcelable.readFromParcel(parcel);
         if (status != NO_ERROR) goto error;
     }
-    return NO_ERROR;
+    return AAudioConvert_aaudioToAndroidStatus(validate());
 error:
-    ALOGE("RingBufferParcelable::readFromParcel() error = %d", status);
+    ALOGE("%s returning %d", __func__, status);
     return status;
 }
 
@@ -151,42 +155,29 @@
     return AAUDIO_OK;
 }
 
-aaudio_result_t RingBufferParcelable::validate() {
-    aaudio_result_t result;
+aaudio_result_t RingBufferParcelable::validate() const {
     if (mCapacityInFrames < 0 || mCapacityInFrames >= 32 * 1024) {
-        ALOGE("RingBufferParcelable invalid mCapacityInFrames = %d", mCapacityInFrames);
+        ALOGE("invalid mCapacityInFrames = %d", mCapacityInFrames);
         return AAUDIO_ERROR_INTERNAL;
     }
     if (mBytesPerFrame < 0 || mBytesPerFrame >= 256) {
-        ALOGE("RingBufferParcelable invalid mBytesPerFrame = %d", mBytesPerFrame);
+        ALOGE("invalid mBytesPerFrame = %d", mBytesPerFrame);
         return AAUDIO_ERROR_INTERNAL;
     }
     if (mFramesPerBurst < 0 || mFramesPerBurst >= 16 * 1024) {
-        ALOGE("RingBufferParcelable invalid mFramesPerBurst = %d", mFramesPerBurst);
+        ALOGE("invalid mFramesPerBurst = %d", mFramesPerBurst);
         return AAUDIO_ERROR_INTERNAL;
     }
-    if ((result = mReadCounterParcelable.validate()) != AAUDIO_OK) {
-        ALOGE("RingBufferParcelable invalid mReadCounterParcelable = %d", result);
-        return result;
-    }
-    if ((result = mWriteCounterParcelable.validate()) != AAUDIO_OK) {
-        ALOGE("RingBufferParcelable invalid mWriteCounterParcelable = %d", result);
-        return result;
-    }
-    if ((result = mDataParcelable.validate()) != AAUDIO_OK) {
-        ALOGE("RingBufferParcelable invalid mDataParcelable = %d", result);
-        return result;
-    }
     return AAUDIO_OK;
 }
 
 
 void RingBufferParcelable::dump() {
-    ALOGD("RingBufferParcelable mCapacityInFrames = %d ---------", mCapacityInFrames);
+    ALOGD("mCapacityInFrames = %d ---------", mCapacityInFrames);
     if (mCapacityInFrames > 0) {
-        ALOGD("RingBufferParcelable mBytesPerFrame = %d", mBytesPerFrame);
-        ALOGD("RingBufferParcelable mFramesPerBurst = %d", mFramesPerBurst);
-        ALOGD("RingBufferParcelable mFlags = %u", mFlags);
+        ALOGD("mBytesPerFrame = %d", mBytesPerFrame);
+        ALOGD("mFramesPerBurst = %d", mFramesPerBurst);
+        ALOGD("mFlags = %u", mFlags);
         mReadCounterParcelable.dump();
         mWriteCounterParcelable.dump();
         mDataParcelable.dump();

diff --git a/media/libaaudio/src/binding/RingBufferParcelable.h b/media/libaaudio/src/binding/RingBufferParcelable.h
index bd562f2..1dbcf07 100644
--- a/media/libaaudio/src/binding/RingBufferParcelable.h
+++ b/media/libaaudio/src/binding/RingBufferParcelable.h

@@ -66,11 +66,12 @@
 
     aaudio_result_t resolve(SharedMemoryParcelable *memoryParcels, RingBufferDescriptor *descriptor);
 
-    aaudio_result_t validate();
-
     void dump();
 
 private:
+
+    aaudio_result_t validate() const;
+
     SharedRegionParcelable  mReadCounterParcelable;
     SharedRegionParcelable  mWriteCounterParcelable;
     SharedRegionParcelable  mDataParcelable;

diff --git a/media/libaaudio/src/binding/SharedMemoryParcelable.cpp b/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
index 90217ab..0b0cf77 100644
--- a/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
+++ b/media/libaaudio/src/binding/SharedMemoryParcelable.cpp

@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG "SharedMemoryParcelable"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -43,16 +43,18 @@
 
 void SharedMemoryParcelable::setup(const unique_fd& fd, int32_t sizeInBytes) {
     mFd.reset(dup(fd.get())); // store a duplicate fd
-    ALOGV("SharedMemoryParcelable::setup(%d -> %d, %d) this = %p\n",
-          fd.get(), mFd.get(), sizeInBytes, this);
+    ALOGV("setup(%d -> %d, %d) this = %p\n", fd.get(), mFd.get(), sizeInBytes, this);
     mSizeInBytes = sizeInBytes;
 }
 
 status_t SharedMemoryParcelable::writeToParcel(Parcel* parcel) const {
-    status_t status = parcel->writeInt32(mSizeInBytes);
+    status_t status = AAudioConvert_aaudioToAndroidStatus(validate());
+    if (status != NO_ERROR) return status;
+
+    status = parcel->writeInt32(mSizeInBytes);
     if (status != NO_ERROR) return status;
     if (mSizeInBytes > 0) {
-        ALOGV("SharedMemoryParcelable::writeToParcel() mFd = %d, this = %p\n", mFd.get(), this);
+        ALOGV("writeToParcel() mFd = %d, this = %p\n", mFd.get(), this);
         status = parcel->writeUniqueFileDescriptor(mFd);
         ALOGE_IF(status != NO_ERROR, "SharedMemoryParcelable writeDupFileDescriptor failed : %d",
                  status);
@@ -62,22 +64,27 @@
 
 status_t SharedMemoryParcelable::readFromParcel(const Parcel* parcel) {
     status_t status = parcel->readInt32(&mSizeInBytes);
-    if (status != NO_ERROR) {
-        return status;
-    }
+    if (status != NO_ERROR) goto error;
+
     if (mSizeInBytes > 0) {
         // The Parcel owns the file descriptor and will close it later.
         unique_fd mmapFd;
         status = parcel->readUniqueFileDescriptor(&mmapFd);
         if (status != NO_ERROR) {
-            ALOGE("SharedMemoryParcelable::readFromParcel() readUniqueFileDescriptor() failed : %d",
-                  status);
-        } else {
-            // Resolve the memory now while we still have the FD from the Parcel.
-            // Closing the FD will not affect the shared memory once mmap() has been called.
-            status = AAudioConvert_androidToAAudioResult(resolveSharedMemory(mmapFd));
+            ALOGE("readFromParcel() readUniqueFileDescriptor() failed : %d", status);
+            goto error;
         }
+
+        // Resolve the memory now while we still have the FD from the Parcel.
+        // Closing the FD will not affect the shared memory once mmap() has been called.
+        aaudio_result_t result = resolveSharedMemory(mmapFd);
+        status = AAudioConvert_aaudioToAndroidStatus(result);
+        if (status != NO_ERROR) goto error;
     }
+
+    return AAudioConvert_aaudioToAndroidStatus(validate());
+
+error:
     return status;
 }
 
@@ -85,7 +92,7 @@
     if (mResolvedAddress != MMAP_UNRESOLVED_ADDRESS) {
         int err = munmap(mResolvedAddress, mSizeInBytes);
         if (err < 0) {
-            ALOGE("SharedMemoryParcelable::close() munmap() failed %d", err);
+            ALOGE("close() munmap() failed %d", err);
             return AAudioConvert_androidToAAudioResult(err);
         }
         mResolvedAddress = MMAP_UNRESOLVED_ADDRESS;
@@ -97,8 +104,7 @@
     mResolvedAddress = (uint8_t *) mmap(0, mSizeInBytes, PROT_READ | PROT_WRITE,
                                         MAP_SHARED, fd.get(), 0);
     if (mResolvedAddress == MMAP_UNRESOLVED_ADDRESS) {
-        ALOGE("SharedMemoryParcelable mmap() failed for fd = %d, errno = %s",
-              fd.get(), strerror(errno));
+        ALOGE("mmap() failed for fd = %d, errno = %s", fd.get(), strerror(errno));
         return AAUDIO_ERROR_INTERNAL;
     }
     return AAUDIO_OK;
@@ -107,10 +113,10 @@
 aaudio_result_t SharedMemoryParcelable::resolve(int32_t offsetInBytes, int32_t sizeInBytes,
                                               void **regionAddressPtr) {
     if (offsetInBytes < 0) {
-        ALOGE("SharedMemoryParcelable illegal offsetInBytes = %d", offsetInBytes);
+        ALOGE("illegal offsetInBytes = %d", offsetInBytes);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     } else if ((offsetInBytes + sizeInBytes) > mSizeInBytes) {
-        ALOGE("SharedMemoryParcelable out of range, offsetInBytes = %d, "
+        ALOGE("out of range, offsetInBytes = %d, "
                       "sizeInBytes = %d, mSizeInBytes = %d",
               offsetInBytes, sizeInBytes, mSizeInBytes);
         return AAUDIO_ERROR_OUT_OF_RANGE;
@@ -122,16 +128,15 @@
         if (mFd.get() != -1) {
             result = resolveSharedMemory(mFd);
         } else {
-            ALOGE("SharedMemoryParcelable has no file descriptor for shared memory.");
+            ALOGE("has no file descriptor for shared memory.");
             result = AAUDIO_ERROR_INTERNAL;
         }
     }
 
     if (result == AAUDIO_OK && mResolvedAddress != MMAP_UNRESOLVED_ADDRESS) {
         *regionAddressPtr = mResolvedAddress + offsetInBytes;
-        ALOGV("SharedMemoryParcelable mResolvedAddress = %p", mResolvedAddress);
-        ALOGV("SharedMemoryParcelable offset by %d, *regionAddressPtr = %p",
-              offsetInBytes, *regionAddressPtr);
+        ALOGV("mResolvedAddress = %p", mResolvedAddress);
+        ALOGV("offset by %d, *regionAddressPtr = %p", offsetInBytes, *regionAddressPtr);
     }
     return result;
 }
@@ -140,16 +145,16 @@
     return mSizeInBytes;
 }
 
-aaudio_result_t SharedMemoryParcelable::validate() {
+aaudio_result_t SharedMemoryParcelable::validate() const {
     if (mSizeInBytes < 0 || mSizeInBytes >= MAX_MMAP_SIZE_BYTES) {
-        ALOGE("SharedMemoryParcelable invalid mSizeInBytes = %d", mSizeInBytes);
+        ALOGE("invalid mSizeInBytes = %d", mSizeInBytes);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }
     return AAUDIO_OK;
 }
 
 void SharedMemoryParcelable::dump() {
-    ALOGD("SharedMemoryParcelable mFd = %d", mFd.get());
-    ALOGD("SharedMemoryParcelable mSizeInBytes = %d", mSizeInBytes);
-    ALOGD("SharedMemoryParcelable mResolvedAddress = %p", mResolvedAddress);
+    ALOGD("mFd = %d", mFd.get());
+    ALOGD("mSizeInBytes = %d", mSizeInBytes);
+    ALOGD("mResolvedAddress = %p", mResolvedAddress);
 }

diff --git a/media/libaaudio/src/binding/SharedMemoryParcelable.h b/media/libaaudio/src/binding/SharedMemoryParcelable.h
index 2a634e0..82c2240 100644
--- a/media/libaaudio/src/binding/SharedMemoryParcelable.h
+++ b/media/libaaudio/src/binding/SharedMemoryParcelable.h

@@ -61,8 +61,6 @@
 
     int32_t getSizeInBytes();
 
-    aaudio_result_t validate();
-
     void dump();
 
 protected:
@@ -74,6 +72,11 @@
     android::base::unique_fd   mFd;
     int32_t     mSizeInBytes = 0;
     uint8_t    *mResolvedAddress = MMAP_UNRESOLVED_ADDRESS;
+
+private:
+
+    aaudio_result_t validate() const;
+
 };
 
 } /* namespace aaudio */

diff --git a/media/libaaudio/src/binding/SharedRegionParcelable.cpp b/media/libaaudio/src/binding/SharedRegionParcelable.cpp
index 7381dcb..c776116 100644
--- a/media/libaaudio/src/binding/SharedRegionParcelable.cpp
+++ b/media/libaaudio/src/binding/SharedRegionParcelable.cpp

@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG "SharedRegionParcelable"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -24,6 +24,7 @@
 #include <binder/Parcelable.h>
 
 #include <aaudio/AAudio.h>
+#include <utility/AAudioUtilities.h>
 
 #include "binding/SharedMemoryParcelable.h"
 #include "binding/SharedRegionParcelable.h"
@@ -47,21 +48,38 @@
 }
 
 status_t SharedRegionParcelable::writeToParcel(Parcel* parcel) const {
-    parcel->writeInt32(mSizeInBytes);
+    status_t status = AAudioConvert_aaudioToAndroidStatus(validate());
+    if (status != NO_ERROR) goto error;
+
+    status = parcel->writeInt32(mSizeInBytes);
+    if (status != NO_ERROR) goto error;
     if (mSizeInBytes > 0) {
-        parcel->writeInt32(mSharedMemoryIndex);
-        parcel->writeInt32(mOffsetInBytes);
+        status = parcel->writeInt32(mSharedMemoryIndex);
+        if (status != NO_ERROR) goto error;
+        status = parcel->writeInt32(mOffsetInBytes);
+        if (status != NO_ERROR) goto error;
     }
-    return NO_ERROR; // TODO check for errors above
+    return NO_ERROR;
+
+error:
+    ALOGE("%s returning %d", __func__, status);
+    return status;
 }
 
 status_t SharedRegionParcelable::readFromParcel(const Parcel* parcel) {
-    parcel->readInt32(&mSizeInBytes);
+    status_t status = parcel->readInt32(&mSizeInBytes);
+    if (status != NO_ERROR) goto error;
     if (mSizeInBytes > 0) {
-        parcel->readInt32(&mSharedMemoryIndex);
-        parcel->readInt32(&mOffsetInBytes);
+        status = parcel->readInt32(&mSharedMemoryIndex);
+        if (status != NO_ERROR) goto error;
+        status = parcel->readInt32(&mOffsetInBytes);
+        if (status != NO_ERROR) goto error;
     }
-    return NO_ERROR; // TODO check for errors above
+    return AAudioConvert_aaudioToAndroidStatus(validate());
+
+error:
+    ALOGE("%s returning %d", __func__, status);
+    return status;
 }
 
 aaudio_result_t SharedRegionParcelable::resolve(SharedMemoryParcelable *memoryParcels,
@@ -71,25 +89,25 @@
         return AAUDIO_OK;
     }
     if (mSharedMemoryIndex < 0) {
-        ALOGE("SharedRegionParcelable invalid mSharedMemoryIndex = %d", mSharedMemoryIndex);
+        ALOGE("invalid mSharedMemoryIndex = %d", mSharedMemoryIndex);
         return AAUDIO_ERROR_INTERNAL;
     }
     SharedMemoryParcelable *memoryParcel = &memoryParcels[mSharedMemoryIndex];
     return memoryParcel->resolve(mOffsetInBytes, mSizeInBytes, regionAddressPtr);
 }
 
-aaudio_result_t SharedRegionParcelable::validate() {
+aaudio_result_t SharedRegionParcelable::validate() const {
     if (mSizeInBytes < 0 || mSizeInBytes >= MAX_MMAP_SIZE_BYTES) {
-        ALOGE("SharedRegionParcelable invalid mSizeInBytes = %d", mSizeInBytes);
+        ALOGE("invalid mSizeInBytes = %d", mSizeInBytes);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }
     if (mSizeInBytes > 0) {
         if (mOffsetInBytes < 0 || mOffsetInBytes >= MAX_MMAP_OFFSET_BYTES) {
-            ALOGE("SharedRegionParcelable invalid mOffsetInBytes = %d", mOffsetInBytes);
+            ALOGE("invalid mOffsetInBytes = %d", mOffsetInBytes);
             return AAUDIO_ERROR_OUT_OF_RANGE;
         }
         if (mSharedMemoryIndex < 0 || mSharedMemoryIndex >= MAX_SHARED_MEMORIES) {
-            ALOGE("SharedRegionParcelable invalid mSharedMemoryIndex = %d", mSharedMemoryIndex);
+            ALOGE("invalid mSharedMemoryIndex = %d", mSharedMemoryIndex);
             return AAUDIO_ERROR_INTERNAL;
         }
     }
@@ -97,9 +115,9 @@
 }
 
 void SharedRegionParcelable::dump() {
-    ALOGD("SharedRegionParcelable mSizeInBytes = %d -----", mSizeInBytes);
+    ALOGD("mSizeInBytes = %d -----", mSizeInBytes);
     if (mSizeInBytes > 0) {
-        ALOGD("SharedRegionParcelable mSharedMemoryIndex = %d", mSharedMemoryIndex);
-        ALOGD("SharedRegionParcelable mOffsetInBytes = %d", mOffsetInBytes);
+        ALOGD("mSharedMemoryIndex = %d", mSharedMemoryIndex);
+        ALOGD("mOffsetInBytes = %d", mOffsetInBytes);
     }
 }

diff --git a/media/libaaudio/src/binding/SharedRegionParcelable.h b/media/libaaudio/src/binding/SharedRegionParcelable.h
index f6babfd..0cd8c04 100644
--- a/media/libaaudio/src/binding/SharedRegionParcelable.h
+++ b/media/libaaudio/src/binding/SharedRegionParcelable.h

@@ -47,14 +47,15 @@
 
     bool isFileDescriptorSafe(SharedMemoryParcelable *memoryParcels);
 
-    aaudio_result_t validate();
-
     void dump();
 
 protected:
     int32_t mSharedMemoryIndex = -1;
     int32_t mOffsetInBytes     = 0;
     int32_t mSizeInBytes       = 0;
+
+private:
+    aaudio_result_t validate() const;
 };
 
 } /* namespace aaudio */

diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index 604eed5..f8e34d1 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp

@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG "AudioEndpoint"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -45,6 +45,7 @@
     delete mUpCommandQueue;
 }
 
+// TODO Consider moving to a method in RingBufferDescriptor
 static aaudio_result_t AudioEndpoint_validateQueueDescriptor(const char *type,
                                                   const RingBufferDescriptor *descriptor) {
     if (descriptor == nullptr) {
@@ -127,19 +128,19 @@
     // ============================ up message queue =============================
     const RingBufferDescriptor *descriptor = &pEndpointDescriptor->upMessageQueueDescriptor;
     if(descriptor->bytesPerFrame != sizeof(AAudioServiceMessage)) {
-        ALOGE("AudioEndpoint.configure() bytesPerFrame != sizeof(AAudioServiceMessage) = %d",
+        ALOGE("configure() bytesPerFrame != sizeof(AAudioServiceMessage) = %d",
               descriptor->bytesPerFrame);
         return AAUDIO_ERROR_INTERNAL;
     }
 
     if(descriptor->readCounterAddress == nullptr || descriptor->writeCounterAddress == nullptr) {
-        ALOGE("AudioEndpoint.configure() NULL counter address");
+        ALOGE("configure() NULL counter address");
         return AAUDIO_ERROR_NULL;
     }
 
     // Prevent memory leak and reuse.
     if(mUpCommandQueue != nullptr || mDataQueue != nullptr) {
-        ALOGE("AudioEndpoint.configure() endpoint already used");
+        ALOGE("configure() endpoint already used");
         return AAUDIO_ERROR_INTERNAL;
     }
 
@@ -153,8 +154,8 @@
 
     // ============================ data queue =============================
     descriptor = &pEndpointDescriptor->dataQueueDescriptor;
-    ALOGV("AudioEndpoint.configure() data framesPerBurst = %d", descriptor->framesPerBurst);
-    ALOGV("AudioEndpoint.configure() data readCounterAddress = %p",
+    ALOGV("configure() data framesPerBurst = %d", descriptor->framesPerBurst);
+    ALOGV("configure() data readCounterAddress = %p",
           descriptor->readCounterAddress);
 
     // An example of free running is when the other side is read or written by hardware DMA
@@ -163,7 +164,7 @@
                              ? descriptor->readCounterAddress // read by other side
                              : descriptor->writeCounterAddress; // written by other side
     mFreeRunning = (remoteCounter == nullptr);
-    ALOGV("AudioEndpoint.configure() mFreeRunning = %d", mFreeRunning ? 1 : 0);
+    ALOGV("configure() mFreeRunning = %d", mFreeRunning ? 1 : 0);
 
     int64_t *readCounterAddress = (descriptor->readCounterAddress == nullptr)
                                   ? &mDataReadCounter
@@ -258,8 +259,8 @@
 }
 
 void AudioEndpoint::dump() const {
-    ALOGD("AudioEndpoint: data readCounter  = %lld", (long long) mDataQueue->getReadCounter());
-    ALOGD("AudioEndpoint: data writeCounter = %lld", (long long) mDataQueue->getWriteCounter());
+    ALOGD("data readCounter  = %lld", (long long) mDataQueue->getReadCounter());
+    ALOGD("data writeCounter = %lld", (long long) mDataQueue->getWriteCounter());
 }
 
 void AudioEndpoint::eraseDataMemory() {

diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 2fdbfaf..9204824 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp

@@ -61,15 +61,12 @@
         , mClockModel()
         , mAudioEndpoint()
         , mServiceStreamHandle(AAUDIO_HANDLE_INVALID)
-        , mFramesPerBurst(16)
         , mInService(inService)
         , mServiceInterface(serviceInterface)
         , mAtomicTimestamp()
         , mWakeupDelayNanos(AAudioProperty_getWakeupDelayMicros() * AAUDIO_NANOS_PER_MICROSECOND)
         , mMinimumSleepNanos(AAudioProperty_getMinimumSleepMicros() * AAUDIO_NANOS_PER_MICROSECOND)
         {
-    ALOGD("AudioStreamInternal(): mWakeupDelayNanos = %d, mMinimumSleepNanos = %d",
-          mWakeupDelayNanos, mMinimumSleepNanos);
 }
 
 AudioStreamInternal::~AudioStreamInternal() {
@@ -79,11 +76,12 @@
 
     aaudio_result_t result = AAUDIO_OK;
     int32_t capacity;
+    int32_t framesPerBurst;
     AAudioStreamRequest request;
     AAudioStreamConfiguration configurationOutput;
 
     if (getState() != AAUDIO_STREAM_STATE_UNINITIALIZED) {
-        ALOGE("AudioStreamInternal::open(): already open! state = %d", getState());
+        ALOGE("%s - already open! state = %d", __func__, getState());
         return AAUDIO_ERROR_INVALID_STATE;
     }
 
@@ -104,7 +102,7 @@
     request.setUserId(getuid());
     request.setProcessId(getpid());
     request.setSharingModeMatchRequired(isSharingModeMatchRequired());
-    request.setInService(mInService);
+    request.setInService(isInService());
 
     request.getConfiguration().setDeviceId(getDeviceId());
     request.getConfiguration().setSampleRate(getSampleRate());
@@ -112,13 +110,30 @@
     request.getConfiguration().setDirection(getDirection());
     request.getConfiguration().setSharingMode(getSharingMode());
 
+    request.getConfiguration().setUsage(getUsage());
+    request.getConfiguration().setContentType(getContentType());
+    request.getConfiguration().setInputPreset(getInputPreset());
+
     request.getConfiguration().setBufferCapacity(builder.getBufferCapacity());
 
+    mDeviceChannelCount = getSamplesPerFrame(); // Assume it will be the same. Update if not.
+
     mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
+    if (mServiceStreamHandle < 0
+            && request.getConfiguration().getSamplesPerFrame() == 1 // mono?
+            && getDirection() == AAUDIO_DIRECTION_OUTPUT
+            && !isInService()) {
+        // if that failed then try switching from mono to stereo if OUTPUT.
+        // Only do this in the client. Otherwise we end up with a mono mixer in the service
+        // that writes to a stereo MMAP stream.
+        ALOGD("%s - openStream() returned %d, try switching from MONO to STEREO",
+              __func__, mServiceStreamHandle);
+        request.getConfiguration().setSamplesPerFrame(2); // stereo
+        mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
+    }
     if (mServiceStreamHandle < 0) {
-        result = mServiceStreamHandle;
-        ALOGE("AudioStreamInternal::open(): openStream() returned %d", result);
-        return result;
+        ALOGE("%s - openStream() returned %d", __func__, mServiceStreamHandle);
+        return mServiceStreamHandle;
     }
 
     result = configurationOutput.validate();
@@ -126,13 +141,22 @@
         goto error;
     }
     // Save results of the open.
+    if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) {
+        setSamplesPerFrame(configurationOutput.getSamplesPerFrame());
+    }
+    mDeviceChannelCount = configurationOutput.getSamplesPerFrame();
+
     setSampleRate(configurationOutput.getSampleRate());
-    setSamplesPerFrame(configurationOutput.getSamplesPerFrame());
     setDeviceId(configurationOutput.getDeviceId());
+    setSessionId(configurationOutput.getSessionId());
     setSharingMode(configurationOutput.getSharingMode());
 
+    setUsage(configurationOutput.getUsage());
+    setContentType(configurationOutput.getContentType());
+    setInputPreset(configurationOutput.getInputPreset());
+
     // Save device format so we can do format conversion and volume scaling together.
-    mDeviceFormat = configurationOutput.getFormat();
+    setDeviceFormat(configurationOutput.getFormat());
 
     result = mServiceInterface.getStreamDescription(mServiceStreamHandle, mEndPointParcelable);
     if (result != AAUDIO_OK) {
@@ -151,17 +175,18 @@
         goto error;
     }
 
-    mFramesPerBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
-    capacity = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
-
     // Validate result from server.
-    if (mFramesPerBurst < 16 || mFramesPerBurst > 16 * 1024) {
-        ALOGE("AudioStreamInternal::open(): framesPerBurst out of range = %d", mFramesPerBurst);
+    framesPerBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
+    if (framesPerBurst < MIN_FRAMES_PER_BURST || framesPerBurst > MAX_FRAMES_PER_BURST) {
+        ALOGE("%s - framesPerBurst out of range = %d", __func__, framesPerBurst);
         result = AAUDIO_ERROR_OUT_OF_RANGE;
         goto error;
     }
-    if (capacity < mFramesPerBurst || capacity > 32 * 1024) {
-        ALOGE("AudioStreamInternal::open(): bufferCapacity out of range = %d", capacity);
+    mFramesPerBurst = framesPerBurst; // only save good value
+
+    capacity = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
+    if (capacity < mFramesPerBurst || capacity > MAX_BUFFER_CAPACITY_IN_FRAMES) {
+        ALOGE("%s - bufferCapacity out of range = %d", __func__, capacity);
         result = AAUDIO_ERROR_OUT_OF_RANGE;
         goto error;
     }
@@ -169,16 +194,16 @@
     mClockModel.setSampleRate(getSampleRate());
     mClockModel.setFramesPerBurst(mFramesPerBurst);
 
-    if (getDataCallbackProc()) {
+    if (isDataCallbackSet()) {
         mCallbackFrames = builder.getFramesPerDataCallback();
         if (mCallbackFrames > getBufferCapacity() / 2) {
-            ALOGE("AudioStreamInternal::open(): framesPerCallback too big = %d, capacity = %d",
-                  mCallbackFrames, getBufferCapacity());
+            ALOGE("%s - framesPerCallback too big = %d, capacity = %d",
+                  __func__, mCallbackFrames, getBufferCapacity());
             result = AAUDIO_ERROR_OUT_OF_RANGE;
             goto error;
 
         } else if (mCallbackFrames < 0) {
-            ALOGE("AudioStreamInternal::open(): framesPerCallback negative");
+            ALOGE("%s - framesPerCallback negative", __func__);
             result = AAUDIO_ERROR_OUT_OF_RANGE;
             goto error;
 
@@ -204,8 +229,7 @@
 
 aaudio_result_t AudioStreamInternal::close() {
     aaudio_result_t result = AAUDIO_OK;
-    ALOGD("close(): mServiceStreamHandle = 0x%08X",
-             mServiceStreamHandle);
+    ALOGD("%s(): mServiceStreamHandle = 0x%08X", __func__, mServiceStreamHandle);
     if (mServiceStreamHandle != AAUDIO_HANDLE_INVALID) {
         // Don't close a stream while it is running.
         aaudio_stream_state_t currentState = getState();
@@ -216,8 +240,8 @@
             result = waitForStateChange(currentState, &nextState,
                                                        timeoutNanoseconds);
             if (result != AAUDIO_OK) {
-                ALOGE("close() waitForStateChange() returned %d %s",
-                result, AAudio_convertResultToText(result));
+                ALOGE("%s() waitForStateChange() returned %d %s",
+                __func__, result, AAudio_convertResultToText(result));
             }
         }
         setState(AAUDIO_STREAM_STATE_CLOSING);
@@ -240,7 +264,7 @@
 static void *aaudio_callback_thread_proc(void *context)
 {
     AudioStreamInternal *stream = (AudioStreamInternal *)context;
-    //LOGD("AudioStreamInternal(): oboe_callback_thread, stream = %p", stream);
+    //LOGD("oboe_callback_thread, stream = %p", stream);
     if (stream != NULL) {
         return stream->callbackLoop();
     } else {
@@ -288,7 +312,7 @@
     mNeedCatchUp.request();  // Ask data processing code to catch up when first timestamp received.
 
     // Start data callback thread.
-    if (result == AAUDIO_OK && getDataCallbackProc() != nullptr) {
+    if (result == AAUDIO_OK && isDataCallbackSet()) {
         // Launch the callback loop thread.
         int64_t periodNanos = mCallbackFrames
                               * AAUDIO_NANOS_PER_SECOND
@@ -329,8 +353,13 @@
     }
 }
 
-aaudio_result_t AudioStreamInternal::requestStopInternal()
+aaudio_result_t AudioStreamInternal::requestStop()
 {
+    aaudio_result_t result = stopCallback();
+    if (result != AAUDIO_OK) {
+        return result;
+    }
+
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
         ALOGE("requestStopInternal() mServiceStreamHandle invalid = 0x%08X",
               mServiceStreamHandle);
@@ -344,16 +373,6 @@
     return mServiceInterface.stopStream(mServiceStreamHandle);
 }
 
-aaudio_result_t AudioStreamInternal::requestStop()
-{
-    aaudio_result_t result = stopCallback();
-    if (result != AAUDIO_OK) {
-        return result;
-    }
-    result = requestStopInternal();
-    return result;
-}
-
 aaudio_result_t AudioStreamInternal::registerThread() {
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
         ALOGE("registerThread() mServiceStreamHandle invalid");
@@ -373,19 +392,25 @@
 }
 
 aaudio_result_t AudioStreamInternal::startClient(const android::AudioClient& client,
-                                                 audio_port_handle_t *clientHandle) {
+                                                 audio_port_handle_t *portHandle) {
+    ALOGV("%s() called", __func__);
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
-
-    return mServiceInterface.startClient(mServiceStreamHandle, client, clientHandle);
+    aaudio_result_t result =  mServiceInterface.startClient(mServiceStreamHandle,
+                                                            client, portHandle);
+    ALOGV("%s(%d) returning %d", __func__, *portHandle, result);
+    return result;
 }
 
-aaudio_result_t AudioStreamInternal::stopClient(audio_port_handle_t clientHandle) {
+aaudio_result_t AudioStreamInternal::stopClient(audio_port_handle_t portHandle) {
+    ALOGV("%s(%d) called", __func__, portHandle);
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
-    return mServiceInterface.stopClient(mServiceStreamHandle, clientHandle);
+    aaudio_result_t result = mServiceInterface.stopClient(mServiceStreamHandle, portHandle);
+    ALOGV("%s(%d) returning %d", __func__, portHandle, result);
+    return result;
 }
 
 aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t clockId,
@@ -448,34 +473,30 @@
     aaudio_result_t result = AAUDIO_OK;
     switch (message->event.event) {
         case AAUDIO_SERVICE_EVENT_STARTED:
-            ALOGD("AudioStreamInternal::onEventFromServer() got AAUDIO_SERVICE_EVENT_STARTED");
+            ALOGD("%s - got AAUDIO_SERVICE_EVENT_STARTED", __func__);
             if (getState() == AAUDIO_STREAM_STATE_STARTING) {
                 setState(AAUDIO_STREAM_STATE_STARTED);
             }
             break;
         case AAUDIO_SERVICE_EVENT_PAUSED:
-            ALOGD("AudioStreamInternal::onEventFromServer() got AAUDIO_SERVICE_EVENT_PAUSED");
+            ALOGD("%s - got AAUDIO_SERVICE_EVENT_PAUSED", __func__);
             if (getState() == AAUDIO_STREAM_STATE_PAUSING) {
                 setState(AAUDIO_STREAM_STATE_PAUSED);
             }
             break;
         case AAUDIO_SERVICE_EVENT_STOPPED:
-            ALOGD("AudioStreamInternal::onEventFromServer() got AAUDIO_SERVICE_EVENT_STOPPED");
+            ALOGD("%s - got AAUDIO_SERVICE_EVENT_STOPPED", __func__);
             if (getState() == AAUDIO_STREAM_STATE_STOPPING) {
                 setState(AAUDIO_STREAM_STATE_STOPPED);
             }
             break;
         case AAUDIO_SERVICE_EVENT_FLUSHED:
-            ALOGD("AudioStreamInternal::onEventFromServer() got AAUDIO_SERVICE_EVENT_FLUSHED");
+            ALOGD("%s - got AAUDIO_SERVICE_EVENT_FLUSHED", __func__);
             if (getState() == AAUDIO_STREAM_STATE_FLUSHING) {
                 setState(AAUDIO_STREAM_STATE_FLUSHED);
                 onFlushFromServer();
             }
             break;
-        case AAUDIO_SERVICE_EVENT_CLOSED:
-            ALOGD("AudioStreamInternal::onEventFromServer() got AAUDIO_SERVICE_EVENT_CLOSED");
-            setState(AAUDIO_STREAM_STATE_CLOSED);
-            break;
         case AAUDIO_SERVICE_EVENT_DISCONNECTED:
             // Prevent hardware from looping on old data and making buzzing sounds.
             if (getDirection() == AAUDIO_DIRECTION_OUTPUT) {
@@ -483,18 +504,18 @@
             }
             result = AAUDIO_ERROR_DISCONNECTED;
             setState(AAUDIO_STREAM_STATE_DISCONNECTED);
-            ALOGW("WARNING - AudioStreamInternal::onEventFromServer()"
-                          " AAUDIO_SERVICE_EVENT_DISCONNECTED - FIFO cleared");
+            ALOGW("%s - AAUDIO_SERVICE_EVENT_DISCONNECTED - FIFO cleared", __func__);
             break;
         case AAUDIO_SERVICE_EVENT_VOLUME:
+            ALOGD("%s - AAUDIO_SERVICE_EVENT_VOLUME %lf", __func__, message->event.dataDouble);
             mStreamVolume = (float)message->event.dataDouble;
             doSetVolume();
-            ALOGD("AudioStreamInternal::onEventFromServer() AAUDIO_SERVICE_EVENT_VOLUME %lf",
-                     message->event.dataDouble);
+            break;
+        case AAUDIO_SERVICE_EVENT_XRUN:
+            mXRunCount = static_cast<int32_t>(message->event.dataLong);
             break;
         default:
-            ALOGW("WARNING - AudioStreamInternal::onEventFromServer() Unrecognized event = %d",
-                 (int) message->event.event);
+            ALOGE("%s - Unrecognized event = %d", __func__, (int) message->event.event);
             break;
     }
     return result;
@@ -519,8 +540,7 @@
                 break;
 
             default:
-                ALOGE("WARNING - drainTimestampsFromService() Unrecognized what = %d",
-                      (int) message.what);
+                ALOGE("%s - unrecognized message.what = %d", __func__, (int) message.what);
                 result = AAUDIO_ERROR_INTERNAL;
                 break;
         }
@@ -533,7 +553,6 @@
     aaudio_result_t result = AAUDIO_OK;
 
     while (result == AAUDIO_OK) {
-        //ALOGD("AudioStreamInternal::processCommands() - looping, %d", result);
         AAudioServiceMessage message;
         if (mAudioEndpoint.readUpCommand(&message) != 1) {
             break; // no command this time, no problem
@@ -552,8 +571,7 @@
             break;
 
         default:
-            ALOGE("WARNING - processCommands() Unrecognized what = %d",
-                 (int) message.what);
+            ALOGE("%s - unrecognized message.what = %d", __func__, (int) message.what);
             result = AAUDIO_ERROR_INTERNAL;
             break;
         }
@@ -614,13 +632,13 @@
             if (wakeTimeNanos > deadlineNanos) {
                 // If we time out, just return the framesWritten so far.
                 // TODO remove after we fix the deadline bug
-                ALOGW("AudioStreamInternal::processData(): entered at %lld nanos, currently %lld",
+                ALOGW("processData(): entered at %lld nanos, currently %lld",
                       (long long) entryTimeNanos, (long long) currentTimeNanos);
-                ALOGW("AudioStreamInternal::processData(): TIMEOUT after %lld nanos",
+                ALOGW("processData(): TIMEOUT after %lld nanos",
                       (long long) timeoutNanoseconds);
-                ALOGW("AudioStreamInternal::processData(): wakeTime = %lld, deadline = %lld nanos",
+                ALOGW("processData(): wakeTime = %lld, deadline = %lld nanos",
                       (long long) wakeTimeNanos, (long long) deadlineNanos);
-                ALOGW("AudioStreamInternal::processData(): past deadline by %d micros",
+                ALOGW("processData(): past deadline by %d micros",
                       (int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND));
                 mClockModel.dump();
                 mAudioEndpoint.dump();
@@ -655,14 +673,29 @@
 }
 
 aaudio_result_t AudioStreamInternal::setBufferSize(int32_t requestedFrames) {
+    int32_t adjustedFrames = requestedFrames;
     int32_t actualFrames = 0;
-    // Round to the next highest burst size.
-    if (getFramesPerBurst() > 0) {
-        int32_t numBursts = (requestedFrames + getFramesPerBurst() - 1) / getFramesPerBurst();
-        requestedFrames = numBursts * getFramesPerBurst();
+    int32_t maximumSize = getBufferCapacity();
+
+    // Clip to minimum size so that rounding up will work better.
+    if (adjustedFrames < 1) {
+        adjustedFrames = 1;
     }
 
-    aaudio_result_t result = mAudioEndpoint.setBufferSizeInFrames(requestedFrames, &actualFrames);
+    if (adjustedFrames > maximumSize) {
+        // Clip to maximum size.
+        adjustedFrames = maximumSize;
+    } else {
+        // Round to the next highest burst size.
+        int32_t numBursts = (adjustedFrames + mFramesPerBurst - 1) / mFramesPerBurst;
+        adjustedFrames = numBursts * mFramesPerBurst;
+        // Rounding may have gone above maximum.
+        if (adjustedFrames > maximumSize) {
+            adjustedFrames = maximumSize;
+        }
+    }
+
+    aaudio_result_t result = mAudioEndpoint.setBufferSizeInFrames(adjustedFrames, &actualFrames);
     ALOGD("setBufferSize() req = %d => %d", requestedFrames, actualFrames);
     if (result < 0) {
         return result;
@@ -680,7 +713,7 @@
 }
 
 int32_t AudioStreamInternal::getFramesPerBurst() const {
-    return mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
+    return mFramesPerBurst;
 }
 
 aaudio_result_t AudioStreamInternal::joinThread(void** returnArg) {

diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 47024c0..0425cd5 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h

@@ -34,6 +34,12 @@
 
 namespace aaudio {
 
+    // These are intended to be outside the range of what is normally encountered.
+    // TODO MAXes should probably be much bigger.
+    constexpr int32_t MIN_FRAMES_PER_BURST = 16; // arbitrary
+    constexpr int32_t MAX_FRAMES_PER_BURST = 16 * 1024;  // arbitrary
+    constexpr int32_t MAX_BUFFER_CAPACITY_IN_FRAMES = 32 * 1024;  // arbitrary
+
 // A stream that talks to the AAudioService or directly to a HAL.
 class AudioStreamInternal : public AudioStream {
 
@@ -115,8 +121,6 @@
 
     aaudio_result_t processCommands();
 
-    aaudio_result_t requestStopInternal();
-
     aaudio_result_t stopCallback();
 
     virtual void advanceClientToMatchServerPosition() = 0;
@@ -134,14 +138,19 @@
     // Calculate timeout for an operation involving framesPerOperation.
     int64_t calculateReasonableTimeout(int32_t framesPerOperation);
 
-    aaudio_format_t          mDeviceFormat = AAUDIO_FORMAT_UNSPECIFIED;
+    int32_t getDeviceChannelCount() const { return mDeviceChannelCount; }
+
+    /**
+     * @return true if running in audio service, versus in app process
+     */
+    bool isInService() const { return mInService; }
 
     IsochronousClockModel    mClockModel;      // timing model for chasing the HAL
 
     AudioEndpoint            mAudioEndpoint;   // source for reads or sink for writes
     aaudio_handle_t          mServiceStreamHandle; // opaque handle returned from service
 
-    int32_t                  mFramesPerBurst;     // frames per HAL transfer
+    int32_t                  mFramesPerBurst = MIN_FRAMES_PER_BURST; // frames per HAL transfer
     int32_t                  mXRunCount = 0;      // how many underrun events?
 
     // Offset from underlying frame position.
@@ -183,6 +192,8 @@
     EndpointDescriptor       mEndpointDescriptor; // buffer description with resolved addresses
 
     int64_t                  mServiceLatencyNanos = 0;
+
+    int32_t                  mDeviceChannelCount = 0;
 };
 
 } /* namespace aaudio */

diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index b792ecd..0719fe1 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp

@@ -14,7 +14,8 @@
  * limitations under the License.
  */
 
-#define LOG_TAG (mInService ? "AAudioService" : "AAudio")
+#define LOG_TAG (mInService ? "AudioStreamInternalCapture_Service" \
+                          : "AudioStreamInternalCapture_Client")
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -101,7 +102,8 @@
     }
 
     // If the write index passed the read index then consider it an overrun.
-    if (mAudioEndpoint.getEmptyFramesAvailable() < 0) {
+    // For shared streams, the xRunCount is passed up from the service.
+    if (mAudioEndpoint.isFreeRunning() && mAudioEndpoint.getEmptyFramesAvailable() < 0) {
         mXRunCount++;
         if (ATRACE_ENABLED()) {
             ATRACE_INT("aaOverRuns", mXRunCount);
@@ -152,7 +154,7 @@
 
 aaudio_result_t AudioStreamInternalCapture::readNowWithConversion(void *buffer,
                                                                 int32_t numFrames) {
-    // ALOGD("AudioStreamInternalCapture::readNowWithConversion(%p, %d)",
+    // ALOGD("readNowWithConversion(%p, %d)",
     //              buffer, numFrames);
     WrappingBuffer wrappingBuffer;
     uint8_t *destination = (uint8_t *) buffer;
@@ -174,16 +176,16 @@
         int32_t numSamples = framesToProcess * getSamplesPerFrame();
 
         // TODO factor this out into a utility function
-        if (mDeviceFormat == getFormat()) {
+        if (getDeviceFormat() == getFormat()) {
             memcpy(destination, wrappingBuffer.data[partIndex], numBytes);
-        } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_I16
+        } else if (getDeviceFormat() == AAUDIO_FORMAT_PCM_I16
                    && getFormat() == AAUDIO_FORMAT_PCM_FLOAT) {
             AAudioConvert_pcm16ToFloat(
                     (const int16_t *) wrappingBuffer.data[partIndex],
                     (float *) destination,
                     numSamples,
                     1.0f);
-        } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT
+        } else if (getDeviceFormat() == AAUDIO_FORMAT_PCM_FLOAT
                    && getFormat() == AAUDIO_FORMAT_PCM_I16) {
             AAudioConvert_floatToPcm16(
                     (const float *) wrappingBuffer.data[partIndex],
@@ -201,7 +203,7 @@
     int32_t framesProcessed = numFrames - framesLeft;
     mAudioEndpoint.advanceReadIndex(framesProcessed);
 
-    //ALOGD("AudioStreamInternalCapture::readNowWithConversion() returns %d", framesProcessed);
+    //ALOGD("readNowWithConversion() returns %d", framesProcessed);
     return framesProcessed;
 }
 
@@ -215,14 +217,14 @@
     // Prevent retrograde motion.
     mLastFramesWritten = std::max(mLastFramesWritten,
                                   framesWrittenHardware + mFramesOffsetFromService);
-    //ALOGD("AudioStreamInternalCapture::getFramesWritten() returns %lld",
+    //ALOGD("getFramesWritten() returns %lld",
     //      (long long)mLastFramesWritten);
     return mLastFramesWritten;
 }
 
 int64_t AudioStreamInternalCapture::getFramesRead() {
     int64_t frames = mAudioEndpoint.getDataReadCounter() + mFramesOffsetFromService;
-    //ALOGD("AudioStreamInternalCapture::getFramesRead() returns %lld", (long long)frames);
+    //ALOGD("getFramesRead() returns %lld", (long long)frames);
     return frames;
 }
 
@@ -230,8 +232,7 @@
 void *AudioStreamInternalCapture::callbackLoop() {
     aaudio_result_t result = AAUDIO_OK;
     aaudio_data_callback_result_t callbackResult = AAUDIO_CALLBACK_RESULT_CONTINUE;
-    AAudioStream_dataCallback appCallback = getDataCallbackProc();
-    if (appCallback == nullptr) return NULL;
+    if (!isDataCallbackSet()) return NULL;
 
     // result might be a frame count
     while (mCallbackEnabled.load() && isActive() && (result >= 0)) {
@@ -242,35 +243,25 @@
         // This is a BLOCKING READ!
         result = read(mCallbackBuffer, mCallbackFrames, timeoutNanos);
         if ((result != mCallbackFrames)) {
-            ALOGE("AudioStreamInternalCapture(): callbackLoop: read() returned %d", result);
+            ALOGE("callbackLoop: read() returned %d", result);
             if (result >= 0) {
                 // Only read some of the frames requested. Must have timed out.
                 result = AAUDIO_ERROR_TIMEOUT;
             }
-            AAudioStream_errorCallback errorCallback = getErrorCallbackProc();
-            if (errorCallback != nullptr) {
-                (*errorCallback)(
-                        (AAudioStream *) this,
-                        getErrorCallbackUserData(),
-                        result);
-            }
+            maybeCallErrorCallback(result);
             break;
         }
 
         // Call application using the AAudio callback interface.
-        callbackResult = (*appCallback)(
-                (AAudioStream *) this,
-                getDataCallbackUserData(),
-                mCallbackBuffer,
-                mCallbackFrames);
+        callbackResult = maybeCallDataCallback(mCallbackBuffer, mCallbackFrames);
 
         if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
-            ALOGD("AudioStreamInternalCapture(): callback returned AAUDIO_CALLBACK_RESULT_STOP");
+            ALOGD("callback returned AAUDIO_CALLBACK_RESULT_STOP");
             break;
         }
     }
 
-    ALOGD("AudioStreamInternalCapture(): callbackLoop() exiting, result = %d, isActive() = %d",
+    ALOGD("callbackLoop() exiting, result = %d, isActive() = %d",
           result, (int) isActive());
     return NULL;
 }

diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 1e02eee..795ba2c 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp

@@ -14,7 +14,8 @@
  * limitations under the License.
  */
 
-#define LOG_TAG (mInService ? "AAudioService" : "AAudio")
+#define LOG_TAG (mInService ? "AudioStreamInternalPlay_Service" \
+                          : "AudioStreamInternalPlay_Client")
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -37,12 +38,26 @@
 
 AudioStreamInternalPlay::~AudioStreamInternalPlay() {}
 
+constexpr int kRampMSec = 10; // time to apply a change in volume
 
-aaudio_result_t AudioStreamInternalPlay::requestPauseInternal()
+aaudio_result_t AudioStreamInternalPlay::open(const AudioStreamBuilder &builder) {
+    aaudio_result_t result = AudioStreamInternal::open(builder);
+    if (result == AAUDIO_OK) {
+        // Sample rate is constrained to common values by now and should not overflow.
+        int32_t numFrames = kRampMSec * getSampleRate() / AAUDIO_MILLIS_PER_SECOND;
+        mVolumeRamp.setLengthInFrames(numFrames);
+    }
+    return result;
+}
+
+aaudio_result_t AudioStreamInternalPlay::requestPause()
 {
+    aaudio_result_t result = stopCallback();
+    if (result != AAUDIO_OK) {
+        return result;
+    }
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
-        ALOGE("AudioStreamInternal::requestPauseInternal() mServiceStreamHandle invalid = 0x%08X",
-              mServiceStreamHandle);
+        ALOGE("%s() mServiceStreamHandle invalid", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
     }
 
@@ -52,20 +67,9 @@
     return mServiceInterface.pauseStream(mServiceStreamHandle);
 }
 
-aaudio_result_t AudioStreamInternalPlay::requestPause()
-{
-    aaudio_result_t result = stopCallback();
-    if (result != AAUDIO_OK) {
-        return result;
-    }
-    result = requestPauseInternal();
-    return result;
-}
-
 aaudio_result_t AudioStreamInternalPlay::requestFlush() {
     if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
-        ALOGE("AudioStreamInternal::requestFlush() mServiceStreamHandle invalid = 0x%08X",
-              mServiceStreamHandle);
+        ALOGE("%s() mServiceStreamHandle invalid", __func__);
         return AAUDIO_ERROR_INVALID_STATE;
     }
 
@@ -80,7 +84,7 @@
     // Bump offset so caller does not see the retrograde motion in getFramesRead().
     int64_t offset = writeCounter - readCounter;
     mFramesOffsetFromService += offset;
-    ALOGD("advanceClientToMatchServerPosition() readN = %lld, writeN = %lld, offset = %lld",
+    ALOGV("%s() readN = %lld, writeN = %lld, offset = %lld", __func__,
           (long long)readCounter, (long long)writeCounter, (long long)mFramesOffsetFromService);
 
     // Force writeCounter to match readCounter.
@@ -94,9 +98,7 @@
 
 // Write the data, block if needed and timeoutMillis > 0
 aaudio_result_t AudioStreamInternalPlay::write(const void *buffer, int32_t numFrames,
-                                           int64_t timeoutNanoseconds)
-
-{
+                                               int64_t timeoutNanoseconds) {
     return processData((void *)buffer, numFrames, timeoutNanoseconds);
 }
 
@@ -115,7 +117,7 @@
         // Still haven't got any timestamps from server.
         // Keep waiting until we get some valid timestamps then start writing to the
         // current buffer position.
-        ALOGD("processDataNow() wait for valid timestamps");
+        ALOGV("%s() wait for valid timestamps", __func__);
         // Sleep very briefly and hope we get a timestamp soon.
         *wakeTimePtr = currentNanoTime + (2000 * AAUDIO_NANOS_PER_MICROSECOND);
         ATRACE_END();
@@ -139,7 +141,8 @@
     }
 
     // If the read index passed the write index then consider it an underrun.
-    if (mAudioEndpoint.getFullFramesAvailable() < 0) {
+    // For shared streams, the xRunCount is passed up from the service.
+    if (mAudioEndpoint.isFreeRunning() && mAudioEndpoint.getFullFramesAvailable() < 0) {
         mXRunCount++;
         if (ATRACE_ENABLED()) {
             ATRACE_INT("aaUnderRuns", mXRunCount);
@@ -196,10 +199,8 @@
 
 aaudio_result_t AudioStreamInternalPlay::writeNowWithConversion(const void *buffer,
                                                             int32_t numFrames) {
-    // ALOGD("AudioStreamInternal::writeNowWithConversion(%p, %d)",
-    //              buffer, numFrames);
     WrappingBuffer wrappingBuffer;
-    uint8_t *source = (uint8_t *) buffer;
+    uint8_t *byteBuffer = (uint8_t *) buffer;
     int32_t framesLeft = numFrames;
 
     mAudioEndpoint.getEmptyFramesAvailable(&wrappingBuffer);
@@ -213,70 +214,26 @@
             if (framesToWrite > framesAvailable) {
                 framesToWrite = framesAvailable;
             }
+
             int32_t numBytes = getBytesPerFrame() * framesToWrite;
-            int32_t numSamples = framesToWrite * getSamplesPerFrame();
             // Data conversion.
             float levelFrom;
             float levelTo;
-            bool ramping = mVolumeRamp.nextSegment(framesToWrite * getSamplesPerFrame(),
-                                                   &levelFrom, &levelTo);
-            // The formats are validated when the stream is opened so we do not have to
-            // check for illegal combinations here.
-            // TODO factor this out into a utility function
-            if (getFormat() == AAUDIO_FORMAT_PCM_FLOAT) {
-                if (mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT) {
-                    AAudio_linearRamp(
-                            (const float *) source,
-                            (float *) wrappingBuffer.data[partIndex],
-                            framesToWrite,
-                            getSamplesPerFrame(),
-                            levelFrom,
-                            levelTo);
-                } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_I16) {
-                    if (ramping) {
-                        AAudioConvert_floatToPcm16(
-                                (const float *) source,
-                                (int16_t *) wrappingBuffer.data[partIndex],
-                                framesToWrite,
-                                getSamplesPerFrame(),
-                                levelFrom,
-                                levelTo);
-                    } else {
-                        AAudioConvert_floatToPcm16(
-                                (const float *) source,
-                                (int16_t *) wrappingBuffer.data[partIndex],
-                                numSamples,
-                                levelTo);
-                    }
-                }
-            } else if (getFormat() == AAUDIO_FORMAT_PCM_I16) {
-                if (mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT) {
-                    if (ramping) {
-                        AAudioConvert_pcm16ToFloat(
-                                (const int16_t *) source,
-                                (float *) wrappingBuffer.data[partIndex],
-                                framesToWrite,
-                                getSamplesPerFrame(),
-                                levelFrom,
-                                levelTo);
-                    } else {
-                        AAudioConvert_pcm16ToFloat(
-                                (const int16_t *) source,
-                                (float *) wrappingBuffer.data[partIndex],
-                                numSamples,
-                                levelTo);
-                    }
-                } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_I16) {
-                    AAudio_linearRamp(
-                            (const int16_t *) source,
-                            (int16_t *) wrappingBuffer.data[partIndex],
-                            framesToWrite,
-                            getSamplesPerFrame(),
-                            levelFrom,
-                            levelTo);
-                }
-            }
-            source += numBytes;
+            mVolumeRamp.nextSegment(framesToWrite, &levelFrom, &levelTo);
+
+            AAudioDataConverter::FormattedData source(
+                    (void *)byteBuffer,
+                    getFormat(),
+                    getSamplesPerFrame());
+            AAudioDataConverter::FormattedData destination(
+                    wrappingBuffer.data[partIndex],
+                    getDeviceFormat(),
+                    getDeviceChannelCount());
+
+            AAudioDataConverter::convert(source, destination, framesToWrite,
+                                         levelFrom, levelTo);
+
+            byteBuffer += numBytes;
             framesLeft -= framesToWrite;
         } else {
             break;
@@ -286,7 +243,6 @@
     int32_t framesWritten = numFrames - framesLeft;
     mAudioEndpoint.advanceWriteIndex(framesWritten);
 
-    // ALOGD("AudioStreamInternal::writeNowWithConversion() returns %d", framesWritten);
     return framesWritten;
 }
 
@@ -305,7 +261,6 @@
     } else {
         mLastFramesRead = framesRead;
     }
-    //ALOGD("AudioStreamInternalPlay::getFramesRead() returns %lld", (long long)framesRead);
     return framesRead;
 }
 
@@ -313,60 +268,51 @@
 {
     int64_t framesWritten = mAudioEndpoint.getDataWriteCounter()
                                + mFramesOffsetFromService;
-    //ALOGD("AudioStreamInternalPlay::getFramesWritten() returns %lld", (long long)framesWritten);
     return framesWritten;
 }
 
 
 // Render audio in the application callback and then write the data to the stream.
 void *AudioStreamInternalPlay::callbackLoop() {
+    ALOGD("%s() entering >>>>>>>>>>>>>>>", __func__);
     aaudio_result_t result = AAUDIO_OK;
     aaudio_data_callback_result_t callbackResult = AAUDIO_CALLBACK_RESULT_CONTINUE;
-    AAudioStream_dataCallback appCallback = getDataCallbackProc();
-    if (appCallback == nullptr) return NULL;
+    if (!isDataCallbackSet()) return NULL;
     int64_t timeoutNanos = calculateReasonableTimeout(mCallbackFrames);
 
     // result might be a frame count
     while (mCallbackEnabled.load() && isActive() && (result >= 0)) {
         // Call application using the AAudio callback interface.
-        callbackResult = (*appCallback)(
-                (AAudioStream *) this,
-                getDataCallbackUserData(),
-                mCallbackBuffer,
-                mCallbackFrames);
+        callbackResult = maybeCallDataCallback(mCallbackBuffer, mCallbackFrames);
 
         if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
             // Write audio data to stream. This is a BLOCKING WRITE!
             result = write(mCallbackBuffer, mCallbackFrames, timeoutNanos);
             if ((result != mCallbackFrames)) {
-                ALOGE("AudioStreamInternalPlay(): callbackLoop: write() returned %d", result);
                 if (result >= 0) {
                     // Only wrote some of the frames requested. Must have timed out.
                     result = AAUDIO_ERROR_TIMEOUT;
                 }
-                AAudioStream_errorCallback errorCallback = getErrorCallbackProc();
-                if (errorCallback != nullptr) {
-                    (*errorCallback)(
-                            (AAudioStream *) this,
-                            getErrorCallbackUserData(),
-                            result);
-                }
+                maybeCallErrorCallback(result);
                 break;
             }
         } else if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
-            ALOGD("AudioStreamInternalPlay(): callback returned AAUDIO_CALLBACK_RESULT_STOP");
+            ALOGV("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
             break;
         }
     }
 
-    ALOGD("AudioStreamInternalPlay(): callbackLoop() exiting, result = %d, isActive() = %d",
-          result, (int) isActive());
+    ALOGD("%s() exiting, result = %d, isActive() = %d <<<<<<<<<<<<<<",
+          __func__, result, (int) isActive());
     return NULL;
 }
 
 //------------------------------------------------------------------------------
 // Implementation of PlayerBase
 status_t AudioStreamInternalPlay::doSetVolume() {
-    mVolumeRamp.setTarget(mStreamVolume * getDuckAndMuteVolume());
+    float combinedVolume = mStreamVolume * getDuckAndMuteVolume();
+    ALOGD("%s() mStreamVolume * duckAndMuteVolume = %f * %f = %f",
+          __func__, mStreamVolume, getDuckAndMuteVolume(), combinedVolume);
+    mVolumeRamp.setTarget(combinedVolume);
     return android::NO_ERROR;
 }

diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.h b/media/libaaudio/src/client/AudioStreamInternalPlay.h
index d5c1b1e..977a909 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.h
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.h

@@ -33,10 +33,22 @@
     AudioStreamInternalPlay(AAudioServiceInterface  &serviceInterface, bool inService = false);
     virtual ~AudioStreamInternalPlay();
 
+    aaudio_result_t open(const AudioStreamBuilder &builder) override;
+
     aaudio_result_t requestPause() override;
 
     aaudio_result_t requestFlush() override;
 
+    bool isFlushSupported() const override {
+        // Only implement FLUSH for OUTPUT streams.
+        return true;
+    }
+
+    bool isPauseSupported() const override {
+        // Only implement PAUSE for OUTPUT streams.
+        return true;
+    }
+
     aaudio_result_t write(const void *buffer,
                           int32_t numFrames,
                           int64_t timeoutNanoseconds) override;
@@ -52,8 +64,6 @@
 
 protected:
 
-    aaudio_result_t requestPauseInternal();
-
     void advanceClientToMatchServerPosition() override;
 
     void onFlushFromServer() override;

diff --git a/media/libaaudio/src/client/IsochronousClockModel.cpp b/media/libaaudio/src/client/IsochronousClockModel.cpp
index bac69f1..95b52be 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.cpp
+++ b/media/libaaudio/src/client/IsochronousClockModel.cpp

@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG "IsochronousClockModel"
 //#define LOG_NDEBUG 0
 #include <log/log.h>
 
@@ -41,20 +41,20 @@
 }
 
 void IsochronousClockModel::setPositionAndTime(int64_t framePosition, int64_t nanoTime) {
-    ALOGV("IsochronousClockModel::setPositionAndTime(%lld, %lld)",
+    ALOGV("setPositionAndTime(%lld, %lld)",
           (long long) framePosition, (long long) nanoTime);
     mMarkerFramePosition = framePosition;
     mMarkerNanoTime = nanoTime;
 }
 
 void IsochronousClockModel::start(int64_t nanoTime) {
-    ALOGV("IsochronousClockModel::start(nanos = %lld)\n", (long long) nanoTime);
+    ALOGV("start(nanos = %lld)\n", (long long) nanoTime);
     mMarkerNanoTime = nanoTime;
     mState = STATE_STARTING;
 }
 
 void IsochronousClockModel::stop(int64_t nanoTime) {
-    ALOGV("IsochronousClockModel::stop(nanos = %lld)\n", (long long) nanoTime);
+    ALOGV("stop(nanos = %lld)\n", (long long) nanoTime);
     setPositionAndTime(convertTimeToPosition(nanoTime), nanoTime);
     // TODO should we set position?
     mState = STATE_STOPPED;
@@ -156,7 +156,7 @@
     int64_t framesDelta = nextBurstPosition - mMarkerFramePosition;
     int64_t nanosDelta = convertDeltaPositionToTime(framesDelta);
     int64_t time = mMarkerNanoTime + nanosDelta;
-//    ALOGD("IsochronousClockModel::convertPositionToTime: pos = %llu --> time = %llu",
+//    ALOGD("convertPositionToTime: pos = %llu --> time = %llu",
 //         (unsigned long long)framePosition,
 //         (unsigned long long)time);
     return time;
@@ -171,19 +171,19 @@
     int64_t nextBurstPosition = mMarkerFramePosition + framesDelta;
     int64_t nextBurstIndex = nextBurstPosition / mFramesPerBurst;
     int64_t position = nextBurstIndex * mFramesPerBurst;
-//    ALOGD("IsochronousClockModel::convertTimeToPosition: time = %llu --> pos = %llu",
+//    ALOGD("convertTimeToPosition: time = %llu --> pos = %llu",
 //         (unsigned long long)nanoTime,
 //         (unsigned long long)position);
-//    ALOGD("IsochronousClockModel::convertTimeToPosition: framesDelta = %llu, mFramesPerBurst = %d",
+//    ALOGD("convertTimeToPosition: framesDelta = %llu, mFramesPerBurst = %d",
 //         (long long) framesDelta, mFramesPerBurst);
     return position;
 }
 
 void IsochronousClockModel::dump() const {
-    ALOGD("IsochronousClockModel::mMarkerFramePosition = %lld", (long long) mMarkerFramePosition);
-    ALOGD("IsochronousClockModel::mMarkerNanoTime      = %lld", (long long) mMarkerNanoTime);
-    ALOGD("IsochronousClockModel::mSampleRate          = %6d", mSampleRate);
-    ALOGD("IsochronousClockModel::mFramesPerBurst      = %6d", mFramesPerBurst);
-    ALOGD("IsochronousClockModel::mMaxLatenessInNanos  = %6d", mMaxLatenessInNanos);
-    ALOGD("IsochronousClockModel::mState               = %6d", mState);
+    ALOGD("mMarkerFramePosition = %lld", (long long) mMarkerFramePosition);
+    ALOGD("mMarkerNanoTime      = %lld", (long long) mMarkerNanoTime);
+    ALOGD("mSampleRate          = %6d", mSampleRate);
+    ALOGD("mFramesPerBurst      = %6d", mFramesPerBurst);
+    ALOGD("mMaxLatenessInNanos  = %6d", mMaxLatenessInNanos);
+    ALOGD("mState               = %6d", mState);
 }

diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 1eaee81..df0db79 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp

@@ -18,6 +18,8 @@
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
+#include <inttypes.h>
+#include <mutex>
 #include <time.h>
 #include <pthread.h>
 
@@ -175,13 +177,38 @@
     streamBuilder->setSharingMode(sharingMode);
 }
 
+AAUDIO_API void AAudioStreamBuilder_setUsage(AAudioStreamBuilder* builder,
+                                             aaudio_usage_t usage) {
+    AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+    streamBuilder->setUsage(usage);
+}
+
+AAUDIO_API void AAudioStreamBuilder_setContentType(AAudioStreamBuilder* builder,
+                                                   aaudio_content_type_t contentType) {
+    AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+    streamBuilder->setContentType(contentType);
+}
+
+AAUDIO_API void AAudioStreamBuilder_setInputPreset(AAudioStreamBuilder* builder,
+                                                   aaudio_input_preset_t inputPreset) {
+    AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+    streamBuilder->setInputPreset(inputPreset);
+}
+
 AAUDIO_API void AAudioStreamBuilder_setBufferCapacityInFrames(AAudioStreamBuilder* builder,
-                                                        int32_t frames)
+                                                              int32_t frames)
 {
     AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
     streamBuilder->setBufferCapacity(frames);
 }
 
+AAUDIO_API void AAudioStreamBuilder_setSessionId(AAudioStreamBuilder* builder,
+                                                 aaudio_session_id_t sessionId)
+{
+    AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+    streamBuilder->setSessionId(sessionId);
+}
+
 AAUDIO_API void AAudioStreamBuilder_setDataCallback(AAudioStreamBuilder* builder,
                                                     AAudioStream_dataCallback callback,
                                                     void *userData)
@@ -238,15 +265,26 @@
 
 AAUDIO_API aaudio_result_t  AAudioStream_close(AAudioStream* stream)
 {
+    aaudio_result_t result = AAUDIO_ERROR_NULL;
     AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
-    ALOGD("AAudioStream_close(%p)", stream);
+    ALOGD("AAudioStream_close(%p) called ---------------", stream);
     if (audioStream != nullptr) {
-        audioStream->close();
-        audioStream->unregisterPlayerBase();
-        delete audioStream;
-        return AAUDIO_OK;
+        result = audioStream->safeClose();
+        // Close will only fail if called illegally, for example, from a callback.
+        // That would result in deleting an active stream, which would cause a crash.
+        if (result == AAUDIO_OK) {
+            audioStream->unregisterPlayerBase();
+            delete audioStream;
+        } else {
+            ALOGW("%s attempt to close failed. Close it from another thread.", __func__);
+        }
     }
-    return AAUDIO_ERROR_NULL;
+    // We're potentially freeing `stream` above, so its use here makes some
+    // static analysis tools unhappy. Casting to uintptr_t helps assure
+    // said tools that we're not doing anything bad here.
+    ALOGD("AAudioStream_close(%#" PRIxPTR ") returned %d ---------",
+          reinterpret_cast<uintptr_t>(stream), result);
+    return result;
 }
 
 AAUDIO_API aaudio_result_t  AAudioStream_requestStart(AAudioStream* stream)
@@ -269,7 +307,7 @@
 {
     AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
     ALOGD("AAudioStream_requestFlush(%p)", stream);
-    return audioStream->requestFlush();
+    return audioStream->safeFlush();
 }
 
 AAUDIO_API aaudio_result_t  AAudioStream_requestStop(AAudioStream* stream)
@@ -324,7 +362,7 @@
     }
 
     // Don't allow writes when playing with a callback.
-    if (audioStream->getDataCallbackProc() != nullptr && audioStream->isActive()) {
+    if (audioStream->isDataCallbackActive()) {
         ALOGE("Cannot write to a callback stream when running.");
         return AAUDIO_ERROR_INVALID_STATE;
     }
@@ -434,6 +472,30 @@
     return audioStream->getSharingMode();
 }
 
+AAUDIO_API aaudio_usage_t AAudioStream_getUsage(AAudioStream* stream)
+{
+    AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+    return audioStream->getUsage();
+}
+
+AAUDIO_API aaudio_content_type_t AAudioStream_getContentType(AAudioStream* stream)
+{
+    AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+    return audioStream->getContentType();
+}
+
+AAUDIO_API aaudio_input_preset_t AAudioStream_getInputPreset(AAudioStream* stream)
+{
+    AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+    return audioStream->getInputPreset();
+}
+
+AAUDIO_API int32_t AAudioStream_getSessionId(AAudioStream* stream)
+{
+    AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+    return audioStream->getSessionId();
+}
+
 AAUDIO_API int64_t AAudioStream_getFramesWritten(AAudioStream* stream)
 {
     AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);

diff --git a/media/libaaudio/src/core/AAudioStreamParameters.cpp b/media/libaaudio/src/core/AAudioStreamParameters.cpp
index 82445e7..d56701b 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.cpp
+++ b/media/libaaudio/src/core/AAudioStreamParameters.cpp

@@ -15,9 +15,9 @@
  */
 
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG "AAudioStreamParameters"
 #include <utils/Log.h>
-#include <hardware/audio.h>
+#include <system/audio.h>
 
 #include "AAudioStreamParameters.h"
 
@@ -38,30 +38,43 @@
     mSamplesPerFrame = other.mSamplesPerFrame;
     mSampleRate      = other.mSampleRate;
     mDeviceId        = other.mDeviceId;
+    mSessionId       = other.mSessionId;
     mSharingMode     = other.mSharingMode;
     mAudioFormat     = other.mAudioFormat;
     mDirection       = other.mDirection;
     mBufferCapacity  = other.mBufferCapacity;
+    mUsage           = other.mUsage;
+    mContentType     = other.mContentType;
+    mInputPreset     = other.mInputPreset;
 }
 
 aaudio_result_t AAudioStreamParameters::validate() const {
     if (mSamplesPerFrame != AAUDIO_UNSPECIFIED
         && (mSamplesPerFrame < SAMPLES_PER_FRAME_MIN || mSamplesPerFrame > SAMPLES_PER_FRAME_MAX)) {
-        ALOGE("AAudioStreamParameters: channelCount out of range = %d", mSamplesPerFrame);
+        ALOGE("channelCount out of range = %d", mSamplesPerFrame);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }
 
     if (mDeviceId < 0) {
-        ALOGE("AAudioStreamParameters: deviceId out of range = %d", mDeviceId);
+        ALOGE("deviceId out of range = %d", mDeviceId);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }
 
+    // All Session ID values are legal.
+    switch (mSessionId) {
+        case AAUDIO_SESSION_ID_NONE:
+        case AAUDIO_SESSION_ID_ALLOCATE:
+            break;
+        default:
+            break;
+    }
+
     switch (mSharingMode) {
         case AAUDIO_SHARING_MODE_EXCLUSIVE:
         case AAUDIO_SHARING_MODE_SHARED:
             break;
         default:
-            ALOGE("AAudioStreamParameters: illegal sharingMode = %d", mSharingMode);
+            ALOGE("illegal sharingMode = %d", mSharingMode);
             return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
             // break;
     }
@@ -72,19 +85,19 @@
         case AAUDIO_FORMAT_PCM_FLOAT:
             break; // valid
         default:
-            ALOGE("AAudioStreamParameters: audioFormat not valid = %d", mAudioFormat);
+            ALOGE("audioFormat not valid = %d", mAudioFormat);
             return AAUDIO_ERROR_INVALID_FORMAT;
             // break;
     }
 
     if (mSampleRate != AAUDIO_UNSPECIFIED
         && (mSampleRate < SAMPLE_RATE_HZ_MIN || mSampleRate > SAMPLE_RATE_HZ_MAX)) {
-        ALOGE("AAudioStreamParameters: sampleRate out of range = %d", mSampleRate);
+        ALOGE("sampleRate out of range = %d", mSampleRate);
         return AAUDIO_ERROR_INVALID_RATE;
     }
 
     if (mBufferCapacity < 0) {
-        ALOGE("AAudioStreamParameters: bufferCapacity out of range = %d", mBufferCapacity);
+        ALOGE("bufferCapacity out of range = %d", mBufferCapacity);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }
 
@@ -93,7 +106,55 @@
         case AAUDIO_DIRECTION_OUTPUT:
             break; // valid
         default:
-            ALOGE("AAudioStreamParameters: direction not valid = %d", mDirection);
+            ALOGE("direction not valid = %d", mDirection);
+            return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+            // break;
+    }
+
+    switch (mUsage) {
+        case AAUDIO_UNSPECIFIED:
+        case AAUDIO_USAGE_MEDIA:
+        case AAUDIO_USAGE_VOICE_COMMUNICATION:
+        case AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+        case AAUDIO_USAGE_ALARM:
+        case AAUDIO_USAGE_NOTIFICATION:
+        case AAUDIO_USAGE_NOTIFICATION_RINGTONE:
+        case AAUDIO_USAGE_NOTIFICATION_EVENT:
+        case AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+        case AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+        case AAUDIO_USAGE_ASSISTANCE_SONIFICATION:
+        case AAUDIO_USAGE_GAME:
+        case AAUDIO_USAGE_ASSISTANT:
+            break; // valid
+        default:
+            ALOGE("usage not valid = %d", mUsage);
+            return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+            // break;
+    }
+
+    switch (mContentType) {
+        case AAUDIO_UNSPECIFIED:
+        case AAUDIO_CONTENT_TYPE_MUSIC:
+        case AAUDIO_CONTENT_TYPE_MOVIE:
+        case AAUDIO_CONTENT_TYPE_SONIFICATION:
+        case AAUDIO_CONTENT_TYPE_SPEECH:
+            break; // valid
+        default:
+            ALOGE("content type not valid = %d", mContentType);
+            return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+            // break;
+    }
+
+    switch (mInputPreset) {
+        case AAUDIO_UNSPECIFIED:
+        case AAUDIO_INPUT_PRESET_GENERIC:
+        case AAUDIO_INPUT_PRESET_CAMCORDER:
+        case AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION:
+        case AAUDIO_INPUT_PRESET_VOICE_RECOGNITION:
+        case AAUDIO_INPUT_PRESET_UNPROCESSED:
+            break; // valid
+        default:
+            ALOGE("input preset not valid = %d", mInputPreset);
             return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
             // break;
     }
@@ -102,12 +163,15 @@
 }
 
 void AAudioStreamParameters::dump() const {
-    ALOGD("AAudioStreamParameters mDeviceId        = %d", mDeviceId);
-    ALOGD("AAudioStreamParameters mSampleRate      = %d", mSampleRate);
-    ALOGD("AAudioStreamParameters mSamplesPerFrame = %d", mSamplesPerFrame);
-    ALOGD("AAudioStreamParameters mSharingMode     = %d", (int)mSharingMode);
-    ALOGD("AAudioStreamParameters mAudioFormat     = %d", (int)mAudioFormat);
-    ALOGD("AAudioStreamParameters mDirection       = %d", mDirection);
-    ALOGD("AAudioStreamParameters mBufferCapacity  = %d", mBufferCapacity);
+    ALOGD("mDeviceId        = %6d", mDeviceId);
+    ALOGD("mSessionId       = %6d", mSessionId);
+    ALOGD("mSampleRate      = %6d", mSampleRate);
+    ALOGD("mSamplesPerFrame = %6d", mSamplesPerFrame);
+    ALOGD("mSharingMode     = %6d", (int)mSharingMode);
+    ALOGD("mAudioFormat     = %6d", (int)mAudioFormat);
+    ALOGD("mDirection       = %6d", mDirection);
+    ALOGD("mBufferCapacity  = %6d", mBufferCapacity);
+    ALOGD("mUsage           = %6d", mUsage);
+    ALOGD("mContentType     = %6d", mContentType);
+    ALOGD("mInputPreset     = %6d", mInputPreset);
 }
-

diff --git a/media/libaaudio/src/core/AAudioStreamParameters.h b/media/libaaudio/src/core/AAudioStreamParameters.h
index 5e67c93..ce5dacd 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.h
+++ b/media/libaaudio/src/core/AAudioStreamParameters.h

@@ -88,6 +88,38 @@
         mDirection = direction;
     }
 
+    aaudio_usage_t getUsage() const {
+        return mUsage;
+    }
+
+    void setUsage(aaudio_usage_t usage) {
+        mUsage = usage;
+    }
+
+    aaudio_content_type_t getContentType() const {
+        return mContentType;
+    }
+
+    void setContentType(aaudio_content_type_t contentType) {
+        mContentType = contentType;
+    }
+
+    aaudio_input_preset_t getInputPreset() const {
+        return mInputPreset;
+    }
+
+    void setInputPreset(aaudio_input_preset_t inputPreset) {
+        mInputPreset = inputPreset;
+    }
+
+    aaudio_session_id_t getSessionId() const {
+        return mSessionId;
+    }
+
+    void setSessionId(aaudio_session_id_t sessionId) {
+        mSessionId = sessionId;
+    }
+
     int32_t calculateBytesPerFrame() const {
         return getSamplesPerFrame() * AAudioConvert_formatToSizeInBytes(getFormat());
     }
@@ -109,7 +141,11 @@
     aaudio_sharing_mode_t      mSharingMode     = AAUDIO_SHARING_MODE_SHARED;
     aaudio_format_t            mAudioFormat     = AAUDIO_FORMAT_UNSPECIFIED;
     aaudio_direction_t         mDirection       = AAUDIO_DIRECTION_OUTPUT;
+    aaudio_usage_t             mUsage           = AAUDIO_UNSPECIFIED;
+    aaudio_content_type_t      mContentType     = AAUDIO_UNSPECIFIED;
+    aaudio_input_preset_t      mInputPreset     = AAUDIO_UNSPECIFIED;
     int32_t                    mBufferCapacity  = AAUDIO_UNSPECIFIED;
+    aaudio_session_id_t        mSessionId       = AAUDIO_SESSION_ID_NONE;
 };
 
 } /* namespace aaudio */

diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 8dcc37a..358021b 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp

@@ -43,7 +43,7 @@
     LOG_ALWAYS_FATAL_IF(!(getState() == AAUDIO_STREAM_STATE_CLOSED
                           || getState() == AAUDIO_STREAM_STATE_UNINITIALIZED
                           || getState() == AAUDIO_STREAM_STATE_DISCONNECTED),
-                        "aaudio stream still in use, state = %s",
+                        "~AudioStream() - still in use, state = %s",
                         AAudio_convertStreamStateToText(getState()));
 
     mPlayerBase->clearParentReference(); // remove reference to this AudioStream
@@ -74,15 +74,28 @@
     }
 
     // Copy parameters from the Builder because the Builder may be deleted after this call.
+    // TODO AudioStream should be a subclass of AudioStreamParameters
     mSamplesPerFrame = builder.getSamplesPerFrame();
     mSampleRate = builder.getSampleRate();
     mDeviceId = builder.getDeviceId();
     mFormat = builder.getFormat();
     mSharingMode = builder.getSharingMode();
     mSharingModeMatchRequired = builder.isSharingModeMatchRequired();
-
     mPerformanceMode = builder.getPerformanceMode();
 
+    mUsage = builder.getUsage();
+    if (mUsage == AAUDIO_UNSPECIFIED) {
+        mUsage = AAUDIO_USAGE_MEDIA;
+    }
+    mContentType = builder.getContentType();
+    if (mContentType == AAUDIO_UNSPECIFIED) {
+        mContentType = AAUDIO_CONTENT_TYPE_MUSIC;
+    }
+    mInputPreset = builder.getInputPreset();
+    if (mInputPreset == AAUDIO_UNSPECIFIED) {
+        mInputPreset = AAUDIO_INPUT_PRESET_VOICE_RECOGNITION;
+    }
+
     // callbacks
     mFramesPerDataCallback = builder.getFramesPerDataCallback();
     mDataCallbackProc = builder.getDataCallbackProc();
@@ -91,18 +104,159 @@
     mErrorCallbackUserData = builder.getErrorCallbackUserData();
 
     // This is very helpful for debugging in the future. Please leave it in.
-    ALOGI("AudioStream::open() rate = %d, channels = %d, format = %d, sharing = %s, dir = %s",
+    ALOGI("open() rate   = %d, channels    = %d, format   = %d, sharing = %s, dir = %s",
           mSampleRate, mSamplesPerFrame, mFormat,
           AudioStream_convertSharingModeToShortText(mSharingMode),
           (getDirection() == AAUDIO_DIRECTION_OUTPUT) ? "OUTPUT" : "INPUT");
-    ALOGI("AudioStream::open() device = %d, perfMode = %d, callback: %s with frames = %d",
-          mDeviceId, mPerformanceMode,
-          (mDataCallbackProc == nullptr ? "OFF" : "ON"),
+    ALOGI("open() device = %d, sessionId   = %d, perfMode = %d, callback: %s with frames = %d",
+          mDeviceId,
+          mSessionId,
+          mPerformanceMode,
+          (isDataCallbackSet() ? "ON" : "OFF"),
           mFramesPerDataCallback);
+    ALOGI("open() usage  = %d, contentType = %d, inputPreset = %d",
+          mUsage, mContentType, mInputPreset);
 
     return AAUDIO_OK;
 }
 
+aaudio_result_t AudioStream::safeStart() {
+    std::lock_guard<std::mutex> lock(mStreamLock);
+    if (collidesWithCallback()) {
+        ALOGE("%s cannot be called from a callback!", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    return requestStart();
+}
+
+aaudio_result_t AudioStream::safePause() {
+    if (!isPauseSupported()) {
+        return AAUDIO_ERROR_UNIMPLEMENTED;
+    }
+
+    std::lock_guard<std::mutex> lock(mStreamLock);
+    if (collidesWithCallback()) {
+        ALOGE("%s cannot be called from a callback!", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+
+    switch (getState()) {
+        // Proceed with pausing.
+        case AAUDIO_STREAM_STATE_STARTING:
+        case AAUDIO_STREAM_STATE_STARTED:
+        case AAUDIO_STREAM_STATE_DISCONNECTED:
+            break;
+
+            // Transition from one inactive state to another.
+        case AAUDIO_STREAM_STATE_OPEN:
+        case AAUDIO_STREAM_STATE_STOPPED:
+        case AAUDIO_STREAM_STATE_FLUSHED:
+            setState(AAUDIO_STREAM_STATE_PAUSED);
+            return AAUDIO_OK;
+
+            // Redundant?
+        case AAUDIO_STREAM_STATE_PAUSING:
+        case AAUDIO_STREAM_STATE_PAUSED:
+            return AAUDIO_OK;
+
+            // Don't interfere with transitional states or when closed.
+        case AAUDIO_STREAM_STATE_STOPPING:
+        case AAUDIO_STREAM_STATE_FLUSHING:
+        case AAUDIO_STREAM_STATE_CLOSING:
+        case AAUDIO_STREAM_STATE_CLOSED:
+        default:
+            ALOGW("safePause() stream not running, state = %s",
+                  AAudio_convertStreamStateToText(getState()));
+            return AAUDIO_ERROR_INVALID_STATE;
+    }
+
+    return requestPause();
+}
+
+aaudio_result_t AudioStream::safeFlush() {
+    if (!isFlushSupported()) {
+        ALOGE("flush not supported for this stream");
+        return AAUDIO_ERROR_UNIMPLEMENTED;
+    }
+
+    std::lock_guard<std::mutex> lock(mStreamLock);
+    if (collidesWithCallback()) {
+        ALOGE("stream cannot be flushed from a callback!");
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+
+    aaudio_result_t result = AAudio_isFlushAllowed(getState());
+    if (result != AAUDIO_OK) {
+        return result;
+    }
+
+    return requestFlush();
+}
+
+aaudio_result_t AudioStream::safeStop() {
+    std::lock_guard<std::mutex> lock(mStreamLock);
+    if (collidesWithCallback()) {
+        ALOGE("stream cannot be stopped from a callback!");
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+
+    switch (getState()) {
+        // Proceed with stopping.
+        case AAUDIO_STREAM_STATE_STARTING:
+        case AAUDIO_STREAM_STATE_STARTED:
+        case AAUDIO_STREAM_STATE_DISCONNECTED:
+            break;
+
+        // Transition from one inactive state to another.
+        case AAUDIO_STREAM_STATE_OPEN:
+        case AAUDIO_STREAM_STATE_PAUSED:
+        case AAUDIO_STREAM_STATE_FLUSHED:
+            setState(AAUDIO_STREAM_STATE_STOPPED);
+            return AAUDIO_OK;
+
+        // Redundant?
+        case AAUDIO_STREAM_STATE_STOPPING:
+        case AAUDIO_STREAM_STATE_STOPPED:
+            return AAUDIO_OK;
+
+        // Don't interfere with transitional states or when closed.
+        case AAUDIO_STREAM_STATE_PAUSING:
+        case AAUDIO_STREAM_STATE_FLUSHING:
+        case AAUDIO_STREAM_STATE_CLOSING:
+        case AAUDIO_STREAM_STATE_CLOSED:
+        default:
+            ALOGW("requestStop() stream not running, state = %s",
+                  AAudio_convertStreamStateToText(getState()));
+            return AAUDIO_ERROR_INVALID_STATE;
+    }
+
+    return requestStop();
+}
+
+aaudio_result_t AudioStream::safeClose() {
+    std::lock_guard<std::mutex> lock(mStreamLock);
+    if (collidesWithCallback()) {
+        ALOGE("%s cannot be called from a callback!", __func__);
+        return AAUDIO_ERROR_INVALID_STATE;
+    }
+    return close();
+}
+
+void AudioStream::setState(aaudio_stream_state_t state) {
+    ALOGV("%s(%p) from %d to %d", __func__, this, mState, state);
+    // CLOSED is a final state
+    if (mState == AAUDIO_STREAM_STATE_CLOSED) {
+        ALOGE("%s(%p) tried to set to %d but already CLOSED", __func__, this, state);
+
+    // Once DISCONNECTED, we can only move to CLOSED state.
+    } else if (mState == AAUDIO_STREAM_STATE_DISCONNECTED
+               && state != AAUDIO_STREAM_STATE_CLOSED) {
+        ALOGE("%s(%p) tried to set to %d but already DISCONNECTED", __func__, this, state);
+
+    } else {
+        mState = state;
+    }
+}
 
 aaudio_result_t AudioStream::waitForStateChange(aaudio_stream_state_t currentState,
                                                 aaudio_stream_state_t *nextState,
@@ -163,7 +317,7 @@
                                      void* threadArg)
 {
     if (mHasThread) {
-        ALOGE("AudioStream::createThread() - mHasThread already true");
+        ALOGE("createThread() - mHasThread already true");
         return AAUDIO_ERROR_INVALID_STATE;
     }
     if (threadProc == nullptr) {
@@ -175,8 +329,22 @@
     setPeriodNanoseconds(periodNanoseconds);
     int err = pthread_create(&mThread, nullptr, AudioStream_internalThreadProc, this);
     if (err != 0) {
-        return AAudioConvert_androidToAAudioResult(-errno);
+        android::status_t status = -errno;
+        ALOGE("createThread() - pthread_create() failed, %d", status);
+        return AAudioConvert_androidToAAudioResult(status);
     } else {
+        // TODO Use AAudioThread or maybe AndroidThread
+        // Name the thread with an increasing index, "AAudio_#", for debugging.
+        static std::atomic<uint32_t> nextThreadIndex{1};
+        char name[16]; // max length for a pthread_name
+        uint32_t index = nextThreadIndex++;
+        // Wrap the index so that we do not hit the 16 char limit
+        // and to avoid hard-to-read large numbers.
+        index = index % 100000;  // arbitrary
+        snprintf(name, sizeof(name), "AAudio_%u", index);
+        err = pthread_setname_np(mThread, name);
+        ALOGW_IF((err != 0), "Could not set name of AAudio thread. err = %d", err);
+
         mHasThread = true;
         return AAUDIO_OK;
     }
@@ -185,7 +353,7 @@
 aaudio_result_t AudioStream::joinThread(void** returnArg, int64_t timeoutNanoseconds)
 {
     if (!mHasThread) {
-        ALOGE("AudioStream::joinThread() - but has no thread");
+        ALOGE("joinThread() - but has no thread");
         return AAUDIO_ERROR_INVALID_STATE;
     }
 #if 0
@@ -199,6 +367,57 @@
     return err ? AAudioConvert_androidToAAudioResult(-errno) : mThreadRegistrationResult;
 }
 
+aaudio_data_callback_result_t AudioStream::maybeCallDataCallback(void *audioData,
+                                                                 int32_t numFrames) {
+    aaudio_data_callback_result_t result = AAUDIO_CALLBACK_RESULT_STOP;
+    AAudioStream_dataCallback dataCallback = getDataCallbackProc();
+    if (dataCallback != nullptr) {
+        // Store thread ID of caller to detect stop() and close() calls from callback.
+        pid_t expected = CALLBACK_THREAD_NONE;
+        if (mDataCallbackThread.compare_exchange_strong(expected, gettid())) {
+            result = (*dataCallback)(
+                    (AAudioStream *) this,
+                    getDataCallbackUserData(),
+                    audioData,
+                    numFrames);
+            mDataCallbackThread.store(CALLBACK_THREAD_NONE);
+        } else {
+            ALOGW("%s() data callback already running!", __func__);
+        }
+    }
+    return result;
+}
+
+void AudioStream::maybeCallErrorCallback(aaudio_result_t result) {
+    AAudioStream_errorCallback errorCallback = getErrorCallbackProc();
+    if (errorCallback != nullptr) {
+        // Store thread ID of caller to detect stop() and close() calls from callback.
+        pid_t expected = CALLBACK_THREAD_NONE;
+        if (mErrorCallbackThread.compare_exchange_strong(expected, gettid())) {
+            (*errorCallback)(
+                    (AAudioStream *) this,
+                    getErrorCallbackUserData(),
+                    result);
+            mErrorCallbackThread.store(CALLBACK_THREAD_NONE);
+        } else {
+            ALOGW("%s() error callback already running!", __func__);
+        }
+    }
+}
+
+// Is this running on the same thread as a callback?
+// Note: This cannot be implemented using a thread_local because that would
+// require using a thread_local variable that is shared between streams.
+// So a thread_local variable would prevent stopping or closing stream A from
+// a callback on stream B, which is currently legal and not so terrible.
+bool AudioStream::collidesWithCallback() const {
+    pid_t thisThread = gettid();
+    // Compare the current thread ID with the thread ID of the callback
+    // threads to see it they match. If so then this code is being
+    // called from one of the stream callback functions.
+    return ((mErrorCallbackThread.load() == thisThread)
+            || (mDataCallbackThread.load() == thisThread));
+}
 
 #if AAUDIO_USE_VOLUME_SHAPER
 android::media::VolumeShaper::Status AudioStream::applyVolumeShaper(
@@ -209,6 +428,12 @@
 }
 #endif
 
+void AudioStream::setDuckAndMuteVolume(float duckAndMuteVolume) {
+    ALOGD("%s() to %f", __func__, duckAndMuteVolume);
+    mDuckAndMuteVolume = duckAndMuteVolume;
+    doSetVolume(); // apply this change
+}
+
 AudioStream::MyPlayerBase::MyPlayerBase(AudioStream *parent) : mParent(parent) {
 }
 
@@ -230,7 +455,6 @@
     }
 }
 
-
 void AudioStream::MyPlayerBase::destroy() {
     unregisterWithAudioManager();
 }

diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 34202d2..31b895c 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h

@@ -39,6 +39,8 @@
 
 class AudioStreamBuilder;
 
+constexpr pid_t        CALLBACK_THREAD_NONE = 0;
+
 /**
  * AAudio audio stream.
  */
@@ -49,14 +51,46 @@
 
     virtual ~AudioStream();
 
+    /**
+     * Lock a mutex and make sure we are not calling from a callback function.
+     * @return result of requestStart();
+     */
+    aaudio_result_t safeStart();
+
+    aaudio_result_t safePause();
+
+    aaudio_result_t safeFlush();
+
+    aaudio_result_t safeStop();
+
+    aaudio_result_t safeClose();
 
     // =========== Begin ABSTRACT methods ===========================
+protected:
 
     /* Asynchronous requests.
      * Use waitForStateChange() to wait for completion.
      */
     virtual aaudio_result_t requestStart() = 0;
 
+    /**
+     * Check the state to see if Pause if currently legal.
+     *
+     * @param result pointer to return code
+     * @return true if OK to continue, if false then return result
+     */
+    bool checkPauseStateTransition(aaudio_result_t *result);
+
+    virtual bool isFlushSupported() const {
+        // Only implement FLUSH for OUTPUT streams.
+        return false;
+    }
+
+    virtual bool isPauseSupported() const {
+        // Only implement PAUSE for OUTPUT streams.
+        return false;
+    }
+
     virtual aaudio_result_t requestPause()
     {
         // Only implement this for OUTPUT streams.
@@ -70,6 +104,7 @@
 
     virtual aaudio_result_t requestStop() = 0;
 
+public:
     virtual aaudio_result_t getTimestamp(clockid_t clockId,
                                        int64_t *framePosition,
                                        int64_t *timeNanoseconds) = 0;
@@ -81,7 +116,6 @@
      */
     virtual aaudio_result_t updateStateMachine() = 0;
 
-
     // =========== End ABSTRACT methods ===========================
 
     virtual aaudio_result_t waitForStateChange(aaudio_stream_state_t currentState,
@@ -188,6 +222,22 @@
 
     virtual aaudio_direction_t getDirection() const = 0;
 
+    aaudio_usage_t getUsage() const {
+        return mUsage;
+    }
+
+    aaudio_content_type_t getContentType() const {
+        return mContentType;
+    }
+
+    aaudio_input_preset_t getInputPreset() const {
+        return mInputPreset;
+    }
+
+    int32_t getSessionId() const {
+        return mSessionId;
+    }
+
     /**
      * This is only valid after setSamplesPerFrame() and setFormat() have been called.
      */
@@ -202,6 +252,20 @@
         return AAudioConvert_formatToSizeInBytes(mFormat);
     }
 
+    /**
+     * This is only valid after setSamplesPerFrame() and setDeviceFormat() have been called.
+     */
+    int32_t getBytesPerDeviceFrame() const {
+        return mSamplesPerFrame * getBytesPerDeviceSample();
+    }
+
+    /**
+     * This is only valid after setDeviceFormat() has been called.
+     */
+    int32_t getBytesPerDeviceSample() const {
+        return AAudioConvert_formatToSizeInBytes(getDeviceFormat());
+    }
+
     virtual int64_t getFramesWritten() = 0;
 
     virtual int64_t getFramesRead() = 0;
@@ -209,13 +273,19 @@
     AAudioStream_dataCallback getDataCallbackProc() const {
         return mDataCallbackProc;
     }
+
     AAudioStream_errorCallback getErrorCallbackProc() const {
         return mErrorCallbackProc;
     }
 
+    aaudio_data_callback_result_t maybeCallDataCallback(void *audioData, int32_t numFrames);
+
+    void maybeCallErrorCallback(aaudio_result_t result);
+
     void *getDataCallbackUserData() const {
         return mDataCallbackUserData;
     }
+
     void *getErrorCallbackUserData() const {
         return mErrorCallbackUserData;
     }
@@ -224,10 +294,25 @@
         return mFramesPerDataCallback;
     }
 
-    bool isDataCallbackActive() {
-        return (mDataCallbackProc != nullptr) && isActive();
+    /**
+     * @return true if data callback has been specified
+     */
+    bool isDataCallbackSet() const {
+        return mDataCallbackProc != nullptr;
     }
 
+    /**
+     * @return true if data callback has been specified and stream is running
+     */
+    bool isDataCallbackActive() const {
+        return isDataCallbackSet() && isActive();
+    }
+
+    /**
+     * @return true if called from the same thread as the callback
+     */
+    bool collidesWithCallback() const;
+
     // ============== I/O ===========================
     // A Stream will only implement read() or write() depending on its direction.
     virtual aaudio_result_t write(const void *buffer __unused,
@@ -243,12 +328,9 @@
     }
 
     // This is used by the AudioManager to duck and mute the stream when changing audio focus.
-    void setDuckAndMuteVolume(float duckAndMuteVolume) {
-        mDuckAndMuteVolume = duckAndMuteVolume;
-        doSetVolume(); // apply this change
-    }
+    void setDuckAndMuteVolume(float duckAndMuteVolume);
 
-    float getDuckAndMuteVolume() {
+    float getDuckAndMuteVolume() const {
         return mDuckAndMuteVolume;
     }
 
@@ -288,11 +370,13 @@
         return mPlayerBase->getResult();
     }
 
+    // Pass pause request through PlayerBase for tracking.
     aaudio_result_t systemPause() {
         mPlayerBase->pause();
         return mPlayerBase->getResult();
     }
 
+    // Pass stop request through PlayerBase for tracking.
     aaudio_result_t systemStop() {
         mPlayerBase->stop();
         return mPlayerBase->getResult();
@@ -331,17 +415,17 @@
 
         android::status_t playerStart() override {
             // mParent should NOT be null. So go ahead and crash if it is.
-            mResult = mParent->requestStart();
+            mResult = mParent->safeStart();
             return AAudioConvert_aaudioToAndroidStatus(mResult);
         }
 
         android::status_t playerPause() override {
-            mResult = mParent->requestPause();
+            mResult = mParent->safePause();
             return AAudioConvert_aaudioToAndroidStatus(mResult);
         }
 
         android::status_t playerStop() override {
-            mResult = mParent->requestStop();
+            mResult = mParent->safeStop();
             return AAudioConvert_aaudioToAndroidStatus(mResult);
         }
 
@@ -371,6 +455,7 @@
 
     /**
      * This should not be called after the open() call.
+     * TODO for multiple setters: assert(mState == AAUDIO_STREAM_STATE_UNINITIALIZED)
      */
     void setSampleRate(int32_t sampleRate) {
         mSampleRate = sampleRate;
@@ -397,15 +482,26 @@
         mFormat = format;
     }
 
-    void setState(aaudio_stream_state_t state) {
-        mState = state;
+    /**
+     * This should not be called after the open() call.
+     */
+    void setDeviceFormat(aaudio_format_t format) {
+        mDeviceFormat = format;
     }
 
+    aaudio_format_t getDeviceFormat() const {
+        return mDeviceFormat;
+    }
+
+    void setState(aaudio_stream_state_t state);
+
     void setDeviceId(int32_t deviceId) {
         mDeviceId = deviceId;
     }
 
-    std::mutex           mStreamMutex;
+    void setSessionId(int32_t sessionId) {
+        mSessionId = sessionId;
+    }
 
     std::atomic<bool>    mCallbackEnabled{false};
 
@@ -413,6 +509,21 @@
 
 protected:
 
+    /**
+     * Either convert the data from device format to app format and return a pointer
+     * to the conversion buffer,
+     * OR just pass back the original pointer.
+     *
+     * Note that this is only used for the INPUT path.
+     *
+     * @param audioData
+     * @param numFrames
+     * @return original pointer or the conversion buffer
+     */
+    virtual const void * maybeConvertDeviceData(const void *audioData, int32_t numFrames) {
+        return audioData;
+    }
+
     void setPeriodNanoseconds(int64_t periodNanoseconds) {
         mPeriodNanoseconds.store(periodNanoseconds, std::memory_order_release);
     }
@@ -421,40 +532,74 @@
         return mPeriodNanoseconds.load(std::memory_order_acquire);
     }
 
+    /**
+     * This should not be called after the open() call.
+     */
+    void setUsage(aaudio_usage_t usage) {
+        mUsage = usage;
+    }
+
+    /**
+     * This should not be called after the open() call.
+     */
+    void setContentType(aaudio_content_type_t contentType) {
+        mContentType = contentType;
+    }
+
+    /**
+     * This should not be called after the open() call.
+     */
+    void setInputPreset(aaudio_input_preset_t inputPreset) {
+        mInputPreset = inputPreset;
+    }
+
 private:
+
+    std::mutex                 mStreamLock;
+
     const android::sp<MyPlayerBase>   mPlayerBase;
 
     // These do not change after open().
-    int32_t                mSamplesPerFrame = AAUDIO_UNSPECIFIED;
-    int32_t                mSampleRate = AAUDIO_UNSPECIFIED;
-    int32_t                mDeviceId = AAUDIO_UNSPECIFIED;
-    aaudio_sharing_mode_t  mSharingMode = AAUDIO_SHARING_MODE_SHARED;
-    bool                   mSharingModeMatchRequired = false; // must match sharing mode requested
-    aaudio_format_t        mFormat = AAUDIO_FORMAT_UNSPECIFIED;
-    aaudio_stream_state_t  mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
+    int32_t                     mSamplesPerFrame = AAUDIO_UNSPECIFIED;
+    int32_t                     mSampleRate = AAUDIO_UNSPECIFIED;
+    int32_t                     mDeviceId = AAUDIO_UNSPECIFIED;
+    aaudio_sharing_mode_t       mSharingMode = AAUDIO_SHARING_MODE_SHARED;
+    bool                        mSharingModeMatchRequired = false; // must match sharing mode requested
+    aaudio_format_t             mFormat = AAUDIO_FORMAT_UNSPECIFIED;
+    aaudio_stream_state_t       mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
+    aaudio_performance_mode_t   mPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
 
-    aaudio_performance_mode_t mPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
+    aaudio_usage_t              mUsage           = AAUDIO_UNSPECIFIED;
+    aaudio_content_type_t       mContentType     = AAUDIO_UNSPECIFIED;
+    aaudio_input_preset_t       mInputPreset     = AAUDIO_UNSPECIFIED;
+
+    int32_t                     mSessionId = AAUDIO_UNSPECIFIED;
+
+    // Sometimes the hardware is operating with a different format from the app.
+    // Then we require conversion in AAudio.
+    aaudio_format_t             mDeviceFormat = AAUDIO_FORMAT_UNSPECIFIED;
 
     // callback ----------------------------------
 
     AAudioStream_dataCallback   mDataCallbackProc = nullptr;  // external callback functions
     void                       *mDataCallbackUserData = nullptr;
     int32_t                     mFramesPerDataCallback = AAUDIO_UNSPECIFIED; // frames
+    std::atomic<pid_t>          mDataCallbackThread{CALLBACK_THREAD_NONE};
 
     AAudioStream_errorCallback  mErrorCallbackProc = nullptr;
     void                       *mErrorCallbackUserData = nullptr;
+    std::atomic<pid_t>          mErrorCallbackThread{CALLBACK_THREAD_NONE};
 
     // background thread ----------------------------------
-    bool                   mHasThread = false;
-    pthread_t              mThread; // initialized in constructor
+    bool                        mHasThread = false;
+    pthread_t                   mThread; // initialized in constructor
 
     // These are set by the application thread and then read by the audio pthread.
-    std::atomic<int64_t>   mPeriodNanoseconds; // for tuning SCHED_FIFO threads
+    std::atomic<int64_t>        mPeriodNanoseconds; // for tuning SCHED_FIFO threads
     // TODO make atomic?
-    aaudio_audio_thread_proc_t mThreadProc = nullptr;
-    void*                  mThreadArg = nullptr;
-    aaudio_result_t        mThreadRegistrationResult = AAUDIO_OK;
-
+    aaudio_audio_thread_proc_t  mThreadProc = nullptr;
+    void                       *mThreadArg = nullptr;
+    aaudio_result_t             mThreadRegistrationResult = AAUDIO_OK;
 
 };
 

diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 09ebb3e..3a7a578 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp

@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AAudio"
+#define LOG_TAG "AudioStreamBuilder"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -87,7 +87,7 @@
             break;
 
         default:
-            ALOGE("AudioStreamBuilder(): bad direction = %d", direction);
+            ALOGE("%s() bad direction = %d", __func__, direction);
             result = AAUDIO_ERROR_ILLEGAL_ARGUMENT;
     }
     return result;
@@ -99,7 +99,7 @@
 aaudio_result_t AudioStreamBuilder::build(AudioStream** streamPtr) {
     AudioStream *audioStream = nullptr;
     if (streamPtr == nullptr) {
-        ALOGE("AudioStreamBuilder::build() streamPtr is null");
+        ALOGE("%s() streamPtr is null", __func__);
         return AAUDIO_ERROR_NULL;
     }
     *streamPtr = nullptr;
@@ -124,13 +124,11 @@
     if (mapExclusivePolicy == AAUDIO_UNSPECIFIED) {
         mapExclusivePolicy = AAUDIO_MMAP_EXCLUSIVE_POLICY_DEFAULT;
     }
-    ALOGD("AudioStreamBuilder(): mmapPolicy = %d, mapExclusivePolicy = %d",
-          mmapPolicy, mapExclusivePolicy);
 
     aaudio_sharing_mode_t sharingMode = getSharingMode();
     if ((sharingMode == AAUDIO_SHARING_MODE_EXCLUSIVE)
         && (mapExclusivePolicy == AAUDIO_POLICY_NEVER)) {
-        ALOGW("AudioStreamBuilder(): EXCLUSIVE sharing mode not supported. Use SHARED.");
+        ALOGD("%s() EXCLUSIVE sharing mode not supported. Use SHARED.", __func__);
         sharingMode = AAUDIO_SHARING_MODE_SHARED;
         setSharingMode(sharingMode);
     }
@@ -141,6 +139,14 @@
     // TODO Support other performance settings in MMAP mode.
     // Disable MMAP if low latency not requested.
     if (getPerformanceMode() != AAUDIO_PERFORMANCE_MODE_LOW_LATENCY) {
+        ALOGD("%s() MMAP not available because AAUDIO_PERFORMANCE_MODE_LOW_LATENCY not used.",
+              __func__);
+        allowMMap = false;
+    }
+
+    // SessionID and Effects are only supported in Legacy mode.
+    if (getSessionId() != AAUDIO_SESSION_ID_NONE) {
+        ALOGD("%s() MMAP not available because sessionId used.", __func__);
         allowMMap = false;
     }
 
@@ -156,7 +162,7 @@
             audioStream = nullptr;
 
             if (isMMap && allowLegacy) {
-                ALOGD("AudioStreamBuilder.build() MMAP stream did not open so try Legacy path");
+                ALOGV("%s() MMAP stream did not open so try Legacy path", __func__);
                 // If MMAP stream failed to open then TRY using a legacy stream.
                 result = builder_createStream(getDirection(), sharingMode,
                                               false, &audioStream);
@@ -190,7 +196,7 @@
         case AAUDIO_PERFORMANCE_MODE_LOW_LATENCY:
             break;
         default:
-            ALOGE("AudioStreamBuilder: illegal performanceMode = %d", mPerformanceMode);
+            ALOGE("illegal performanceMode = %d", mPerformanceMode);
             return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
             // break;
     }
@@ -199,7 +205,7 @@
     if (mFramesPerDataCallback != AAUDIO_UNSPECIFIED
         && (mFramesPerDataCallback < FRAMES_PER_DATA_CALLBACK_MIN
             || mFramesPerDataCallback > FRAMES_PER_DATA_CALLBACK_MAX)) {
-        ALOGE("AudioStreamBuilder: framesPerDataCallback out of range = %d",
+        ALOGE("framesPerDataCallback out of range = %d",
               mFramesPerDataCallback);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }

diff --git a/media/libaaudio/src/fifo/FifoBuffer.cpp b/media/libaaudio/src/fifo/FifoBuffer.cpp
index c8ec15079f..b09258e 100644
--- a/media/libaaudio/src/fifo/FifoBuffer.cpp
+++ b/media/libaaudio/src/fifo/FifoBuffer.cpp

@@ -45,7 +45,7 @@
     int32_t bytesPerBuffer = bytesPerFrame * capacityInFrames;
     mStorage = new uint8_t[bytesPerBuffer];
     mStorageOwned = true;
-    ALOGD("FifoBuffer: capacityInFrames = %d, bytesPerFrame = %d",
+    ALOGV("capacityInFrames = %d, bytesPerFrame = %d",
           capacityInFrames, bytesPerFrame);
 }
 

diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
index ee2504d..a6b9f5d 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp

@@ -19,10 +19,12 @@
 #include <utils/Log.h>
 
 #include <stdint.h>
-#include <utils/String16.h>
+
+#include <aaudio/AAudio.h>
+#include <audio_utils/primitives.h>
 #include <media/AudioTrack.h>
 #include <media/AudioTimestamp.h>
-#include <aaudio/AAudio.h>
+#include <utils/String16.h>
 
 #include "core/AudioStream.h"
 #include "legacy/AudioStreamLegacy.h"
@@ -48,19 +50,17 @@
     return AudioStreamLegacy_callback;
 }
 
-int32_t AudioStreamLegacy::callDataCallbackFrames(uint8_t *buffer, int32_t numFrames) {
+aaudio_data_callback_result_t AudioStreamLegacy::callDataCallbackFrames(uint8_t *buffer,
+                                                                        int32_t numFrames) {
+    void *finalAudioData = buffer;
     if (getDirection() == AAUDIO_DIRECTION_INPUT) {
         // Increment before because we already got the data from the device.
         incrementFramesRead(numFrames);
+        finalAudioData = (void *) maybeConvertDeviceData(buffer, numFrames);
     }
 
     // Call using the AAudio callback interface.
-    AAudioStream_dataCallback appCallback = getDataCallbackProc();
-    aaudio_data_callback_result_t callbackResult = (*appCallback)(
-            (AAudioStream *) this,
-            getDataCallbackUserData(),
-            buffer,
-            numFrames);
+    aaudio_data_callback_result_t callbackResult = maybeCallDataCallback(finalAudioData, numFrames);
 
     if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE
             && getDirection() == AAUDIO_DIRECTION_OUTPUT) {
@@ -72,31 +72,40 @@
 
 // Implement FixedBlockProcessor
 int32_t AudioStreamLegacy::onProcessFixedBlock(uint8_t *buffer, int32_t numBytes) {
-    int32_t numFrames = numBytes / getBytesPerFrame();
-    return callDataCallbackFrames(buffer, numFrames);
+    int32_t numFrames = numBytes / getBytesPerDeviceFrame();
+    return (int32_t) callDataCallbackFrames(buffer, numFrames);
 }
 
 void AudioStreamLegacy::processCallbackCommon(aaudio_callback_operation_t opcode, void *info) {
     aaudio_data_callback_result_t callbackResult;
+    // This illegal size can be used to tell AudioFlinger to stop calling us.
+    // This takes advantage of AudioFlinger killing the stream.
+    // TODO add to API in AudioRecord and AudioTrack
+    const size_t SIZE_STOP_CALLBACKS = SIZE_MAX;
 
     switch (opcode) {
         case AAUDIO_CALLBACK_OPERATION_PROCESS_DATA: {
-            checkForDisconnectRequest();
+            (void) checkForDisconnectRequest(true);
 
             // Note that this code assumes an AudioTrack::Buffer is the same as
             // AudioRecord::Buffer
             // TODO define our own AudioBuffer and pass it from the subclasses.
             AudioTrack::Buffer *audioBuffer = static_cast<AudioTrack::Buffer *>(info);
-            if (getState() == AAUDIO_STREAM_STATE_DISCONNECTED || !mCallbackEnabled.load()) {
-                audioBuffer->size = 0; // silence the buffer
+            if (getState() == AAUDIO_STREAM_STATE_DISCONNECTED) {
+                ALOGW("processCallbackCommon() data, stream disconnected");
+                audioBuffer->size = SIZE_STOP_CALLBACKS;
+            } else if (!mCallbackEnabled.load()) {
+                ALOGW("processCallbackCommon() stopping because callback disabled");
+                audioBuffer->size = SIZE_STOP_CALLBACKS;
             } else {
                 if (audioBuffer->frameCount == 0) {
+                    ALOGW("processCallbackCommon() data, frameCount is zero");
                     return;
                 }
 
                 // If the caller specified an exact size then use a block size adapter.
                 if (mBlockAdapter != nullptr) {
-                    int32_t byteCount = audioBuffer->frameCount * getBytesPerFrame();
+                    int32_t byteCount = audioBuffer->frameCount * getBytesPerDeviceFrame();
                     callbackResult = mBlockAdapter->processVariableBlock(
                             (uint8_t *) audioBuffer->raw, byteCount);
                 } else {
@@ -105,9 +114,12 @@
                                                             audioBuffer->frameCount);
                 }
                 if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
-                    audioBuffer->size = audioBuffer->frameCount * getBytesPerFrame();
-                } else {
-                    audioBuffer->size = 0;
+                    audioBuffer->size = audioBuffer->frameCount * getBytesPerDeviceFrame();
+                } else { // STOP or invalid result
+                    ALOGW("%s() callback requested stop, fake an error", __func__);
+                    audioBuffer->size = SIZE_STOP_CALLBACKS;
+                    // Disable the callback just in case AudioFlinger keeps trying to call us.
+                    mCallbackEnabled.store(false);
                 }
 
                 if (updateStateMachine() != AAUDIO_OK) {
@@ -130,26 +142,23 @@
     }
 }
 
-
-
-void AudioStreamLegacy::checkForDisconnectRequest() {
+aaudio_result_t AudioStreamLegacy::checkForDisconnectRequest(bool errorCallbackEnabled) {
     if (mRequestDisconnect.isRequested()) {
         ALOGD("checkForDisconnectRequest() mRequestDisconnect acknowledged");
-        forceDisconnect();
+        forceDisconnect(errorCallbackEnabled);
         mRequestDisconnect.acknowledge();
         mCallbackEnabled.store(false);
+        return AAUDIO_ERROR_DISCONNECTED;
+    } else {
+        return AAUDIO_OK;
     }
 }
 
-void AudioStreamLegacy::forceDisconnect() {
+void AudioStreamLegacy::forceDisconnect(bool errorCallbackEnabled) {
     if (getState() != AAUDIO_STREAM_STATE_DISCONNECTED) {
         setState(AAUDIO_STREAM_STATE_DISCONNECTED);
-        if (getErrorCallbackProc() != nullptr) {
-            (*getErrorCallbackProc())(
-                    (AAudioStream *) this,
-                    getErrorCallbackUserData(),
-                    AAUDIO_ERROR_DISCONNECTED
-            );
+        if (errorCallbackEnabled) {
+            maybeCallErrorCallback(AAUDIO_ERROR_DISCONNECTED);
         }
     }
 }
@@ -175,19 +184,17 @@
     int64_t localPosition;
     status_t status = extendedTimestamp->getBestTimestamp(&localPosition, timeNanoseconds,
                                                           timebase, &location);
-    // use MonotonicCounter to prevent retrograde motion.
-    mTimestampPosition.update32((int32_t)localPosition);
-    *framePosition = mTimestampPosition.get();
+    if (status == OK) {
+        // use MonotonicCounter to prevent retrograde motion.
+        mTimestampPosition.update32((int32_t) localPosition);
+        *framePosition = mTimestampPosition.get();
+    }
 
 //    ALOGD("getBestTimestamp() fposition: server = %6lld, kernel = %6lld, location = %d",
 //          (long long) extendedTimestamp->mPosition[ExtendedTimestamp::Location::LOCATION_SERVER],
 //          (long long) extendedTimestamp->mPosition[ExtendedTimestamp::Location::LOCATION_KERNEL],
 //          (int)location);
-    if (status == WOULD_BLOCK) {
-        return AAUDIO_ERROR_INVALID_STATE;
-    } else {
-        return AAudioConvert_androidToAAudioResult(status);
-    }
+    return AAudioConvert_androidToAAudioResult(status);
 }
 
 void AudioStreamLegacy::onAudioDeviceUpdate(audio_port_handle_t deviceId)

diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.h b/media/libaaudio/src/legacy/AudioStreamLegacy.h
index 7e28579..494edbc 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.h
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.h

@@ -112,12 +112,14 @@
 
     void onAudioDeviceUpdate(audio_port_handle_t deviceId);
 
-    void checkForDisconnectRequest();
+    /*
+     * Check to see whether a callback thread has requested a disconnected.
+     * @param errorCallbackEnabled set true to call errorCallback on disconnect
+     * @return AAUDIO_OK or AAUDIO_ERROR_DISCONNECTED
+     */
+    aaudio_result_t checkForDisconnectRequest(bool errorCallbackEnabled);
 
-    void forceDisconnect();
-
-    void onStart() { mCallbackEnabled.store(true); }
-    void onStop() { mCallbackEnabled.store(false); }
+    void forceDisconnect(bool errorCallbackEnabled = true);
 
     int64_t incrementFramesWritten(int32_t frames) {
         return mFramesWritten.increment(frames);

diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index bc6e60c..505f2ee 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp

@@ -19,13 +19,15 @@
 #include <utils/Log.h>
 
 #include <stdint.h>
-#include <utils/String16.h>
-#include <media/AudioRecord.h>
-#include <aaudio/AAudio.h>
 
-#include "AudioClock.h"
+#include <aaudio/AAudio.h>
+#include <audio_utils/primitives.h>
+#include <media/AudioRecord.h>
+#include <utils/String16.h>
+
 #include "legacy/AudioStreamLegacy.h"
 #include "legacy/AudioStreamRecord.h"
+#include "utility/AudioClock.h"
 #include "utility/FixedBlockWriter.h"
 
 using namespace android;
@@ -63,10 +65,6 @@
     size_t frameCount = (builder.getBufferCapacity() == AAUDIO_UNSPECIFIED) ? 0
                         : builder.getBufferCapacity();
 
-    // TODO implement an unspecified Android format then use that.
-    audio_format_t format = (getFormat() == AAUDIO_FORMAT_UNSPECIFIED)
-            ? AUDIO_FORMAT_PCM_FLOAT
-            : AAudioConvert_aaudioToAndroidDataFormat(getFormat());
 
     audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE;
     aaudio_performance_mode_t perfMode = getPerformanceMode();
@@ -82,6 +80,35 @@
             break;
     }
 
+    // Preserve behavior of API 26
+    if (getFormat() == AAUDIO_FORMAT_UNSPECIFIED) {
+        setFormat(AAUDIO_FORMAT_PCM_FLOAT);
+    }
+
+    // Maybe change device format to get a FAST path.
+    // AudioRecord does not support FAST mode for FLOAT data.
+    // TODO AudioRecord should allow FLOAT data paths for FAST tracks.
+    // So IF the user asks for low latency FLOAT
+    // AND the sampleRate is likely to be compatible with FAST
+    // THEN request I16 and convert to FLOAT when passing to user.
+    // Note that hard coding 48000 Hz is not ideal because the sampleRate
+    // for a FAST path might not be 48000 Hz.
+    // It normally is but there is a chance that it is not.
+    // And there is no reliable way to know that in advance.
+    // Luckily the consequences of a wrong guess are minor.
+    // We just may not get a FAST track.
+    // But we wouldn't have anyway without this hack.
+    constexpr int32_t kMostLikelySampleRateForFast = 48000;
+    if (getFormat() == AAUDIO_FORMAT_PCM_FLOAT
+            && perfMode == AAUDIO_PERFORMANCE_MODE_LOW_LATENCY
+            && (samplesPerFrame <= 2) // FAST only for mono and stereo
+            && (getSampleRate() == kMostLikelySampleRateForFast
+                || getSampleRate() == AAUDIO_UNSPECIFIED)) {
+        setDeviceFormat(AAUDIO_FORMAT_PCM_I16);
+    } else {
+        setDeviceFormat(getFormat());
+    }
+
     uint32_t notificationFrames = 0;
 
     // Setup the callback if there is one.
@@ -96,47 +123,84 @@
     }
     mCallbackBufferSize = builder.getFramesPerDataCallback();
 
-    ALOGD("AudioStreamRecord::open(), request notificationFrames = %u, frameCount = %u",
-          notificationFrames, (uint)frameCount);
-    mAudioRecord = new AudioRecord(
-            mOpPackageName // const String16& opPackageName TODO does not compile
-            );
-    if (getDeviceId() != AAUDIO_UNSPECIFIED) {
-        mAudioRecord->setInputDevice(getDeviceId());
-    }
-    mAudioRecord->set(
-            AUDIO_SOURCE_VOICE_RECOGNITION,
-            getSampleRate(),
-            format,
-            channelMask,
-            frameCount,
-            callback,
-            callbackData,
-            notificationFrames,
-            false /*threadCanCallJava*/,
-            AUDIO_SESSION_ALLOCATE,
-            streamTransferType,
-            flags
-            //   int uid = -1,
-            //   pid_t pid = -1,
-            //   const audio_attributes_t* pAttributes = nullptr
-            );
+    // Don't call mAudioRecord->setInputDevice() because it will be overwritten by set()!
+    audio_port_handle_t selectedDeviceId = (getDeviceId() == AAUDIO_UNSPECIFIED)
+                                           ? AUDIO_PORT_HANDLE_NONE
+                                           : getDeviceId();
 
-    // Did we get a valid track?
-    status_t status = mAudioRecord->initCheck();
-    if (status != OK) {
-        close();
-        ALOGE("AudioStreamRecord::open(), initCheck() returned %d", status);
-        return AAudioConvert_androidToAAudioResult(status);
+    const audio_content_type_t contentType =
+            AAudioConvert_contentTypeToInternal(builder.getContentType());
+    const audio_source_t source =
+            AAudioConvert_inputPresetToAudioSource(builder.getInputPreset());
+
+    const audio_attributes_t attributes = {
+            .content_type = contentType,
+            .usage = AUDIO_USAGE_UNKNOWN, // only used for output
+            .source = source,
+            .flags = AUDIO_FLAG_NONE, // Different than the AUDIO_INPUT_FLAGS
+            .tags = ""
+    };
+
+    aaudio_session_id_t requestedSessionId = builder.getSessionId();
+    audio_session_t sessionId = AAudioConvert_aaudioToAndroidSessionId(requestedSessionId);
+
+    // ----------- open the AudioRecord ---------------------
+    // Might retry, but never more than once.
+    for (int i = 0; i < 2; i ++) {
+        audio_format_t requestedInternalFormat =
+                AAudioConvert_aaudioToAndroidDataFormat(getDeviceFormat());
+
+        mAudioRecord = new AudioRecord(
+                mOpPackageName // const String16& opPackageName TODO does not compile
+        );
+        mAudioRecord->set(
+                AUDIO_SOURCE_DEFAULT, // ignored because we pass attributes below
+                getSampleRate(),
+                requestedInternalFormat,
+                channelMask,
+                frameCount,
+                callback,
+                callbackData,
+                notificationFrames,
+                false /*threadCanCallJava*/,
+                sessionId,
+                streamTransferType,
+                flags,
+                AUDIO_UID_INVALID, // DEFAULT uid
+                -1,                // DEFAULT pid
+                &attributes,
+                selectedDeviceId
+        );
+
+        // Did we get a valid track?
+        status_t status = mAudioRecord->initCheck();
+        if (status != OK) {
+            close();
+            ALOGE("open(), initCheck() returned %d", status);
+            return AAudioConvert_androidToAAudioResult(status);
+        }
+
+        // Check to see if it was worth hacking the deviceFormat.
+        bool gotFastPath = (mAudioRecord->getFlags() & AUDIO_INPUT_FLAG_FAST)
+                           == AUDIO_INPUT_FLAG_FAST;
+        if (getFormat() != getDeviceFormat() && !gotFastPath) {
+            // We tried to get a FAST path by switching the device format.
+            // But it didn't work. So we might as well reopen using the same
+            // format for device and for app.
+            ALOGD("%s() used a different device format but no FAST path, reopen", __func__);
+            mAudioRecord.clear();
+            setDeviceFormat(getFormat());
+        } else {
+            break; // Keep the one we just opened.
+        }
     }
 
     // Get the actual values from the AudioRecord.
     setSamplesPerFrame(mAudioRecord->channelCount());
-    setFormat(AAudioConvert_androidToAAudioDataFormat(mAudioRecord->format()));
 
     int32_t actualSampleRate = mAudioRecord->getSampleRate();
     ALOGW_IF(actualSampleRate != getSampleRate(),
-             "AudioStreamRecord::open() sampleRate changed from %d to %d",
+             "open() sampleRate changed from %d to %d",
              getSampleRate(), actualSampleRate);
     setSampleRate(actualSampleRate);
 
@@ -149,6 +213,29 @@
         mBlockAdapter = nullptr;
     }
 
+    // Allocate format conversion buffer if needed.
+    if (getDeviceFormat() == AAUDIO_FORMAT_PCM_I16
+        && getFormat() == AAUDIO_FORMAT_PCM_FLOAT) {
+
+        if (builder.getDataCallbackProc() != nullptr) {
+            // If we have a callback then we need to convert the data into an internal float
+            // array and then pass that entire array to the app.
+            mFormatConversionBufferSizeInFrames =
+                    (mCallbackBufferSize != AAUDIO_UNSPECIFIED)
+                    ? mCallbackBufferSize : getFramesPerBurst();
+            int32_t numSamples = mFormatConversionBufferSizeInFrames * getSamplesPerFrame();
+            mFormatConversionBufferFloat = std::make_unique<float[]>(numSamples);
+        } else {
+            // If we don't have a callback then we will read into an internal short array
+            // and then convert into the app float array in read().
+            mFormatConversionBufferSizeInFrames = getFramesPerBurst();
+            int32_t numSamples = mFormatConversionBufferSizeInFrames * getSamplesPerFrame();
+            mFormatConversionBufferI16 = std::make_unique<int16_t[]>(numSamples);
+        }
+        ALOGD("%s() setup I16>FLOAT conversion buffer with %d frames",
+              __func__, mFormatConversionBufferSizeInFrames);
+    }
+
     // Update performance mode based on the actual stream.
     // For example, if the sample rate does not match native then you won't get a FAST track.
     audio_input_flags_t actualFlags = mAudioRecord->getFlags();
@@ -164,14 +251,21 @@
 
     // Log warning if we did not get what we asked for.
     ALOGW_IF(actualFlags != flags,
-             "AudioStreamRecord::open() flags changed from 0x%08X to 0x%08X",
+             "open() flags changed from 0x%08X to 0x%08X",
              flags, actualFlags);
     ALOGW_IF(actualPerformanceMode != perfMode,
-             "AudioStreamRecord::open() perfMode changed from %d to %d",
+             "open() perfMode changed from %d to %d",
              perfMode, actualPerformanceMode);
 
     setState(AAUDIO_STREAM_STATE_OPEN);
     setDeviceId(mAudioRecord->getRoutedDeviceId());
+
+    aaudio_session_id_t actualSessionId =
+            (requestedSessionId == AAUDIO_SESSION_ID_NONE)
+            ? AAUDIO_SESSION_ID_NONE
+            : (aaudio_session_id_t) mAudioRecord->getSessionId();
+    setSessionId(actualSessionId);
+
     mAudioRecord->addAudioDeviceCallback(mDeviceCallback);
 
     return AAUDIO_OK;
@@ -189,6 +283,24 @@
     return AudioStream::close();
 }
 
+const void * AudioStreamRecord::maybeConvertDeviceData(const void *audioData, int32_t numFrames) {
+    if (mFormatConversionBufferFloat.get() != nullptr) {
+        LOG_ALWAYS_FATAL_IF(numFrames > mFormatConversionBufferSizeInFrames,
+                            "%s() conversion size %d too large for buffer %d",
+                            __func__, numFrames, mFormatConversionBufferSizeInFrames);
+
+        int32_t numSamples = numFrames * getSamplesPerFrame();
+        // Only conversion supported is I16 to FLOAT
+        memcpy_to_float_from_i16(
+                    mFormatConversionBufferFloat.get(),
+                    (const int16_t *) audioData,
+                    numSamples);
+        return mFormatConversionBufferFloat.get();
+    } else {
+        return audioData;
+    }
+}
+
 void AudioStreamRecord::processCallback(int event, void *info) {
     switch (event) {
         case AudioRecord::EVENT_MORE_DATA:
@@ -217,11 +329,13 @@
         return AAudioConvert_androidToAAudioResult(err);
     }
 
+    // Enable callback before starting AudioTrack to avoid shutting
+    // down because of a race condition.
+    mCallbackEnabled.store(true);
     err = mAudioRecord->start();
     if (err != OK) {
         return AAudioConvert_androidToAAudioResult(err);
     } else {
-        onStart();
         setState(AAUDIO_STREAM_STATE_STARTING);
     }
     return AAUDIO_OK;
@@ -231,15 +345,16 @@
     if (mAudioRecord.get() == nullptr) {
         return AAUDIO_ERROR_INVALID_STATE;
     }
-    onStop();
     setState(AAUDIO_STREAM_STATE_STOPPING);
     incrementFramesWritten(getFramesRead() - getFramesWritten()); // TODO review
     mTimestampPosition.set(getFramesRead());
     mAudioRecord->stop();
-    mFramesRead.reset32();
+    mCallbackEnabled.store(false);
+    mFramesWritten.reset32(); // service writes frames, service position reset on flush
     mTimestampPosition.reset32();
-    checkForDisconnectRequest();
-    return AAUDIO_OK;
+    // Pass false to prevent errorCallback from being called after disconnect
+    // when app has already requested a stop().
+    return checkForDisconnectRequest(false);
 }
 
 aaudio_result_t AudioStreamRecord::updateStateMachine()
@@ -272,9 +387,10 @@
                                       int32_t numFrames,
                                       int64_t timeoutNanoseconds)
 {
-    int32_t bytesPerFrame = getBytesPerFrame();
+    int32_t bytesPerDeviceFrame = getBytesPerDeviceFrame();
     int32_t numBytes;
-    aaudio_result_t result = AAudioConvert_framesToBytes(numFrames, bytesPerFrame, &numBytes);
+    // This will detect out of range values for numFrames.
+    aaudio_result_t result = AAudioConvert_framesToBytes(numFrames, bytesPerDeviceFrame, &numBytes);
     if (result != AAUDIO_OK) {
         return result;
     }
@@ -285,19 +401,49 @@
 
     // TODO add timeout to AudioRecord
     bool blocking = (timeoutNanoseconds > 0);
-    ssize_t bytesRead = mAudioRecord->read(buffer, numBytes, blocking);
-    if (bytesRead == WOULD_BLOCK) {
+
+    ssize_t bytesActuallyRead = 0;
+    ssize_t totalBytesRead = 0;
+    if (mFormatConversionBufferI16.get() != nullptr) {
+        // Convert I16 data to float using an intermediate buffer.
+        float *floatBuffer = (float *) buffer;
+        int32_t framesLeft = numFrames;
+        // Perform conversion using multiple read()s if necessary.
+        while (framesLeft > 0) {
+            // Read into short internal buffer.
+            int32_t framesToRead = std::min(framesLeft, mFormatConversionBufferSizeInFrames);
+            size_t bytesToRead = framesToRead * bytesPerDeviceFrame;
+            bytesActuallyRead = mAudioRecord->read(mFormatConversionBufferI16.get(), bytesToRead, blocking);
+            if (bytesActuallyRead <= 0) {
+                break;
+            }
+            totalBytesRead += bytesActuallyRead;
+            int32_t framesToConvert = bytesActuallyRead / bytesPerDeviceFrame;
+            // Convert into app float buffer.
+            size_t numSamples = framesToConvert * getSamplesPerFrame();
+            memcpy_to_float_from_i16(
+                    floatBuffer,
+                    mFormatConversionBufferI16.get(),
+                    numSamples);
+            floatBuffer += numSamples;
+            framesLeft -= framesToConvert;
+        }
+    } else {
+        bytesActuallyRead = mAudioRecord->read(buffer, numBytes, blocking);
+        totalBytesRead = bytesActuallyRead;
+    }
+    if (bytesActuallyRead == WOULD_BLOCK) {
         return 0;
-    } else if (bytesRead < 0) {
-        // in this context, a DEAD_OBJECT is more likely to be a disconnect notification due to
-        // AudioRecord invalidation
-        if (bytesRead == DEAD_OBJECT) {
+    } else if (bytesActuallyRead < 0) {
+        // In this context, a DEAD_OBJECT is more likely to be a disconnect notification due to
+        // AudioRecord invalidation.
+        if (bytesActuallyRead == DEAD_OBJECT) {
             setState(AAUDIO_STREAM_STATE_DISCONNECTED);
             return AAUDIO_ERROR_DISCONNECTED;
         }
-        return AAudioConvert_androidToAAudioResult(bytesRead);
+        return AAudioConvert_androidToAAudioResult(bytesActuallyRead);
     }
-    int32_t framesRead = (int32_t)(bytesRead / bytesPerFrame);
+    int32_t framesRead = (int32_t)(totalBytesRead / bytesPerDeviceFrame);
     incrementFramesRead(framesRead);
 
     result = updateStateMachine();

diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.h b/media/libaaudio/src/legacy/AudioStreamRecord.h
index c1723ba..2f41d34 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.h
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.h

@@ -76,6 +76,8 @@
         return incrementFramesRead(frames);
     }
 
+    const void * maybeConvertDeviceData(const void *audioData, int32_t numFrames) override;
+
 private:
     android::sp<android::AudioRecord> mAudioRecord;
     // adapts between variable sized blocks and fixed size blocks
@@ -83,6 +85,11 @@
 
     // TODO add 64-bit position reporting to AudioRecord and use it.
     android::String16                mOpPackageName;
+
+    // Only one type of conversion buffer is used.
+    std::unique_ptr<float[]>         mFormatConversionBufferFloat;
+    std::unique_ptr<int16_t[]>       mFormatConversionBufferI16;
+    int32_t                          mFormatConversionBufferSizeInFrames = 0;
 };
 
 } /* namespace aaudio */

diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 0e9aaef..505cd77 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp

@@ -22,6 +22,7 @@
 #include <media/AudioTrack.h>
 
 #include <aaudio/AAudio.h>
+#include <system/audio.h>
 #include "utility/AudioClock.h"
 #include "legacy/AudioStreamLegacy.h"
 #include "legacy/AudioStreamTrack.h"
@@ -113,14 +114,35 @@
     }
     mCallbackBufferSize = builder.getFramesPerDataCallback();
 
-    ALOGD("AudioStreamTrack::open(), request notificationFrames = %d, frameCount = %u",
+    ALOGD("open(), request notificationFrames = %d, frameCount = %u",
           notificationFrames, (uint)frameCount);
-    mAudioTrack = new AudioTrack(); // TODO review
-    if (getDeviceId() != AAUDIO_UNSPECIFIED) {
-        mAudioTrack->setOutputDevice(getDeviceId());
-    }
+
+    // Don't call mAudioTrack->setDeviceId() because it will be overwritten by set()!
+    audio_port_handle_t selectedDeviceId = (getDeviceId() == AAUDIO_UNSPECIFIED)
+                                           ? AUDIO_PORT_HANDLE_NONE
+                                           : getDeviceId();
+
+    const audio_content_type_t contentType =
+            AAudioConvert_contentTypeToInternal(builder.getContentType());
+    const audio_usage_t usage =
+            AAudioConvert_usageToInternal(builder.getUsage());
+
+    const audio_attributes_t attributes = {
+            .content_type = contentType,
+            .usage = usage,
+            .source = AUDIO_SOURCE_DEFAULT, // only used for recording
+            .flags = AUDIO_FLAG_NONE, // Different than the AUDIO_OUTPUT_FLAGS
+            .tags = ""
+    };
+
+    static_assert(AAUDIO_UNSPECIFIED == AUDIO_SESSION_ALLOCATE, "Session IDs should match");
+
+    aaudio_session_id_t requestedSessionId = builder.getSessionId();
+    audio_session_t sessionId = AAudioConvert_aaudioToAndroidSessionId(requestedSessionId);
+
+    mAudioTrack = new AudioTrack();
     mAudioTrack->set(
-            (audio_stream_type_t) AUDIO_STREAM_MUSIC,
+            AUDIO_STREAM_DEFAULT,  // ignored because we pass attributes below
             getSampleRate(),
             format,
             channelMask,
@@ -129,17 +151,26 @@
             callback,
             callbackData,
             notificationFrames,
-            0 /*sharedBuffer*/,
-            false /*threadCanCallJava*/,
-            AUDIO_SESSION_ALLOCATE,
-            streamTransferType
-            );
+            0,       // DEFAULT sharedBuffer*/,
+            false,   // DEFAULT threadCanCallJava
+            sessionId,
+            streamTransferType,
+            NULL,    // DEFAULT audio_offload_info_t
+            AUDIO_UID_INVALID, // DEFAULT uid
+            -1,      // DEFAULT pid
+            &attributes,
+            // WARNING - If doNotReconnect set true then audio stops after plugging and unplugging
+            // headphones a few times.
+            false,   // DEFAULT doNotReconnect,
+            1.0f,    // DEFAULT maxRequiredSpeed
+            selectedDeviceId
+    );
 
     // Did we get a valid track?
     status_t status = mAudioTrack->initCheck();
     if (status != NO_ERROR) {
         close();
-        ALOGE("AudioStreamTrack::open(), initCheck() returned %d", status);
+        ALOGE("open(), initCheck() returned %d", status);
         return AAudioConvert_androidToAAudioResult(status);
     }
 
@@ -150,10 +181,11 @@
     aaudio_format_t aaudioFormat =
             AAudioConvert_androidToAAudioDataFormat(mAudioTrack->format());
     setFormat(aaudioFormat);
+    setDeviceFormat(aaudioFormat);
 
     int32_t actualSampleRate = mAudioTrack->getSampleRate();
     ALOGW_IF(actualSampleRate != getSampleRate(),
-             "AudioStreamTrack::open() sampleRate changed from %d to %d",
+             "open() sampleRate changed from %d to %d",
              getSampleRate(), actualSampleRate);
     setSampleRate(actualSampleRate);
 
@@ -168,6 +200,13 @@
 
     setState(AAUDIO_STREAM_STATE_OPEN);
     setDeviceId(mAudioTrack->getRoutedDeviceId());
+
+    aaudio_session_id_t actualSessionId =
+            (requestedSessionId == AAUDIO_SESSION_ID_NONE)
+            ? AAUDIO_SESSION_ID_NONE
+            : (aaudio_session_id_t) mAudioTrack->getSessionId();
+    setSessionId(actualSessionId);
+
     mAudioTrack->addAudioDeviceCallback(mDeviceCallback);
 
     // Update performance mode based on the actual stream flags.
@@ -186,10 +225,10 @@
 
     // Log warning if we did not get what we asked for.
     ALOGW_IF(actualFlags != flags,
-             "AudioStreamTrack::open() flags changed from 0x%08X to 0x%08X",
+             "open() flags changed from 0x%08X to 0x%08X",
              flags, actualFlags);
     ALOGW_IF(actualPerformanceMode != perfMode,
-             "AudioStreamTrack::open() perfMode changed from %d to %d",
+             "open() perfMode changed from %d to %d",
              perfMode, actualPerformanceMode);
 
     return AAUDIO_OK;
@@ -224,10 +263,8 @@
 }
 
 aaudio_result_t AudioStreamTrack::requestStart() {
-    std::lock_guard<std::mutex> lock(mStreamMutex);
-
     if (mAudioTrack.get() == nullptr) {
-        ALOGE("AudioStreamTrack::requestStart() no AudioTrack");
+        ALOGE("requestStart() no AudioTrack");
         return AAUDIO_ERROR_INVALID_STATE;
     }
     // Get current position so we can detect when the track is playing.
@@ -236,73 +273,62 @@
         return AAudioConvert_androidToAAudioResult(err);
     }
 
+    // Enable callback before starting AudioTrack to avoid shutting
+    // down because of a race condition.
+    mCallbackEnabled.store(true);
     err = mAudioTrack->start();
     if (err != OK) {
         return AAudioConvert_androidToAAudioResult(err);
     } else {
-        onStart();
         setState(AAUDIO_STREAM_STATE_STARTING);
     }
     return AAUDIO_OK;
 }
 
 aaudio_result_t AudioStreamTrack::requestPause() {
-    std::lock_guard<std::mutex> lock(mStreamMutex);
-
     if (mAudioTrack.get() == nullptr) {
         ALOGE("requestPause() no AudioTrack");
         return AAUDIO_ERROR_INVALID_STATE;
-    } else if (getState() != AAUDIO_STREAM_STATE_STARTING
-            && getState() != AAUDIO_STREAM_STATE_STARTED) {
-        ALOGE("requestPause(), called when state is %s",
-              AAudio_convertStreamStateToText(getState()));
-        return AAUDIO_ERROR_INVALID_STATE;
     }
-    onStop();
+
     setState(AAUDIO_STREAM_STATE_PAUSING);
     mAudioTrack->pause();
-    checkForDisconnectRequest();
+    mCallbackEnabled.store(false);
     status_t err = mAudioTrack->getPosition(&mPositionWhenPausing);
     if (err != OK) {
         return AAudioConvert_androidToAAudioResult(err);
     }
-    return AAUDIO_OK;
+    return checkForDisconnectRequest(false);
 }
 
 aaudio_result_t AudioStreamTrack::requestFlush() {
-    std::lock_guard<std::mutex> lock(mStreamMutex);
-
     if (mAudioTrack.get() == nullptr) {
-        ALOGE("AudioStreamTrack::requestFlush() no AudioTrack");
-        return AAUDIO_ERROR_INVALID_STATE;
-    } else if (getState() != AAUDIO_STREAM_STATE_PAUSED) {
-        ALOGE("AudioStreamTrack::requestFlush() not paused");
+        ALOGE("requestFlush() no AudioTrack");
         return AAUDIO_ERROR_INVALID_STATE;
     }
+
     setState(AAUDIO_STREAM_STATE_FLUSHING);
     incrementFramesRead(getFramesWritten() - getFramesRead());
     mAudioTrack->flush();
-    mFramesWritten.reset32();
+    mFramesRead.reset32(); // service reads frames, service position reset on flush
     mTimestampPosition.reset32();
     return AAUDIO_OK;
 }
 
 aaudio_result_t AudioStreamTrack::requestStop() {
-    std::lock_guard<std::mutex> lock(mStreamMutex);
-
     if (mAudioTrack.get() == nullptr) {
-        ALOGE("AudioStreamTrack::requestStop() no AudioTrack");
+        ALOGE("requestStop() no AudioTrack");
         return AAUDIO_ERROR_INVALID_STATE;
     }
-    onStop();
+
     setState(AAUDIO_STREAM_STATE_STOPPING);
     incrementFramesRead(getFramesWritten() - getFramesRead()); // TODO review
     mTimestampPosition.set(getFramesWritten());
-    mFramesWritten.reset32();
+    mFramesRead.reset32(); // service reads frames, service position reset on stop
     mTimestampPosition.reset32();
     mAudioTrack->stop();
-    checkForDisconnectRequest();
-    return AAUDIO_OK;
+    mCallbackEnabled.store(false);
+    return checkForDisconnectRequest(false);;
 }
 
 aaudio_result_t AudioStreamTrack::updateStateMachine()

diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.h b/media/libaaudio/src/legacy/AudioStreamTrack.h
index a871db4..68608de 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.h
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.h

@@ -48,6 +48,16 @@
     aaudio_result_t requestFlush() override;
     aaudio_result_t requestStop() override;
 
+    bool isFlushSupported() const override {
+        // Only implement FLUSH for OUTPUT streams.
+        return true;
+    }
+
+    bool isPauseSupported() const override {
+        // Only implement PAUSE for OUTPUT streams.
+        return true;
+    }
+
     aaudio_result_t getTimestamp(clockid_t clockId,
                                        int64_t *framePosition,
                                        int64_t *timeNanoseconds) override;

diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index 2450920..40ebb76 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp

@@ -25,6 +25,9 @@
 
 #include "aaudio/AAudio.h"
 #include <aaudio/AAudioTesting.h>
+#include <math.h>
+#include <system/audio-base.h>
+#include <assert.h>
 
 #include "utility/AAudioUtilities.h"
 
@@ -50,50 +53,21 @@
     return size;
 }
 
-
 // TODO expose and call clamp16_from_float function in primitives.h
 static inline int16_t clamp16_from_float(float f) {
-    /* Offset is used to expand the valid range of [-1.0, 1.0) into the 16 lsbs of the
-     * floating point significand. The normal shift is 3<<22, but the -15 offset
-     * is used to multiply by 32768.
-     */
-    static const float offset = (float)(3 << (22 - 15));
-    /* zero = (0x10f << 22) =  0x43c00000 (not directly used) */
-    static const int32_t limneg = (0x10f << 22) /*zero*/ - 32768; /* 0x43bf8000 */
-    static const int32_t limpos = (0x10f << 22) /*zero*/ + 32767; /* 0x43c07fff */
-
-    union {
-        float f;
-        int32_t i;
-    } u;
-
-    u.f = f + offset; /* recenter valid range */
-    /* Now the valid range is represented as integers between [limneg, limpos].
-     * Clamp using the fact that float representation (as an integer) is an ordered set.
-     */
-    if (u.i < limneg)
-        u.i = -32768;
-    else if (u.i > limpos)
-        u.i = 32767;
-    return u.i; /* Return lower 16 bits, the part of interest in the significand. */
+    static const float scale = 1 << 15;
+    return (int16_t) roundf(fmaxf(fminf(f * scale, scale - 1.f), -scale));
 }
 
-// Same but without clipping.
-// Convert -1.0f to +1.0f to -32768 to +32767
-static inline int16_t floatToInt16(float f) {
-    static const float offset = (float)(3 << (22 - 15));
-    union {
-        float f;
-        int32_t i;
-    } u;
-    u.f = f + offset; /* recenter valid range */
-    return u.i; /* Return lower 16 bits, the part of interest in the significand. */
+// Clip to valid range of a float sample to prevent excessive volume.
+// By using fmin and fmax we also protect against NaN.
+static float clipToMinMaxHeadroom(float input) {
+    return fmin(MAX_HEADROOM, fmax(MIN_HEADROOM, input));
 }
 
 static float clipAndClampFloatToPcm16(float sample, float scaler) {
     // Clip to valid range of a float sample to prevent excessive volume.
-    if (sample > MAX_HEADROOM) sample = MAX_HEADROOM;
-    else if (sample < MIN_HEADROOM) sample = MIN_HEADROOM;
+    sample = clipToMinMaxHeadroom(sample);
 
     // Scale and convert to a short.
     float fval = sample * scaler;
@@ -104,7 +78,7 @@
                                 int16_t *destination,
                                 int32_t numSamples,
                                 float amplitude) {
-    float scaler = amplitude;
+    const float scaler = amplitude;
     for (int i = 0; i < numSamples; i++) {
         float sample = *source++;
         *destination++ = clipAndClampFloatToPcm16(sample, scaler);
@@ -135,7 +109,7 @@
                                 float *destination,
                                 int32_t numSamples,
                                 float amplitude) {
-    float scaler = amplitude / SHORT_SCALE;
+    const float scaler = amplitude / SHORT_SCALE;
     for (int i = 0; i < numSamples; i++) {
         destination[i] = source[i] * scaler;
     }
@@ -149,7 +123,7 @@
                                 float amplitude1,
                                 float amplitude2) {
     float scaler = amplitude1 / SHORT_SCALE;
-    float delta = (amplitude2 - amplitude1) / (SHORT_SCALE * (float) numFrames);
+    const float delta = (amplitude2 - amplitude1) / (SHORT_SCALE * (float) numFrames);
     for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
         for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
             *destination++ = *source++ * scaler;
@@ -158,6 +132,7 @@
     }
 }
 
+
 // This code assumes amplitude1 and amplitude2 are between 0.0 and 1.0
 void AAudio_linearRamp(const float *source,
                        float *destination,
@@ -166,14 +141,12 @@
                        float amplitude1,
                        float amplitude2) {
     float scaler = amplitude1;
-    float delta = (amplitude2 - amplitude1) / numFrames;
+    const float delta = (amplitude2 - amplitude1) / numFrames;
     for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
         for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
             float sample = *source++;
-
             // Clip to valid range of a float sample to prevent excessive volume.
-            if (sample > MAX_HEADROOM) sample = MAX_HEADROOM;
-            else if (sample < MIN_HEADROOM) sample = MIN_HEADROOM;
+            sample = clipToMinMaxHeadroom(sample);
 
             *destination++ = sample * scaler;
         }
@@ -188,18 +161,267 @@
                        int32_t samplesPerFrame,
                        float amplitude1,
                        float amplitude2) {
-    float scaler = amplitude1 / SHORT_SCALE;
-    float delta = (amplitude2 - amplitude1) / (SHORT_SCALE * (float) numFrames);
+    // Because we are converting from int16 to 1nt16, we do not have to scale by 1/32768.
+    float scaler = amplitude1;
+    const float delta = (amplitude2 - amplitude1) / numFrames;
     for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
         for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
             // No need to clip because int16_t range is inherently limited.
             float sample =  *source++ * scaler;
-            *destination++ =  floatToInt16(sample);
+            *destination++ = (int16_t) roundf(sample);
         }
         scaler += delta;
     }
 }
 
+// *************************************************************************************
+// Convert Mono To Stereo at the same time as converting format.
+void AAudioConvert_formatMonoToStereo(const float *source,
+                                      int16_t *destination,
+                                      int32_t numFrames,
+                                      float amplitude) {
+    const float scaler = amplitude;
+    for (int i = 0; i < numFrames; i++) {
+        float sample = *source++;
+        int16_t sample16 = clipAndClampFloatToPcm16(sample, scaler);
+        *destination++ = sample16;
+        *destination++ = sample16;
+    }
+}
+
+void AAudioConvert_formatMonoToStereo(const float *source,
+                                      int16_t *destination,
+                                      int32_t numFrames,
+                                      float amplitude1,
+                                      float amplitude2) {
+    // divide by numFrames so that we almost reach amplitude2
+    const float delta = (amplitude2 - amplitude1) / numFrames;
+    for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+        const float scaler = amplitude1 + (frameIndex * delta);
+        const float sample = *source++;
+        int16_t sample16 = clipAndClampFloatToPcm16(sample, scaler);
+        *destination++ = sample16;
+        *destination++ = sample16;
+    }
+}
+
+void AAudioConvert_formatMonoToStereo(const int16_t *source,
+                                      float *destination,
+                                      int32_t numFrames,
+                                      float amplitude) {
+    const float scaler = amplitude / SHORT_SCALE;
+    for (int i = 0; i < numFrames; i++) {
+        float sample = source[i] * scaler;
+        *destination++ = sample;
+        *destination++ = sample;
+    }
+}
+
+// This code assumes amplitude1 and amplitude2 are between 0.0 and 1.0
+void AAudioConvert_formatMonoToStereo(const int16_t *source,
+                                      float *destination,
+                                      int32_t numFrames,
+                                      float amplitude1,
+                                      float amplitude2) {
+    const float scaler1 = amplitude1 / SHORT_SCALE;
+    const float delta = (amplitude2 - amplitude1) / (SHORT_SCALE * (float) numFrames);
+    for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+        float scaler = scaler1 + (frameIndex * delta);
+        float sample = source[frameIndex] * scaler;
+        *destination++ = sample;
+        *destination++ = sample;
+    }
+}
+
+// This code assumes amplitude1 and amplitude2 are between 0.0 and 1.0
+void AAudio_linearRampMonoToStereo(const float *source,
+                                   float *destination,
+                                   int32_t numFrames,
+                                   float amplitude1,
+                                   float amplitude2) {
+    const float delta = (amplitude2 - amplitude1) / numFrames;
+    for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+        float sample = *source++;
+
+        // Clip to valid range of a float sample to prevent excessive volume.
+        sample = clipToMinMaxHeadroom(sample);
+
+        const float scaler = amplitude1 + (frameIndex * delta);
+        float sampleScaled = sample * scaler;
+        *destination++ = sampleScaled;
+        *destination++ = sampleScaled;
+    }
+}
+
+// This code assumes amplitude1 and amplitude2 are between 0.0 and 1.0
+void AAudio_linearRampMonoToStereo(const int16_t *source,
+                                   int16_t *destination,
+                                   int32_t numFrames,
+                                   float amplitude1,
+                                   float amplitude2) {
+    // Because we are converting from int16 to 1nt16, we do not have to scale by 1/32768.
+    const float delta = (amplitude2 - amplitude1) / numFrames;
+    for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+        const float scaler = amplitude1 + (frameIndex * delta);
+        // No need to clip because int16_t range is inherently limited.
+        const float sample =  *source++ * scaler;
+        int16_t sample16 = (int16_t) roundf(sample);
+        *destination++ = sample16;
+        *destination++ = sample16;
+    }
+}
+
+// *************************************************************************************
+void AAudioDataConverter::convert(
+        const FormattedData &source,
+        const FormattedData &destination,
+        int32_t numFrames,
+        float levelFrom,
+        float levelTo) {
+
+    if (source.channelCount == 1 && destination.channelCount == 2) {
+        convertMonoToStereo(source,
+                            destination,
+                            numFrames,
+                            levelFrom,
+                            levelTo);
+    } else {
+        // We only support mono to stereo conversion. Otherwise source and destination
+        // must match.
+        assert(source.channelCount == destination.channelCount);
+        convertChannelsMatch(source,
+                             destination,
+                             numFrames,
+                             levelFrom,
+                             levelTo);
+    }
+}
+
+void AAudioDataConverter::convertMonoToStereo(
+        const FormattedData &source,
+        const FormattedData &destination,
+        int32_t numFrames,
+        float levelFrom,
+        float levelTo) {
+
+    // The formats are validated when the stream is opened so we do not have to
+    // check for illegal combinations here.
+    if (source.format == AAUDIO_FORMAT_PCM_FLOAT) {
+        if (destination.format == AAUDIO_FORMAT_PCM_FLOAT) {
+            AAudio_linearRampMonoToStereo(
+                    (const float *) source.data,
+                    (float *) destination.data,
+                    numFrames,
+                    levelFrom,
+                    levelTo);
+        } else if (destination.format == AAUDIO_FORMAT_PCM_I16) {
+            if (levelFrom != levelTo) {
+                AAudioConvert_formatMonoToStereo(
+                        (const float *) source.data,
+                        (int16_t *) destination.data,
+                        numFrames,
+                        levelFrom,
+                        levelTo);
+            } else {
+                AAudioConvert_formatMonoToStereo(
+                        (const float *) source.data,
+                        (int16_t *) destination.data,
+                        numFrames,
+                        levelTo);
+            }
+        }
+    } else if (source.format == AAUDIO_FORMAT_PCM_I16) {
+        if (destination.format == AAUDIO_FORMAT_PCM_FLOAT) {
+            if (levelFrom != levelTo) {
+                AAudioConvert_formatMonoToStereo(
+                        (const int16_t *) source.data,
+                        (float *) destination.data,
+                        numFrames,
+                        levelFrom,
+                        levelTo);
+            } else {
+                AAudioConvert_formatMonoToStereo(
+                        (const int16_t *) source.data,
+                        (float *) destination.data,
+                        numFrames,
+                        levelTo);
+            }
+        } else if (destination.format == AAUDIO_FORMAT_PCM_I16) {
+            AAudio_linearRampMonoToStereo(
+                    (const int16_t *) source.data,
+                    (int16_t *) destination.data,
+                    numFrames,
+                    levelFrom,
+                    levelTo);
+        }
+    }
+}
+
+void AAudioDataConverter::convertChannelsMatch(
+        const FormattedData &source,
+        const FormattedData &destination,
+        int32_t numFrames,
+        float levelFrom,
+        float levelTo) {
+    const int32_t numSamples = numFrames * source.channelCount;
+
+    // The formats are validated when the stream is opened so we do not have to
+    // check for illegal combinations here.
+    if (source.format == AAUDIO_FORMAT_PCM_FLOAT) {
+        if (destination.format == AAUDIO_FORMAT_PCM_FLOAT) {
+            AAudio_linearRamp(
+                    (const float *) source.data,
+                    (float *) destination.data,
+                    numFrames,
+                    source.channelCount,
+                    levelFrom,
+                    levelTo);
+        } else if (destination.format == AAUDIO_FORMAT_PCM_I16) {
+            if (levelFrom != levelTo) {
+                AAudioConvert_floatToPcm16(
+                        (const float *) source.data,
+                        (int16_t *) destination.data,
+                        numFrames,
+                        source.channelCount,
+                        levelFrom,
+                        levelTo);
+            } else {
+                AAudioConvert_floatToPcm16(
+                        (const float *) source.data,
+                        (int16_t *) destination.data,
+                        numSamples,
+                        levelTo);
+            }
+        }
+    } else if (source.format == AAUDIO_FORMAT_PCM_I16) {
+        if (destination.format == AAUDIO_FORMAT_PCM_FLOAT) {
+            if (levelFrom != levelTo) {
+                AAudioConvert_pcm16ToFloat(
+                        (const int16_t *) source.data,
+                        (float *) destination.data,
+                        numFrames,
+                        source.channelCount,
+                        levelFrom,
+                        levelTo);
+            } else {
+                AAudioConvert_pcm16ToFloat(
+                        (const int16_t *) source.data,
+                        (float *) destination.data,
+                        numSamples,
+                        levelTo);
+            }
+        } else if (destination.format == AAUDIO_FORMAT_PCM_I16) {
+            AAudio_linearRamp(
+                    (const int16_t *) source.data,
+                    (int16_t *) destination.data,
+                    numFrames,
+                    source.channelCount,
+                    levelFrom,
+                    levelTo);
+        }
+    }
+}
+
 status_t AAudioConvert_aaudioToAndroidStatus(aaudio_result_t result) {
     // This covers the case for AAUDIO_OK and for positive results.
     if (result >= 0) {
@@ -281,6 +503,13 @@
     return result;
 }
 
+audio_session_t AAudioConvert_aaudioToAndroidSessionId(aaudio_session_id_t sessionId) {
+    // If not a regular sessionId then convert to a safe value of AUDIO_SESSION_ALLOCATE.
+    return (sessionId == AAUDIO_SESSION_ID_ALLOCATE || sessionId == AAUDIO_SESSION_ID_NONE)
+           ? AUDIO_SESSION_ALLOCATE
+           : (audio_session_t) sessionId;
+}
+
 audio_format_t AAudioConvert_aaudioToAndroidDataFormat(aaudio_format_t aaudioFormat) {
     audio_format_t androidFormat;
     switch (aaudioFormat) {
@@ -315,17 +544,77 @@
     return aaudioFormat;
 }
 
+// Make a message string from the condition.
+#define STATIC_ASSERT(condition) static_assert(condition, #condition)
+
+audio_usage_t AAudioConvert_usageToInternal(aaudio_usage_t usage) {
+    // The public aaudio_content_type_t constants are supposed to have the same
+    // values as the internal audio_content_type_t values.
+    STATIC_ASSERT(AAUDIO_USAGE_MEDIA == AUDIO_USAGE_MEDIA);
+    STATIC_ASSERT(AAUDIO_USAGE_VOICE_COMMUNICATION == AUDIO_USAGE_VOICE_COMMUNICATION);
+    STATIC_ASSERT(AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING
+                  == AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING);
+    STATIC_ASSERT(AAUDIO_USAGE_ALARM == AUDIO_USAGE_ALARM);
+    STATIC_ASSERT(AAUDIO_USAGE_NOTIFICATION == AUDIO_USAGE_NOTIFICATION);
+    STATIC_ASSERT(AAUDIO_USAGE_NOTIFICATION_RINGTONE
+                  == AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE);
+    STATIC_ASSERT(AAUDIO_USAGE_NOTIFICATION_EVENT == AUDIO_USAGE_NOTIFICATION_EVENT);
+    STATIC_ASSERT(AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY == AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY);
+    STATIC_ASSERT(AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE
+                  == AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE);
+    STATIC_ASSERT(AAUDIO_USAGE_ASSISTANCE_SONIFICATION == AUDIO_USAGE_ASSISTANCE_SONIFICATION);
+    STATIC_ASSERT(AAUDIO_USAGE_GAME == AUDIO_USAGE_GAME);
+    STATIC_ASSERT(AAUDIO_USAGE_ASSISTANT == AUDIO_USAGE_ASSISTANT);
+    if (usage == AAUDIO_UNSPECIFIED) {
+        usage = AAUDIO_USAGE_MEDIA;
+    }
+    return (audio_usage_t) usage; // same value
+}
+
+audio_content_type_t AAudioConvert_contentTypeToInternal(aaudio_content_type_t contentType) {
+    // The public aaudio_content_type_t constants are supposed to have the same
+    // values as the internal audio_content_type_t values.
+    STATIC_ASSERT(AAUDIO_CONTENT_TYPE_MUSIC == AUDIO_CONTENT_TYPE_MUSIC);
+    STATIC_ASSERT(AAUDIO_CONTENT_TYPE_SPEECH == AUDIO_CONTENT_TYPE_SPEECH);
+    STATIC_ASSERT(AAUDIO_CONTENT_TYPE_SONIFICATION == AUDIO_CONTENT_TYPE_SONIFICATION);
+    STATIC_ASSERT(AAUDIO_CONTENT_TYPE_MOVIE == AUDIO_CONTENT_TYPE_MOVIE);
+    if (contentType == AAUDIO_UNSPECIFIED) {
+        contentType = AAUDIO_CONTENT_TYPE_MUSIC;
+    }
+    return (audio_content_type_t) contentType; // same value
+}
+
+audio_source_t AAudioConvert_inputPresetToAudioSource(aaudio_input_preset_t preset) {
+    // The public aaudio_input_preset_t constants are supposed to have the same
+    // values as the internal audio_source_t values.
+    STATIC_ASSERT(AAUDIO_UNSPECIFIED == AUDIO_SOURCE_DEFAULT);
+    STATIC_ASSERT(AAUDIO_INPUT_PRESET_GENERIC == AUDIO_SOURCE_MIC);
+    STATIC_ASSERT(AAUDIO_INPUT_PRESET_CAMCORDER == AUDIO_SOURCE_CAMCORDER);
+    STATIC_ASSERT(AAUDIO_INPUT_PRESET_VOICE_RECOGNITION == AUDIO_SOURCE_VOICE_RECOGNITION);
+    STATIC_ASSERT(AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION == AUDIO_SOURCE_VOICE_COMMUNICATION);
+    STATIC_ASSERT(AAUDIO_INPUT_PRESET_UNPROCESSED == AUDIO_SOURCE_UNPROCESSED);
+    if (preset == AAUDIO_UNSPECIFIED) {
+        preset = AAUDIO_INPUT_PRESET_VOICE_RECOGNITION;
+    }
+    return (audio_source_t) preset; // same value
+}
+
 int32_t AAudioConvert_framesToBytes(int32_t numFrames,
-                                            int32_t bytesPerFrame,
-                                            int32_t *sizeInBytes) {
-    // TODO implement more elegantly
-    const int32_t maxChannels = 256; // ridiculously large
-    const int32_t maxBytesPerFrame = maxChannels * sizeof(float);
-    // Prevent overflow by limiting multiplicands.
-    if (bytesPerFrame > maxBytesPerFrame || numFrames > (0x3FFFFFFF / maxBytesPerFrame)) {
-        ALOGE("size overflow, numFrames = %d, frameSize = %zd", numFrames, bytesPerFrame);
+                                    int32_t bytesPerFrame,
+                                    int32_t *sizeInBytes) {
+    *sizeInBytes = 0;
+
+    if (numFrames < 0 || bytesPerFrame < 0) {
+        ALOGE("negative size, numFrames = %d, frameSize = %d", numFrames, bytesPerFrame);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }
+
+    // Prevent numeric overflow.
+    if (numFrames > (INT32_MAX / bytesPerFrame)) {
+        ALOGE("size overflow, numFrames = %d, frameSize = %d", numFrames, bytesPerFrame);
+        return AAUDIO_ERROR_OUT_OF_RANGE;
+    }
+
     *sizeInBytes = numFrames * bytesPerFrame;
     return AAUDIO_OK;
 }
@@ -410,3 +699,31 @@
     }
     return prop;
 }
+
+aaudio_result_t AAudio_isFlushAllowed(aaudio_stream_state_t state) {
+    aaudio_result_t result = AAUDIO_OK;
+    switch (state) {
+// Proceed with flushing.
+        case AAUDIO_STREAM_STATE_OPEN:
+        case AAUDIO_STREAM_STATE_PAUSED:
+        case AAUDIO_STREAM_STATE_STOPPED:
+        case AAUDIO_STREAM_STATE_FLUSHED:
+            break;
+
+// Transition from one inactive state to another.
+        case AAUDIO_STREAM_STATE_STARTING:
+        case AAUDIO_STREAM_STATE_STARTED:
+        case AAUDIO_STREAM_STATE_STOPPING:
+        case AAUDIO_STREAM_STATE_PAUSING:
+        case AAUDIO_STREAM_STATE_FLUSHING:
+        case AAUDIO_STREAM_STATE_CLOSING:
+        case AAUDIO_STREAM_STATE_CLOSED:
+        case AAUDIO_STREAM_STATE_DISCONNECTED:
+        default:
+            ALOGE("can only flush stream when PAUSED, OPEN or STOPPED, state = %s",
+                  AAudio_convertStreamStateToText(state));
+            result =  AAUDIO_ERROR_INVALID_STATE;
+            break;
+    }
+    return result;
+}

diff --git a/media/libaaudio/src/utility/AAudioUtilities.h b/media/libaaudio/src/utility/AAudioUtilities.h
index 3afa976..4b975e8 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.h
+++ b/media/libaaudio/src/utility/AAudioUtilities.h

@@ -23,7 +23,7 @@
 #include <sys/types.h>
 
 #include <utils/Errors.h>
-#include <hardware/audio.h>
+#include <system/audio.h>
 
 #include "aaudio/AAudio.h"
 
@@ -38,6 +38,13 @@
 aaudio_result_t AAudioConvert_androidToAAudioResult(android::status_t status);
 
 /**
+ * Convert an aaudio_session_id_t to a value that is safe to pass to AudioFlinger.
+ * @param sessionId
+ * @return safe value
+ */
+audio_session_t AAudioConvert_aaudioToAndroidSessionId(aaudio_session_id_t sessionId);
+
+/**
  * Convert an array of floats to an array of int16_t.
  *
  * @param source
@@ -152,21 +159,81 @@
                        float amplitude1,
                        float amplitude2);
 
+class AAudioDataConverter {
+public:
+
+    struct FormattedData {
+
+        FormattedData(void *data, aaudio_format_t format, int32_t channelCount)
+            : data(data)
+            , format(format)
+            , channelCount(channelCount) {}
+
+        const void            *data = nullptr;
+        const aaudio_format_t  format = AAUDIO_FORMAT_UNSPECIFIED;
+        const int32_t          channelCount = 1;
+    };
+
+    static void convert(const FormattedData &source,
+                        const FormattedData &destination,
+                        int32_t numFrames,
+                        float levelFrom,
+                        float levelTo);
+
+private:
+    static void convertMonoToStereo(const FormattedData &source,
+                                    const FormattedData &destination,
+                                    int32_t numFrames,
+                                    float levelFrom,
+                                    float levelTo);
+
+    static void convertChannelsMatch(const FormattedData &source,
+                                     const FormattedData &destination,
+                                     int32_t numFrames,
+                                     float levelFrom,
+                                     float levelTo);
+};
+
 /**
  * Calculate the number of bytes and prevent numeric overflow.
+ * The *sizeInBytes will be set to zero if there is an error.
+ *
  * @param numFrames frame count
  * @param bytesPerFrame size of a frame in bytes
- * @param sizeInBytes total size in bytes
+ * @param sizeInBytes pointer to a variable to receive total size in bytes
  * @return AAUDIO_OK or negative error, eg. AAUDIO_ERROR_OUT_OF_RANGE
  */
 int32_t AAudioConvert_framesToBytes(int32_t numFrames,
-                                            int32_t bytesPerFrame,
-                                            int32_t *sizeInBytes);
+                                    int32_t bytesPerFrame,
+                                    int32_t *sizeInBytes);
 
 audio_format_t AAudioConvert_aaudioToAndroidDataFormat(aaudio_format_t aaudio_format);
 
 aaudio_format_t AAudioConvert_androidToAAudioDataFormat(audio_format_t format);
 
+
+/**
+ * Note that this function does not validate the passed in value.
+ * That is done somewhere else.
+ * @return internal value
+ */
+
+audio_usage_t AAudioConvert_usageToInternal(aaudio_usage_t usage);
+
+/**
+ * Note that this function does not validate the passed in value.
+ * That is done somewhere else.
+ * @return internal value
+ */
+audio_content_type_t AAudioConvert_contentTypeToInternal(aaudio_content_type_t contentType);
+
+/**
+ * Note that this function does not validate the passed in value.
+ * That is done somewhere else.
+ * @return internal audio source
+ */
+audio_source_t AAudioConvert_inputPresetToAudioSource(aaudio_input_preset_t preset);
+
 /**
  * @return the size of a sample of the given format in bytes or AAUDIO_ERROR_ILLEGAL_ARGUMENT
  */
@@ -235,6 +302,14 @@
  */
 int32_t AAudioProperty_getHardwareBurstMinMicros();
 
+
+/**
+ * Is flush allowed for the given state?
+ * @param state
+ * @return AAUDIO_OK if allowed or an error
+ */
+aaudio_result_t AAudio_isFlushAllowed(aaudio_stream_state_t state);
+
 /**
  * Try a function f until it returns true.
  *

diff --git a/media/libaaudio/src/utility/AudioClock.h b/media/libaaudio/src/utility/AudioClock.h
index 43b71b0..d5d4ef4 100644
--- a/media/libaaudio/src/utility/AudioClock.h
+++ b/media/libaaudio/src/utility/AudioClock.h

@@ -17,6 +17,7 @@
 #ifndef UTILITY_AUDIO_CLOCK_H
 #define UTILITY_AUDIO_CLOCK_H
 
+#include <errno.h>
 #include <stdint.h>
 #include <time.h>
 

diff --git a/media/libaaudio/src/utility/LinearRamp.h b/media/libaaudio/src/utility/LinearRamp.h
index ff09dce..2b1b8e0 100644
--- a/media/libaaudio/src/utility/LinearRamp.h
+++ b/media/libaaudio/src/utility/LinearRamp.h

@@ -87,7 +87,7 @@
 
     std::atomic<float>   mTarget;
 
-    int32_t mLengthInFrames  = 48000 / 50; // 20 msec at 48000 Hz
+    int32_t mLengthInFrames  = 48000 / 100; // 10 msec at 48000 Hz
     int32_t mRemaining       = 0;
     float   mLevelFrom       = 0.0f;
     float   mLevelTo         = 0.0f;

diff --git a/media/libaaudio/tests/Android.bp b/media/libaaudio/tests/Android.bp
new file mode 100644
index 0000000..68194db
--- /dev/null
+++ b/media/libaaudio/tests/Android.bp

@@ -0,0 +1,169 @@
+cc_defaults {
+    name: "libaaudio_tests_defaults",
+    cflags: [
+        "-Wall",
+        "-Werror",
+    ],
+}
+
+cc_test {
+    name: "test_aaudio_marshalling",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_marshalling.cpp"],
+    shared_libs: [
+        "libaaudio",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+}
+
+cc_test {
+    name: "test_block_adapter",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_block_adapter.cpp"],
+    shared_libs: ["libaaudio"],
+}
+
+cc_test {
+    name: "test_timestamps",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_timestamps.cpp"],
+    header_libs: ["libaaudio_example_utils"],
+    shared_libs: ["libaaudio"],
+}
+
+cc_test {
+    name: "test_linear_ramp",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_linear_ramp.cpp"],
+    shared_libs: ["libaaudio"],
+}
+
+cc_test {
+    name: "test_open_params",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_open_params.cpp"],
+    shared_libs: [
+        "libaaudio",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+}
+
+cc_test {
+    name: "test_no_close",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_no_close.cpp"],
+    shared_libs: [
+        "libaaudio",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+}
+
+cc_test {
+    name: "test_aaudio_recovery",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_recovery.cpp"],
+    shared_libs: [
+        "libaaudio",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+}
+
+cc_test {
+    name: "test_n_streams",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_n_streams.cpp"],
+    shared_libs: [
+        "libaaudio",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+}
+
+cc_test {
+    name: "test_bad_disconnect",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_bad_disconnect.cpp"],
+    shared_libs: [
+        "libaaudio",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+}
+
+cc_test {
+    name: "test_various",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_various.cpp"],
+    shared_libs: [
+        "libaaudio",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+}
+
+cc_test {
+    name: "test_session_id",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_session_id.cpp"],
+    shared_libs: [
+        "libaaudio",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+}
+
+cc_test {
+    name: "test_aaudio_monkey",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_aaudio_monkey.cpp"],
+    header_libs: ["libaaudio_example_utils"],
+    shared_libs: [
+        "libaaudio",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+}
+
+cc_test {
+    name: "test_attributes",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_attributes.cpp"],
+    shared_libs: [
+        "libaaudio",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+}
+
+cc_test {
+    name: "test_interference",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_interference.cpp"],
+    shared_libs: [
+        "libaaudio",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+}
+
+cc_test {
+    name: "test_atomic_fifo",
+    defaults: ["libaaudio_tests_defaults"],
+    srcs: ["test_atomic_fifo.cpp"],
+    shared_libs: ["libaaudio"],
+}

diff --git a/media/libaaudio/tests/Android.mk b/media/libaaudio/tests/Android.mk
deleted file mode 100644
index 8117181..0000000
--- a/media/libaaudio/tests/Android.mk
+++ /dev/null

@@ -1,92 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/src
-LOCAL_SRC_FILES:= test_marshalling.cpp
-LOCAL_SHARED_LIBRARIES := libaaudio libbinder libcutils libutils
-LOCAL_MODULE := test_aaudio_marshalling
-include $(BUILD_NATIVE_TEST)
-
-include $(CLEAR_VARS)
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/src
-LOCAL_SRC_FILES:= test_block_adapter.cpp
-LOCAL_SHARED_LIBRARIES := libaaudio
-LOCAL_MODULE := test_block_adapter
-include $(BUILD_NATIVE_TEST)
-
-include $(CLEAR_VARS)
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/src \
-    frameworks/av/media/libaaudio/examples
-LOCAL_SRC_FILES:= test_timestamps.cpp
-LOCAL_SHARED_LIBRARIES := libaaudio
-LOCAL_MODULE := test_timestamps
-include $(BUILD_NATIVE_TEST)
-
-include $(CLEAR_VARS)
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/src
-LOCAL_SRC_FILES:= test_linear_ramp.cpp
-LOCAL_SHARED_LIBRARIES := libaaudio
-LOCAL_MODULE := test_linear_ramp
-include $(BUILD_NATIVE_TEST)
-
-include $(CLEAR_VARS)
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/src
-LOCAL_SRC_FILES:= test_open_params.cpp
-LOCAL_SHARED_LIBRARIES := libaaudio libbinder libcutils libutils
-LOCAL_MODULE := test_open_params
-include $(BUILD_NATIVE_TEST)
-
-include $(CLEAR_VARS)
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/src
-LOCAL_SRC_FILES:= test_no_close.cpp
-LOCAL_SHARED_LIBRARIES := libaaudio libbinder libcutils libutils
-LOCAL_MODULE := test_no_close
-include $(BUILD_NATIVE_TEST)
-
-include $(CLEAR_VARS)
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/src
-LOCAL_SRC_FILES:= test_recovery.cpp
-LOCAL_SHARED_LIBRARIES := libaaudio libbinder libcutils libutils
-LOCAL_MODULE := test_aaudio_recovery
-include $(BUILD_NATIVE_TEST)
-
-include $(CLEAR_VARS)
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/src
-LOCAL_SRC_FILES:= test_n_streams.cpp
-LOCAL_SHARED_LIBRARIES := libaaudio libbinder libcutils libutils
-LOCAL_MODULE := test_n_streams
-include $(BUILD_NATIVE_TEST)
-
-include $(CLEAR_VARS)
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/src
-LOCAL_SRC_FILES:= test_atomic_fifo.cpp
-LOCAL_SHARED_LIBRARIES := libaaudio libbinder libcutils libutils
-LOCAL_MODULE := test_atomic_fifo
-include $(BUILD_NATIVE_TEST)

diff --git a/media/libaaudio/tests/test_aaudio_monkey.cpp b/media/libaaudio/tests/test_aaudio_monkey.cpp
new file mode 100644
index 0000000..be54835
--- /dev/null
+++ b/media/libaaudio/tests/test_aaudio_monkey.cpp

@@ -0,0 +1,307 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Try to trigger bugs by playing randomly on multiple streams.
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <vector>
+
+#include <aaudio/AAudio.h>
+#include "AAudioArgsParser.h"
+#include "AAudioExampleUtils.h"
+#include "AAudioSimplePlayer.h"
+#include "SineGenerator.h"
+
+#define DEFAULT_TIMEOUT_NANOS  (1 * NANOS_PER_SECOND)
+
+#define NUM_LOOPS          1000
+#define MAX_MICROS_DELAY   (2 * 1000 * 1000)
+
+// TODO Consider adding an input stream.
+#define PROB_START   (0.20)
+#define PROB_PAUSE   (PROB_START + 0.10)
+#define PROB_FLUSH   (PROB_PAUSE + 0.10)
+#define PROB_STOP    (PROB_FLUSH + 0.10)
+#define PROB_CLOSE   (PROB_STOP + 0.10)
+static_assert(PROB_CLOSE < 0.9, "Probability sum too high.");
+
+aaudio_data_callback_result_t AAudioMonkeyDataCallback(
+        AAudioStream *stream,
+        void *userData,
+        void *audioData,
+        int32_t numFrames);
+
+void AAudioMonkeyErrorCallbackProc(
+        AAudioStream *stream __unused,
+        void *userData __unused,
+        aaudio_result_t error) {
+    printf("Error Callback, error: %d\n",(int)error);
+}
+
+// This function is not thread safe. Only use this from a single thread.
+double nextRandomDouble() {
+    return drand48();
+}
+
+class AAudioMonkey : public AAudioSimplePlayer {
+public:
+
+    AAudioMonkey(int index, AAudioArgsParser *argParser)
+            : mArgParser(argParser)
+            , mIndex(index) {}
+
+    aaudio_result_t open() {
+        printf("Monkey # %d ---------------------------------------------- OPEN\n", mIndex);
+        double offset = mIndex * 50;
+        mSine1.setup(440.0, 48000);
+        mSine1.setSweep(300.0 + offset, 600.0 + offset, 5.0);
+        mSine2.setup(660.0, 48000);
+        mSine2.setSweep(350.0 + offset, 900.0 + offset, 7.0);
+
+        aaudio_result_t result = AAudioSimplePlayer::open(*mArgParser,
+                                      AAudioMonkeyDataCallback,
+                                      AAudioMonkeyErrorCallbackProc,
+                                      this);
+        if (result != AAUDIO_OK) {
+            printf("ERROR -  player.open() returned %d\n", result);
+        }
+
+        mArgParser->compareWithStream(getStream());
+        return result;
+    }
+
+    bool isOpen() {
+        return (getStream() != nullptr);
+
+    }
+    /**
+     *
+     * @return true if stream passes tests
+     */
+    bool validate() {
+        if (!isOpen()) return true; // closed is OK
+
+        // update and query stream state
+        aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
+        aaudio_result_t result = AAudioStream_waitForStateChange(getStream(),
+            AAUDIO_STREAM_STATE_UNKNOWN, &state, 0);
+        if (result != AAUDIO_OK) {
+            printf("ERROR - AAudioStream_waitForStateChange returned %d\n", result);
+            return false;
+        }
+
+        int64_t framesRead = AAudioStream_getFramesRead(getStream());
+        int64_t framesWritten = AAudioStream_getFramesWritten(getStream());
+        int32_t xRuns = AAudioStream_getXRunCount(getStream());
+        // Print status
+        printf("%30s, framesWritten = %8lld, framesRead = %8lld, xRuns = %d\n",
+               AAudio_convertStreamStateToText(state),
+               (unsigned long long) framesWritten,
+               (unsigned long long) framesRead,
+               xRuns);
+
+        if (framesWritten < framesRead) {
+            printf("WARNING - UNDERFLOW - diff = %d !!!!!!!!!!!!\n",
+                   (int) (framesWritten - framesRead));
+        }
+        return true;
+    }
+
+    aaudio_result_t invoke() {
+        aaudio_result_t result = AAUDIO_OK;
+        if (!isOpen()) {
+            result = open();
+            if (result != AAUDIO_OK) return result;
+        }
+
+        if (!validate()) {
+            return -1;
+        }
+
+        double dice = nextRandomDouble();
+        // Select an action based on a weighted probability.
+        if (dice < PROB_START) {
+            printf("start\n");
+            result = AAudioStream_requestStart(getStream());
+        } else if (dice < PROB_PAUSE) {
+            printf("pause\n");
+            result = AAudioStream_requestPause(getStream());
+        } else if (dice < PROB_FLUSH) {
+            printf("flush\n");
+            result = AAudioStream_requestFlush(getStream());
+        } else if (dice < PROB_STOP) {
+            printf("stop\n");
+            result = AAudioStream_requestStop(getStream());
+        } else if (dice < PROB_CLOSE) {
+            printf("close\n");
+            result = close();
+        } else {
+            printf("do nothing\n");
+        }
+
+        if (result == AAUDIO_ERROR_INVALID_STATE) {
+            printf("    got AAUDIO_ERROR_INVALID_STATE - expected from a monkey\n");
+            result = AAUDIO_OK;
+        }
+        if (result == AAUDIO_OK && isOpen()) {
+            if (!validate()) {
+                result = -1;
+            }
+        }
+        return result;
+    }
+
+    aaudio_data_callback_result_t renderAudio(
+            AAudioStream *stream,
+            void *audioData,
+            int32_t numFrames) {
+
+        int32_t samplesPerFrame = AAudioStream_getChannelCount(stream);
+        // This code only plays on the first one or two channels.
+        // TODO Support arbitrary number of channels.
+        switch (AAudioStream_getFormat(stream)) {
+            case AAUDIO_FORMAT_PCM_I16: {
+                int16_t *audioBuffer = (int16_t *) audioData;
+                // Render sine waves as shorts to first channel.
+                mSine1.render(&audioBuffer[0], samplesPerFrame, numFrames);
+                // Render sine waves to second channel if there is one.
+                if (samplesPerFrame > 1) {
+                    mSine2.render(&audioBuffer[1], samplesPerFrame, numFrames);
+                }
+            }
+                break;
+            case AAUDIO_FORMAT_PCM_FLOAT: {
+                float *audioBuffer = (float *) audioData;
+                // Render sine waves as floats to first channel.
+                mSine1.render(&audioBuffer[0], samplesPerFrame, numFrames);
+                // Render sine waves to second channel if there is one.
+                if (samplesPerFrame > 1) {
+                    mSine2.render(&audioBuffer[1], samplesPerFrame, numFrames);
+                }
+            }
+                break;
+            default:
+                return AAUDIO_CALLBACK_RESULT_STOP;
+        }
+        return AAUDIO_CALLBACK_RESULT_CONTINUE;
+    }
+
+private:
+    const AAudioArgsParser  *mArgParser;
+    const int                mIndex;
+    SineGenerator            mSine1;
+    SineGenerator            mSine2;
+};
+
+// Callback function that fills the audio output buffer.
+aaudio_data_callback_result_t AAudioMonkeyDataCallback(
+        AAudioStream *stream,
+        void *userData,
+        void *audioData,
+        int32_t numFrames
+) {
+    // should not happen but just in case...
+    if (userData == nullptr) {
+        printf("ERROR - AAudioMonkeyDataCallback needs userData\n");
+        return AAUDIO_CALLBACK_RESULT_STOP;
+    }
+    AAudioMonkey *monkey = (AAudioMonkey *) userData;
+    return monkey->renderAudio(stream, audioData, numFrames);
+}
+
+
+static void usage() {
+    AAudioArgsParser::usage();
+    printf("      -i{seed}  Initial random seed\n");
+    printf("      -t{count} number of monkeys in the Troop\n");
+}
+
+int main(int argc, const char **argv) {
+    AAudioArgsParser argParser;
+    std::vector<AAudioMonkey> monkeys;
+    aaudio_result_t result;
+    int numMonkeys = 1;
+
+    // Make printf print immediately so that debug info is not stuck
+    // in a buffer if we hang or crash.
+    setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
+
+    printf("%s - Monkeys\n", argv[0]);
+
+    long int seed = (long int)getNanoseconds();  // different every time by default
+
+    for (int i = 1; i < argc; i++) {
+        const char *arg = argv[i];
+        if (argParser.parseArg(arg)) {
+            // Handle options that are not handled by the ArgParser
+            if (arg[0] == '-') {
+                char option = arg[1];
+                switch (option) {
+                    case 'i':
+                        seed = atol(&arg[2]);
+                        break;
+                    case 't':
+                        numMonkeys = atoi(&arg[2]);
+                        break;
+                    default:
+                        usage();
+                        exit(EXIT_FAILURE);
+                        break;
+                }
+            } else {
+                usage();
+                exit(EXIT_FAILURE);
+                break;
+            }
+        }
+    }
+
+    srand48(seed);
+    printf("seed = %ld, nextRandomDouble() = %f\n", seed, nextRandomDouble());
+
+    for (int m = 0; m < numMonkeys; m++) {
+        monkeys.emplace_back(m, &argParser);
+    }
+
+    for (int i = 0; i < NUM_LOOPS; i++) {
+        // pick a random monkey and invoke it
+        double dice = nextRandomDouble();
+        int monkeyIndex = floor(dice * numMonkeys);
+        printf("----------- Monkey #%d\n", monkeyIndex);
+        result = monkeys[monkeyIndex].invoke();
+        if (result != AAUDIO_OK) {
+            goto error;
+        }
+
+        // sleep some random time
+        dice = nextRandomDouble();
+        dice = dice * dice * dice; // skew towards smaller delays
+        int micros = (int) (dice * MAX_MICROS_DELAY);
+        usleep(micros);
+
+        // TODO consider making this multi-threaded, one thread per monkey, to catch more bugs
+    }
+
+    printf("PASS\n");
+    return EXIT_SUCCESS;
+
+error:
+    printf("FAIL - AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
+    usleep(1000 * 1000); // give me time to stop the logcat
+    return EXIT_FAILURE;
+}
+

diff --git a/media/libaaudio/tests/test_attributes.cpp b/media/libaaudio/tests/test_attributes.cpp
new file mode 100644
index 0000000..b01af25
--- /dev/null
+++ b/media/libaaudio/tests/test_attributes.cpp

@@ -0,0 +1,179 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Test AAudio attributes such as Usage, ContentType and InputPreset.
+
+#include <stdio.h>
+#include <unistd.h>
+
+#include <aaudio/AAudio.h>
+#include <gtest/gtest.h>
+
+constexpr int64_t kNanosPerSecond = 1000000000;
+constexpr int kNumFrames = 256;
+constexpr int kChannelCount = 2;
+
+constexpr int32_t DONT_SET = -1000;
+
+static void checkAttributes(aaudio_performance_mode_t perfMode,
+                            aaudio_usage_t usage,
+                            aaudio_content_type_t contentType,
+                            aaudio_input_preset_t preset = DONT_SET,
+                            aaudio_direction_t direction = AAUDIO_DIRECTION_OUTPUT) {
+
+    float *buffer = new float[kNumFrames * kChannelCount];
+
+    AAudioStreamBuilder *aaudioBuilder = nullptr;
+    AAudioStream *aaudioStream = nullptr;
+
+    // Use an AAudioStreamBuilder to contain requested parameters.
+    ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+
+    // Request stream properties.
+    AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
+    AAudioStreamBuilder_setDirection(aaudioBuilder, direction);
+
+    // Set the attribute in the builder.
+    if (usage != DONT_SET) {
+        AAudioStreamBuilder_setUsage(aaudioBuilder, usage);
+    }
+    if (contentType != DONT_SET) {
+        AAudioStreamBuilder_setContentType(aaudioBuilder, contentType);
+    }
+    if (preset != DONT_SET) {
+        AAudioStreamBuilder_setInputPreset(aaudioBuilder, preset);
+    }
+
+    // Create an AAudioStream using the Builder.
+    ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream));
+    AAudioStreamBuilder_delete(aaudioBuilder);
+
+    // Make sure we get the same attributes back from the stream.
+    aaudio_usage_t expectedUsage =
+            (usage == DONT_SET || usage == AAUDIO_UNSPECIFIED)
+            ? AAUDIO_USAGE_MEDIA // default
+            : usage;
+    EXPECT_EQ(expectedUsage, AAudioStream_getUsage(aaudioStream));
+
+    aaudio_content_type_t expectedContentType =
+            (contentType == DONT_SET || contentType == AAUDIO_UNSPECIFIED)
+            ? AAUDIO_CONTENT_TYPE_MUSIC // default
+            : contentType;
+    EXPECT_EQ(expectedContentType, AAudioStream_getContentType(aaudioStream));
+
+    aaudio_input_preset_t expectedPreset =
+            (preset == DONT_SET || preset == AAUDIO_UNSPECIFIED)
+            ? AAUDIO_INPUT_PRESET_VOICE_RECOGNITION // default
+            : preset;
+    EXPECT_EQ(expectedPreset, AAudioStream_getInputPreset(aaudioStream));
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_requestStart(aaudioStream));
+
+    if (direction == AAUDIO_DIRECTION_INPUT) {
+        EXPECT_EQ(kNumFrames,
+                  AAudioStream_read(aaudioStream, buffer, kNumFrames, kNanosPerSecond));
+    } else {
+        EXPECT_EQ(kNumFrames,
+                  AAudioStream_write(aaudioStream, buffer, kNumFrames, kNanosPerSecond));
+    }
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_requestStop(aaudioStream));
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream));
+    delete[] buffer;
+}
+
+static const aaudio_usage_t sUsages[] = {
+    DONT_SET,
+    AAUDIO_UNSPECIFIED,
+    AAUDIO_USAGE_MEDIA,
+    AAUDIO_USAGE_VOICE_COMMUNICATION,
+    AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
+    AAUDIO_USAGE_ALARM,
+    AAUDIO_USAGE_NOTIFICATION,
+    AAUDIO_USAGE_NOTIFICATION_RINGTONE,
+    AAUDIO_USAGE_NOTIFICATION_EVENT,
+    AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY,
+    AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE,
+    AAUDIO_USAGE_ASSISTANCE_SONIFICATION,
+    AAUDIO_USAGE_GAME,
+    AAUDIO_USAGE_ASSISTANT
+};
+
+static const aaudio_content_type_t sContentypes[] = {
+    DONT_SET,
+    AAUDIO_UNSPECIFIED,
+    AAUDIO_CONTENT_TYPE_SPEECH,
+    AAUDIO_CONTENT_TYPE_MUSIC,
+    AAUDIO_CONTENT_TYPE_MOVIE,
+    AAUDIO_CONTENT_TYPE_SONIFICATION
+};
+
+static const aaudio_input_preset_t sInputPresets[] = {
+    DONT_SET,
+    AAUDIO_UNSPECIFIED,
+    AAUDIO_INPUT_PRESET_GENERIC,
+    AAUDIO_INPUT_PRESET_CAMCORDER,
+    AAUDIO_INPUT_PRESET_VOICE_RECOGNITION,
+    AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION,
+    AAUDIO_INPUT_PRESET_UNPROCESSED,
+};
+
+static void checkAttributesUsage(aaudio_performance_mode_t perfMode) {
+    for (aaudio_usage_t usage : sUsages) {
+        checkAttributes(perfMode, usage, DONT_SET);
+    }
+}
+
+static void checkAttributesContentType(aaudio_input_preset_t perfMode) {
+    for (aaudio_content_type_t contentType : sContentypes) {
+        checkAttributes(perfMode, DONT_SET, contentType);
+    }
+}
+
+static void checkAttributesInputPreset(aaudio_performance_mode_t perfMode) {
+    for (aaudio_input_preset_t inputPreset : sInputPresets) {
+        checkAttributes(perfMode,
+                        DONT_SET,
+                        DONT_SET,
+                        inputPreset,
+                        AAUDIO_DIRECTION_INPUT);
+    }
+}
+
+TEST(test_attributes, aaudio_usage_perfnone) {
+    checkAttributesUsage(AAUDIO_PERFORMANCE_MODE_NONE);
+}
+
+TEST(test_attributes, aaudio_content_type_perfnone) {
+    checkAttributesContentType(AAUDIO_PERFORMANCE_MODE_NONE);
+}
+
+TEST(test_attributes, aaudio_input_preset_perfnone) {
+    checkAttributesInputPreset(AAUDIO_PERFORMANCE_MODE_NONE);
+}
+
+TEST(test_attributes, aaudio_usage_lowlat) {
+    checkAttributesUsage(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+}
+
+TEST(test_attributes, aaudio_content_type_lowlat) {
+    checkAttributesContentType(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+}
+
+TEST(test_attributes, aaudio_input_preset_lowlat) {
+    checkAttributesInputPreset(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+}

diff --git a/media/libaaudio/tests/test_bad_disconnect.cpp b/media/libaaudio/tests/test_bad_disconnect.cpp
new file mode 100644
index 0000000..435990d
--- /dev/null
+++ b/media/libaaudio/tests/test_bad_disconnect.cpp

@@ -0,0 +1,168 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/**
+ * Handle a DISCONNECT by only opening and starting a new stream
+ * without stopping and closing the old one.
+ * This caused the new stream to use the old disconnected device.
+ */
+
+#include <stdio.h>
+#include <thread>
+#include <unistd.h>
+
+#include <aaudio/AAudio.h>
+
+#define DEFAULT_TIMEOUT_NANOS  ((int64_t)1000000000)
+
+static void s_myErrorCallbackProc(
+        AAudioStream *stream,
+        void *userData,
+        aaudio_result_t error);
+
+struct AudioEngine {
+    AAudioStreamBuilder *builder = nullptr;
+    AAudioStream *stream = nullptr;
+    std::thread *thread = nullptr;
+    int64_t framesRead = 0;
+};
+
+AudioEngine s_AudioEngine;
+
+// Callback function that fills the audio output buffer.
+static aaudio_data_callback_result_t s_myDataCallbackProc(
+        AAudioStream *stream,
+        void *userData,
+        void *audioData,
+        int32_t numFrames
+) {
+    (void) userData;
+    (void) audioData;
+    (void) numFrames;
+    s_AudioEngine.framesRead = AAudioStream_getFramesRead(stream);
+    return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+static aaudio_result_t s_StartAudio() {
+    int32_t framesPerBurst = 0;
+    int32_t deviceId = 0;
+
+    // Use an AAudioStreamBuilder to contain requested parameters.
+    aaudio_result_t result = AAudio_createStreamBuilder(&s_AudioEngine.builder);
+    if (result != AAUDIO_OK) {
+        printf("AAudio_createStreamBuilder returned %s",
+               AAudio_convertResultToText(result));
+        return result;
+    }
+
+    // Request stream properties.
+    AAudioStreamBuilder_setFormat(s_AudioEngine.builder, AAUDIO_FORMAT_PCM_FLOAT);
+    AAudioStreamBuilder_setPerformanceMode(s_AudioEngine.builder, AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+    AAudioStreamBuilder_setDataCallback(s_AudioEngine.builder, s_myDataCallbackProc, nullptr);
+    AAudioStreamBuilder_setErrorCallback(s_AudioEngine.builder, s_myErrorCallbackProc, nullptr);
+
+    // Create an AAudioStream using the Builder.
+    result = AAudioStreamBuilder_openStream(s_AudioEngine.builder, &s_AudioEngine.stream);
+    if (result != AAUDIO_OK) {
+        printf("AAudioStreamBuilder_openStream returned %s",
+               AAudio_convertResultToText(result));
+        return result;
+    }
+
+    result = AAudioStream_requestStart(s_AudioEngine.stream);
+    if (result != AAUDIO_OK) {
+        printf("AAudioStream_requestStart returned %s",
+               AAudio_convertResultToText(result));
+    }
+
+    // Check to see what kind of stream we actually got.
+    deviceId = AAudioStream_getDeviceId(s_AudioEngine.stream);
+    framesPerBurst = AAudioStream_getFramesPerBurst(s_AudioEngine.stream);
+
+    printf("-------- started: deviceId = %3d, framesPerBurst = %3d\n", deviceId, framesPerBurst);
+
+    return result;
+}
+
+static aaudio_result_t s_StopAudio() {
+    aaudio_result_t result = AAUDIO_OK;
+    if (s_AudioEngine.stream != nullptr) {
+        result = AAudioStream_requestStop(s_AudioEngine.stream);
+        if (result != AAUDIO_OK) {
+            printf("AAudioStream_requestStop returned %s\n",
+                   AAudio_convertResultToText(result));
+        }
+        result = AAudioStream_close(s_AudioEngine.stream);
+        if (result != AAUDIO_OK) {
+            printf("AAudioStream_close returned %s\n",
+                   AAudio_convertResultToText(result));
+        }
+        s_AudioEngine.stream = nullptr;
+        AAudioStreamBuilder_delete(s_AudioEngine.builder);
+        s_AudioEngine.builder = nullptr;
+    }
+    return result;
+}
+
+static void s_StartThreadProc() {
+    // A good app would call s_StopAudio here! This test simulates a bad app.
+    s_StartAudio();
+    s_AudioEngine.thread = nullptr;
+}
+
+static void s_myErrorCallbackProc(
+        AAudioStream *stream __unused,
+        void *userData __unused,
+        aaudio_result_t error) {
+    if (error == AAUDIO_ERROR_DISCONNECTED) {
+        // Handle stream restart on a separate thread
+        if (s_AudioEngine.thread == nullptr) {
+            s_AudioEngine.thread = new std::thread(s_StartThreadProc);
+        }
+    }
+}
+
+int main(int argc, char **argv) {
+    (void) argc;
+    (void) argv;
+
+    aaudio_result_t result = AAUDIO_OK;
+
+    // Make printf print immediately so that debug info is not stuck
+    // in a buffer if we hang or crash.
+    setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
+
+    printf("Test Bad Disconnect V1.0\n");
+    printf("\n=========== Please PLUG and UNPLUG headphones! ==============\n\n");
+    printf("You should see the deviceID change on each plug event.\n");
+    printf("Headphones will generally get a new deviceId each time.\n");
+    printf("Speakers will have the same deviceId each time.\n");
+    printf("The framesRead should reset on each plug event then increase over time.\n");
+    printf("\n");
+
+    result = s_StartAudio();
+
+    if (result == AAUDIO_OK) {
+        for (int i = 20; i > 0; i--) {
+            sleep(1);
+            printf("playing silence #%d, framesRead = %d\n", i, (int) s_AudioEngine.framesRead);
+        }
+    }
+
+    s_StopAudio();
+
+    printf("result = %d = %s\n", result, AAudio_convertResultToText(result));
+}

diff --git a/media/libaaudio/tests/test_interference.cpp b/media/libaaudio/tests/test_interference.cpp
new file mode 100644
index 0000000..7eaf225
--- /dev/null
+++ b/media/libaaudio/tests/test_interference.cpp

@@ -0,0 +1,97 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Play a shared stream that might use MMAP.
+// Then play a second stream at a different sample rate.
+// Make sure the first stream is still running.
+// See: b/73369112 | AAudio disconnects shared stream if second MMAP open fails
+
+#include <memory.h>
+#include <stdio.h>
+#include <unistd.h>
+
+#include <android-base/macros.h>
+#include <aaudio/AAudio.h>
+
+#include <gtest/gtest.h>
+
+// Callback function that fills the audio output buffer.
+aaudio_data_callback_result_t MyDataCallbackProc(
+        AAudioStream *stream,
+        void *userData,
+        void *audioData,
+        int32_t numFrames) {
+    (void) userData;
+    int32_t numSamples = AAudioStream_getChannelCount(stream) * numFrames;
+    aaudio_format_t format = AAudioStream_getFormat(stream);
+    if (format == AAUDIO_FORMAT_PCM_I16) {
+        memset(audioData, 0, numSamples * sizeof(int16_t));
+    } else if (format == AAUDIO_FORMAT_PCM_FLOAT) {
+        memset(audioData, 0, numSamples * sizeof(float));
+    }
+    return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+//void foo() { // for tricking the Android Studio formatter
+TEST(test_interference, aaudio_mmap_interference) {
+
+    AAudioStreamBuilder *aaudioBuilder = nullptr;
+    AAudioStream *aaudioStream1 = nullptr;
+    AAudioStream *aaudioStream2 = nullptr;
+
+    // Use an AAudioStreamBuilder to contain requested parameters.
+    ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+
+    // Request stream properties.
+    AAudioStreamBuilder_setSampleRate(aaudioBuilder, 48000);
+    AAudioStreamBuilder_setDataCallback(aaudioBuilder, MyDataCallbackProc, nullptr);
+    AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+
+    // Create an AAudioStream using the Builder.
+    ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream1));
+    // Start it running.
+    ASSERT_EQ(AAUDIO_OK, AAudioStream_requestStart(aaudioStream1));
+
+    // Verify that the stream is running.
+    sleep(1);
+    EXPECT_LT(0, AAudioStream_getFramesRead(aaudioStream1));
+    ASSERT_EQ(AAUDIO_STREAM_STATE_STARTED, AAudioStream_getState(aaudioStream1));
+
+    // Now try to open a second stream with a different rate.
+    AAudioStreamBuilder_setSampleRate(aaudioBuilder, 44100);
+    ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream2));
+    ASSERT_EQ(AAUDIO_OK, AAudioStream_requestStart(aaudioStream2));
+
+    // Verify that the second stream is running.
+    sleep(1);
+    EXPECT_LT(0, AAudioStream_getFramesRead(aaudioStream2));
+
+    EXPECT_EQ(AAUDIO_STREAM_STATE_STARTED, AAudioStream_getState(aaudioStream2));
+
+    // Now verify that the first stream is still running.
+    EXPECT_EQ(AAUDIO_STREAM_STATE_STARTED, AAudioStream_getState(aaudioStream1));
+
+    int32_t framesRead1_1 = AAudioStream_getFramesRead(aaudioStream1);
+    EXPECT_LT(0, framesRead1_1);
+    sleep(1);
+    int32_t framesRead1_2 = AAudioStream_getFramesRead(aaudioStream1);
+    EXPECT_LT(0, framesRead1_2);
+    EXPECT_LT(framesRead1_1, framesRead1_2); // advancing?
+
+    AAudioStream_close(aaudioStream2);
+    AAudioStream_close(aaudioStream1);
+    AAudioStreamBuilder_delete(aaudioBuilder);
+}

diff --git a/media/libaaudio/tests/test_linear_ramp.cpp b/media/libaaudio/tests/test_linear_ramp.cpp
index 5c53982..93226ba 100644
--- a/media/libaaudio/tests/test_linear_ramp.cpp
+++ b/media/libaaudio/tests/test_linear_ramp.cpp

@@ -15,13 +15,13 @@
  */
 
 #include <iostream>
+#include <math.h>
 
 #include <gtest/gtest.h>
 
 #include "utility/AAudioUtilities.h"
 #include "utility/LinearRamp.h"
 
-
 TEST(test_linear_ramp, linear_ramp_segments) {
     LinearRamp ramp;
     const float source[4] = {1.0f, 1.0f, 1.0f, 1.0f };
@@ -32,40 +32,40 @@
     ramp.setLengthInFrames(8);
     ramp.setTarget(8.0f);
 
-    ASSERT_EQ(8, ramp.getLengthInFrames());
+    EXPECT_EQ(8, ramp.getLengthInFrames());
 
     bool ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
-    ASSERT_EQ(1, ramping);
-    ASSERT_EQ(0.0f, levelFrom);
-    ASSERT_EQ(4.0f, levelTo);
+    EXPECT_EQ(1, ramping);
+    EXPECT_EQ(0.0f, levelFrom);
+    EXPECT_EQ(4.0f, levelTo);
 
     AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
-    ASSERT_EQ(0.0f, destination[0]);
-    ASSERT_EQ(1.0f, destination[1]);
-    ASSERT_EQ(2.0f, destination[2]);
-    ASSERT_EQ(3.0f, destination[3]);
+    EXPECT_EQ(0.0f, destination[0]);
+    EXPECT_EQ(1.0f, destination[1]);
+    EXPECT_EQ(2.0f, destination[2]);
+    EXPECT_EQ(3.0f, destination[3]);
 
     ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
-    ASSERT_EQ(1, ramping);
-    ASSERT_EQ(4.0f, levelFrom);
-    ASSERT_EQ(8.0f, levelTo);
+    EXPECT_EQ(1, ramping);
+    EXPECT_EQ(4.0f, levelFrom);
+    EXPECT_EQ(8.0f, levelTo);
 
     AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
-    ASSERT_EQ(4.0f, destination[0]);
-    ASSERT_EQ(5.0f, destination[1]);
-    ASSERT_EQ(6.0f, destination[2]);
-    ASSERT_EQ(7.0f, destination[3]);
+    EXPECT_EQ(4.0f, destination[0]);
+    EXPECT_EQ(5.0f, destination[1]);
+    EXPECT_EQ(6.0f, destination[2]);
+    EXPECT_EQ(7.0f, destination[3]);
 
     ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
-    ASSERT_EQ(0, ramping);
-    ASSERT_EQ(8.0f, levelFrom);
-    ASSERT_EQ(8.0f, levelTo);
+    EXPECT_EQ(0, ramping);
+    EXPECT_EQ(8.0f, levelFrom);
+    EXPECT_EQ(8.0f, levelTo);
 
     AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
-    ASSERT_EQ(8.0f, destination[0]);
-    ASSERT_EQ(8.0f, destination[1]);
-    ASSERT_EQ(8.0f, destination[2]);
-    ASSERT_EQ(8.0f, destination[3]);
+    EXPECT_EQ(8.0f, destination[0]);
+    EXPECT_EQ(8.0f, destination[1]);
+    EXPECT_EQ(8.0f, destination[2]);
+    EXPECT_EQ(8.0f, destination[3]);
 
 };
 
@@ -80,29 +80,101 @@
     ramp.setLengthInFrames(4);
     ramp.setTarget(8.0f);
     ramp.forceCurrent(4.0f);
-    ASSERT_EQ(4.0f, ramp.getCurrent());
+    EXPECT_EQ(4.0f, ramp.getCurrent());
 
     bool ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
-    ASSERT_EQ(1, ramping);
-    ASSERT_EQ(4.0f, levelFrom);
-    ASSERT_EQ(8.0f, levelTo);
+    EXPECT_EQ(1, ramping);
+    EXPECT_EQ(4.0f, levelFrom);
+    EXPECT_EQ(8.0f, levelTo);
 
     AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
-    ASSERT_EQ(4.0f, destination[0]);
-    ASSERT_EQ(5.0f, destination[1]);
-    ASSERT_EQ(6.0f, destination[2]);
-    ASSERT_EQ(7.0f, destination[3]);
+    EXPECT_EQ(4.0f, destination[0]);
+    EXPECT_EQ(5.0f, destination[1]);
+    EXPECT_EQ(6.0f, destination[2]);
+    EXPECT_EQ(7.0f, destination[3]);
 
     ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
-    ASSERT_EQ(0, ramping);
-    ASSERT_EQ(8.0f, levelFrom);
-    ASSERT_EQ(8.0f, levelTo);
+    EXPECT_EQ(0, ramping);
+    EXPECT_EQ(8.0f, levelFrom);
+    EXPECT_EQ(8.0f, levelTo);
 
     AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
-    ASSERT_EQ(8.0f, destination[0]);
-    ASSERT_EQ(8.0f, destination[1]);
-    ASSERT_EQ(8.0f, destination[2]);
-    ASSERT_EQ(8.0f, destination[3]);
+    EXPECT_EQ(8.0f, destination[0]);
+    EXPECT_EQ(8.0f, destination[1]);
+    EXPECT_EQ(8.0f, destination[2]);
+    EXPECT_EQ(8.0f, destination[3]);
 
 };
 
+constexpr int16_t kMaxI16 = INT16_MAX;
+constexpr int16_t kMinI16 = INT16_MIN;
+constexpr int16_t kHalfI16 = 16384;
+constexpr int16_t kTenthI16 = 3277;
+
+//void AAudioConvert_floatToPcm16(const float *source,
+//                                int16_t *destination,
+//                                int32_t numSamples,
+//                                float amplitude);
+TEST(test_linear_ramp, float_to_i16) {
+    const float source[] = {12345.6f, 1.0f, 0.5f, 0.1f, 0.0f, -0.1f, -0.5f, -1.0f, -12345.6f};
+    constexpr size_t count = sizeof(source) / sizeof(source[0]);
+    int16_t destination[count];
+    const int16_t expected[count] = {kMaxI16, kMaxI16, kHalfI16, kTenthI16, 0,
+                                     -kTenthI16, -kHalfI16, kMinI16, kMinI16};
+
+    AAudioConvert_floatToPcm16(source, destination, count, 1.0f);
+    for (size_t i = 0; i < count; i++) {
+        EXPECT_EQ(expected[i], destination[i]);
+    }
+
+}
+
+//void AAudioConvert_pcm16ToFloat(const int16_t *source,
+//                                float *destination,
+//                                int32_t numSamples,
+//                                float amplitude);
+TEST(test_linear_ramp, i16_to_float) {
+    const int16_t source[] = {kMaxI16, kHalfI16, kTenthI16, 0,
+                              -kTenthI16, -kHalfI16, kMinI16};
+    constexpr size_t count = sizeof(source) / sizeof(source[0]);
+    float destination[count];
+    const float expected[count] = {(32767.0f / 32768.0f), 0.5f, 0.1f, 0.0f, -0.1f, -0.5f, -1.0f};
+
+    AAudioConvert_pcm16ToFloat(source, destination, count, 1.0f);
+    for (size_t i = 0; i < count; i++) {
+        EXPECT_NEAR(expected[i], destination[i], 0.0001f);
+    }
+
+}
+
+//void AAudio_linearRamp(const int16_t *source,
+//                       int16_t *destination,
+//                       int32_t numFrames,
+//                       int32_t samplesPerFrame,
+//                       float amplitude1,
+//                       float amplitude2);
+TEST(test_linear_ramp, ramp_i16_to_i16) {
+    const int16_t source[] = {1, 1, 1, 1, 1, 1, 1, 1};
+    constexpr size_t count = sizeof(source) / sizeof(source[0]);
+    int16_t destination[count];
+    // Ramp will sweep from -1 to almost +1
+    const int16_t expected[count] = {
+            -1, // from -1.00
+            -1, // from -0.75
+            -1, // from -0.55, round away from zero
+            0,  // from -0.25, round up to zero
+            0,  // from  0.00
+            0,  // from  0.25, round down to zero
+            1,  // from  0.50, round away from zero
+            1   // from  0.75
+    };
+
+    // sweep across zero to test symmetry
+    constexpr float amplitude1 = -1.0;
+    constexpr float amplitude2 = 1.0;
+    AAudio_linearRamp(source, destination, count, 1, amplitude1, amplitude2);
+    for (size_t i = 0; i < count; i++) {
+        EXPECT_EQ(expected[i], destination[i]);
+    }
+
+}

diff --git a/media/libaaudio/tests/test_n_streams.cpp b/media/libaaudio/tests/test_n_streams.cpp
index 271d024..e2d4a82 100644
--- a/media/libaaudio/tests/test_n_streams.cpp
+++ b/media/libaaudio/tests/test_n_streams.cpp

@@ -71,7 +71,6 @@
 
     AAudioStreamBuilder_delete(aaudioBuilder);
 
-finish:
     return result;
 }
 

diff --git a/media/libaaudio/tests/test_open_params.cpp b/media/libaaudio/tests/test_open_params.cpp
index 01b8799..3451242 100644
--- a/media/libaaudio/tests/test_open_params.cpp
+++ b/media/libaaudio/tests/test_open_params.cpp

@@ -25,21 +25,6 @@
 
 #include <gtest/gtest.h>
 
-static const char *getSharingModeText(aaudio_sharing_mode_t mode) {
-    const char *modeText = "unknown";
-    switch (mode) {
-    case AAUDIO_SHARING_MODE_EXCLUSIVE:
-        modeText = "EXCLUSIVE";
-        break;
-    case AAUDIO_SHARING_MODE_SHARED:
-        modeText = "SHARED";
-        break;
-    default:
-        break;
-    }
-    return modeText;
-}
-
 // Callback function that fills the audio output buffer.
 aaudio_data_callback_result_t MyDataCallbackProc(
         AAudioStream *stream,
@@ -67,7 +52,6 @@
     int32_t actualChannelCount = 0;
     int32_t actualSampleRate = 0;
     aaudio_format_t actualDataFormat = AAUDIO_FORMAT_UNSPECIFIED;
-    aaudio_sharing_mode_t actualSharingMode = AAUDIO_SHARING_MODE_SHARED;
     aaudio_direction_t actualDirection;
 
     AAudioStreamBuilder *aaudioBuilder = nullptr;

diff --git a/media/libaaudio/tests/test_recovery.cpp b/media/libaaudio/tests/test_recovery.cpp
index 7268a30..6e89f83 100644
--- a/media/libaaudio/tests/test_recovery.cpp
+++ b/media/libaaudio/tests/test_recovery.cpp

@@ -23,24 +23,9 @@
 
 #define DEFAULT_TIMEOUT_NANOS  ((int64_t)1000000000)
 
-static const char *getSharingModeText(aaudio_sharing_mode_t mode) {
-    const char *modeText = "unknown";
-    switch (mode) {
-        case AAUDIO_SHARING_MODE_EXCLUSIVE:
-            modeText = "EXCLUSIVE";
-            break;
-        case AAUDIO_SHARING_MODE_SHARED:
-            modeText = "SHARED";
-            break;
-        default:
-            break;
-    }
-    return modeText;
-}
-
 int main(int argc, char **argv) {
     (void) argc;
-    (void *)argv;
+    (void) argv;
 
     aaudio_result_t result = AAUDIO_OK;
 
@@ -52,7 +37,6 @@
     int32_t actualChannelCount = 0;
     int32_t actualSampleRate = 0;
     aaudio_format_t actualDataFormat = AAUDIO_FORMAT_PCM_FLOAT;
-    aaudio_sharing_mode_t actualSharingMode = AAUDIO_SHARING_MODE_SHARED;
 
     AAudioStreamBuilder *aaudioBuilder = nullptr;
     AAudioStream *aaudioStream = nullptr;

diff --git a/media/libaaudio/tests/test_session_id.cpp b/media/libaaudio/tests/test_session_id.cpp
new file mode 100644
index 0000000..3f7d4fc
--- /dev/null
+++ b/media/libaaudio/tests/test_session_id.cpp

@@ -0,0 +1,166 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Test AAudio SessionId, which is used to associate Effects with a stream
+
+#include <stdio.h>
+#include <unistd.h>
+
+#include <aaudio/AAudio.h>
+#include <gtest/gtest.h>
+
+constexpr int64_t kNanosPerSecond = 1000000000;
+constexpr int kNumFrames = 256;
+constexpr int kChannelCount = 2;
+
+// Test AAUDIO_SESSION_ID_NONE default
+static void checkSessionIdNone(aaudio_performance_mode_t perfMode) {
+
+    float *buffer = new float[kNumFrames * kChannelCount];
+
+    AAudioStreamBuilder *aaudioBuilder = nullptr;
+
+    AAudioStream *aaudioStream1 = nullptr;
+    int32_t       sessionId1 = 0;
+
+    // Use an AAudioStreamBuilder to contain requested parameters.
+    ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+
+    // Request stream properties.
+    AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
+
+    // Create an AAudioStream using the Builder.
+    ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream1));
+
+    // Since we did not request or specify a SessionID, we should get NONE
+    sessionId1 = AAudioStream_getSessionId(aaudioStream1);
+    ASSERT_EQ(AAUDIO_SESSION_ID_NONE, sessionId1);
+
+    ASSERT_EQ(AAUDIO_OK, AAudioStream_requestStart(aaudioStream1));
+
+    ASSERT_EQ(kNumFrames, AAudioStream_write(aaudioStream1, buffer, kNumFrames, kNanosPerSecond));
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_requestStop(aaudioStream1));
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream1));
+    delete[] buffer;
+    AAudioStreamBuilder_delete(aaudioBuilder);
+}
+
+TEST(test_session_id, aaudio_session_id_none_perfnone) {
+    checkSessionIdNone(AAUDIO_PERFORMANCE_MODE_NONE);
+}
+
+TEST(test_session_id, aaudio_session_id_none_lowlat) {
+    checkSessionIdNone(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+}
+
+// Test AAUDIO_SESSION_ID_ALLOCATE
+static void checkSessionIdAllocate(aaudio_performance_mode_t perfMode,
+                                   aaudio_direction_t direction) {
+
+    float *buffer = new float[kNumFrames * kChannelCount];
+
+    AAudioStreamBuilder *aaudioBuilder = nullptr;
+
+    AAudioStream *aaudioStream1 = nullptr;
+    int32_t       sessionId1 = 0;
+    AAudioStream *aaudioStream2 = nullptr;
+    int32_t       sessionId2 = 0;
+
+    // Use an AAudioStreamBuilder to contain requested parameters.
+    ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+
+    // Request stream properties.
+    AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
+    // This stream could be input or output.
+    AAudioStreamBuilder_setDirection(aaudioBuilder, direction);
+
+    // Ask AAudio to allocate a Session ID.
+    AAudioStreamBuilder_setSessionId(aaudioBuilder, AAUDIO_SESSION_ID_ALLOCATE);
+
+    // Create an AAudioStream using the Builder.
+    ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream1));
+
+    // Get the allocated ID from the stream.
+    sessionId1 = AAudioStream_getSessionId(aaudioStream1);
+
+    // Check for invalid session IDs.
+    ASSERT_NE(AAUDIO_SESSION_ID_NONE, sessionId1);
+    ASSERT_NE(AAUDIO_SESSION_ID_ALLOCATE, sessionId1);
+
+    ASSERT_EQ(AAUDIO_OK, AAudioStream_requestStart(aaudioStream1));
+
+    if (direction == AAUDIO_DIRECTION_INPUT) {
+        ASSERT_EQ(kNumFrames, AAudioStream_read(aaudioStream1,
+                                                buffer, kNumFrames, kNanosPerSecond));
+    } else {
+        ASSERT_EQ(kNumFrames, AAudioStream_write(aaudioStream1,
+                                         buffer, kNumFrames, kNanosPerSecond));
+    }
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_requestStop(aaudioStream1));
+
+    // Now open a second stream using the same session ID. ==================
+    AAudioStreamBuilder_setSessionId(aaudioBuilder, sessionId1);
+
+    // Reverse direction for second stream.
+    aaudio_direction_t otherDirection = (direction == AAUDIO_DIRECTION_OUTPUT)
+                                        ? AAUDIO_DIRECTION_INPUT
+                                        : AAUDIO_DIRECTION_OUTPUT;
+    AAudioStreamBuilder_setDirection(aaudioBuilder, otherDirection);
+
+    // Create an AAudioStream using the Builder.
+    ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream2));
+
+    // Get the allocated ID from the stream.
+    // It should match the ID that we set it to in the builder.
+    sessionId2 = AAudioStream_getSessionId(aaudioStream2);
+    ASSERT_EQ(sessionId1, sessionId2);
+
+    ASSERT_EQ(AAUDIO_OK, AAudioStream_requestStart(aaudioStream2));
+
+    if (otherDirection == AAUDIO_DIRECTION_INPUT) {
+        ASSERT_EQ(kNumFrames, AAudioStream_read(aaudioStream2,
+                                                 buffer, kNumFrames, kNanosPerSecond));
+    } else {
+        ASSERT_EQ(kNumFrames, AAudioStream_write(aaudioStream2,
+                                                 buffer, kNumFrames, kNanosPerSecond));
+    }
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_requestStop(aaudioStream2));
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream2));
+
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream1));
+    delete[] buffer;
+    AAudioStreamBuilder_delete(aaudioBuilder);
+}
+
+TEST(test_session_id, aaudio_session_id_alloc_perfnone_in) {
+    checkSessionIdAllocate(AAUDIO_PERFORMANCE_MODE_NONE, AAUDIO_DIRECTION_INPUT);
+}
+TEST(test_session_id, aaudio_session_id_alloc_perfnone_out) {
+    checkSessionIdAllocate(AAUDIO_PERFORMANCE_MODE_NONE, AAUDIO_DIRECTION_OUTPUT);
+}
+
+TEST(test_session_id, aaudio_session_id_alloc_lowlat_in) {
+    checkSessionIdAllocate(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, AAUDIO_DIRECTION_INPUT);
+}
+TEST(test_session_id, aaudio_session_id_alloc_lowlat_out) {
+    checkSessionIdAllocate(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, AAUDIO_DIRECTION_OUTPUT);
+}

diff --git a/media/libaaudio/tests/test_timestamps.cpp b/media/libaaudio/tests/test_timestamps.cpp
index fb363e7..dfa7815 100644
--- a/media/libaaudio/tests/test_timestamps.cpp
+++ b/media/libaaudio/tests/test_timestamps.cpp

@@ -22,8 +22,7 @@
 
 #include <aaudio/AAudio.h>
 #include <aaudio/AAudioTesting.h>
-#include "utils/AAudioExampleUtils.h"
-#include "../examples/utils/AAudioExampleUtils.h"
+#include "AAudioExampleUtils.h"
 
 // Arbitrary period for glitches, once per second at 48000 Hz.
 #define FORCED_UNDERRUN_PERIOD_FRAMES    48000
@@ -111,8 +110,6 @@
     aaudio_result_t result = AAUDIO_OK;
 
     int32_t framesPerBurst = 0;
-    float *buffer = nullptr;
-
     int32_t actualChannelCount = 0;
     int32_t actualSampleRate = 0;
     int32_t originalBufferSize = 0;
@@ -287,7 +284,7 @@
 
 int main(int argc, char **argv) {
     (void) argc;
-    (void *) argv;
+    (void) argv;
 
     aaudio_result_t result = AAUDIO_OK;
 

diff --git a/media/libaaudio/tests/test_various.cpp b/media/libaaudio/tests/test_various.cpp
new file mode 100644
index 0000000..4b065c9
--- /dev/null
+++ b/media/libaaudio/tests/test_various.cpp

@@ -0,0 +1,628 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Test various AAudio features including AAudioStream_setBufferSizeInFrames().
+
+#include <condition_variable>
+#include <mutex>
+#include <stdio.h>
+
+#include <android-base/macros.h>
+#include <aaudio/AAudio.h>
+
+#include <gtest/gtest.h>
+#include <unistd.h>
+
+// Callback function that does nothing.
+aaudio_data_callback_result_t NoopDataCallbackProc(
+        AAudioStream *stream,
+        void *userData,
+        void *audioData,
+        int32_t numFrames
+) {
+    (void) stream;
+    (void) userData;
+    (void) audioData;
+    (void) numFrames;
+    return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+// Test AAudioStream_setBufferSizeInFrames()
+
+constexpr int64_t NANOS_PER_MILLISECOND = 1000 * 1000;
+
+enum FunctionToCall {
+    CALL_START, CALL_STOP, CALL_PAUSE, CALL_FLUSH
+};
+
+void checkStateTransition(aaudio_performance_mode_t perfMode,
+                          aaudio_stream_state_t originalState,
+                          FunctionToCall functionToCall,
+                          aaudio_result_t expectedResult,
+                          aaudio_stream_state_t expectedState) {
+    AAudioStreamBuilder *aaudioBuilder = nullptr;
+    AAudioStream *aaudioStream = nullptr;
+
+    // Use an AAudioStreamBuilder to contain requested parameters.
+    ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+
+    // Request stream properties.
+    AAudioStreamBuilder_setDataCallback(aaudioBuilder, NoopDataCallbackProc, nullptr);
+    AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
+
+    // Create an AAudioStream using the Builder.
+    ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream));
+
+    // Verify Open State
+    aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(aaudioStream,
+                                                         AAUDIO_STREAM_STATE_UNKNOWN, &state,
+                                                         1000 * NANOS_PER_MILLISECOND));
+    EXPECT_EQ(AAUDIO_STREAM_STATE_OPEN, state);
+
+    // Put stream into desired state.
+    aaudio_stream_state_t inputState = AAUDIO_STREAM_STATE_UNINITIALIZED;
+    if (originalState != AAUDIO_STREAM_STATE_OPEN) {
+
+        ASSERT_EQ(AAUDIO_OK, AAudioStream_requestStart(aaudioStream));
+
+        if (originalState != AAUDIO_STREAM_STATE_STARTING) {
+
+            ASSERT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(aaudioStream,
+                                                                 AAUDIO_STREAM_STATE_STARTING,
+                                                                 &state,
+                                                                 1000 * NANOS_PER_MILLISECOND));
+            ASSERT_EQ(AAUDIO_STREAM_STATE_STARTED, state);
+
+            if (originalState == AAUDIO_STREAM_STATE_STOPPING) {
+                ASSERT_EQ(AAUDIO_OK, AAudioStream_requestStop(aaudioStream));
+            } else if (originalState == AAUDIO_STREAM_STATE_STOPPED) {
+                ASSERT_EQ(AAUDIO_OK, AAudioStream_requestStop(aaudioStream));
+                inputState = AAUDIO_STREAM_STATE_STOPPING;
+            } else if (originalState == AAUDIO_STREAM_STATE_PAUSING) {
+                ASSERT_EQ(AAUDIO_OK, AAudioStream_requestPause(aaudioStream));
+            } else if (originalState == AAUDIO_STREAM_STATE_PAUSED) {
+                ASSERT_EQ(AAUDIO_OK, AAudioStream_requestPause(aaudioStream));
+                inputState = AAUDIO_STREAM_STATE_PAUSING;
+            }
+        }
+    }
+
+    // Wait until past transitional state.
+    if (inputState != AAUDIO_STREAM_STATE_UNINITIALIZED) {
+        ASSERT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(aaudioStream,
+                                                             inputState,
+                                                             &state,
+                                                             1000 * NANOS_PER_MILLISECOND));
+        ASSERT_EQ(originalState, state);
+    }
+
+    aaudio_stream_state_t transitionalState = originalState;
+    switch(functionToCall) {
+        case FunctionToCall::CALL_START:
+            EXPECT_EQ(expectedResult, AAudioStream_requestStart(aaudioStream));
+            transitionalState = AAUDIO_STREAM_STATE_STARTING;
+            break;
+        case FunctionToCall::CALL_STOP:
+            EXPECT_EQ(expectedResult, AAudioStream_requestStop(aaudioStream));
+            transitionalState = AAUDIO_STREAM_STATE_STOPPING;
+            break;
+        case FunctionToCall::CALL_PAUSE:
+            EXPECT_EQ(expectedResult, AAudioStream_requestPause(aaudioStream));
+            transitionalState = AAUDIO_STREAM_STATE_PAUSING;
+            break;
+        case FunctionToCall::CALL_FLUSH:
+            EXPECT_EQ(expectedResult, AAudioStream_requestFlush(aaudioStream));
+            transitionalState = AAUDIO_STREAM_STATE_FLUSHING;
+            break;
+    }
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(aaudioStream,
+                                                         transitionalState,
+                                                         &state,
+                                                         1000 * NANOS_PER_MILLISECOND));
+    // We should not change state when a function fails.
+    if (expectedResult != AAUDIO_OK) {
+        ASSERT_EQ(originalState, expectedState);
+    }
+    EXPECT_EQ(expectedState, state);
+    if (state != expectedState) {
+        printf("ERROR - expected %s, actual = %s\n",
+                AAudio_convertStreamStateToText(expectedState),
+                AAudio_convertStreamStateToText(state));
+        fflush(stdout);
+    }
+
+    AAudioStream_close(aaudioStream);
+    AAudioStreamBuilder_delete(aaudioBuilder);
+}
+
+// TODO Use parameterized tests instead of these individual specific tests.
+
+// OPEN =================================================================
+TEST(test_various, aaudio_state_lowlat_open_start) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_OPEN,
+            FunctionToCall::CALL_START,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_STARTED);
+}
+
+TEST(test_various, aaudio_state_none_open_start) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_OPEN,
+            FunctionToCall::CALL_START,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_STARTED);
+}
+
+TEST(test_various, aaudio_state_lowlat_open_stop) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_OPEN,
+            FunctionToCall::CALL_STOP,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_STOPPED);
+}
+
+TEST(test_various, aaudio_state_none_open_stop) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_OPEN,
+            FunctionToCall::CALL_STOP,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_STOPPED);
+}
+
+TEST(test_various, aaudio_state_lowlat_open_pause) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_OPEN,
+            FunctionToCall::CALL_PAUSE,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_PAUSED);
+}
+
+TEST(test_various, aaudio_state_none_open_pause) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_OPEN,
+            FunctionToCall::CALL_PAUSE,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_PAUSED);
+}
+
+TEST(test_various, aaudio_state_lowlat_open_flush) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_OPEN,
+            FunctionToCall::CALL_FLUSH,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_FLUSHED);
+}
+
+TEST(test_various, aaudio_state_none_open_flush) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_OPEN,
+            FunctionToCall::CALL_FLUSH,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_FLUSHED);
+}
+
+
+// STARTED =================================================================
+TEST(test_various, aaudio_state_lowlat_started_start) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_STARTED,
+            FunctionToCall::CALL_START,
+            AAUDIO_ERROR_INVALID_STATE,
+            AAUDIO_STREAM_STATE_STARTED);
+}
+
+TEST(test_various, aaudio_state_none_started_start) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_STARTED,
+            FunctionToCall::CALL_START,
+            AAUDIO_ERROR_INVALID_STATE,
+            AAUDIO_STREAM_STATE_STARTED);
+}
+
+TEST(test_various, aaudio_state_lowlat_started_stop) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_STARTED,
+            FunctionToCall::CALL_STOP,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_STOPPED);
+}
+
+TEST(test_various, aaudio_state_none_started_stop) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_STARTED,
+            FunctionToCall::CALL_STOP,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_STOPPED);
+}
+
+TEST(test_various, aaudio_state_lowlat_started_pause) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_STARTED,
+            FunctionToCall::CALL_PAUSE,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_PAUSED);
+}
+
+TEST(test_various, aaudio_state_none_started_pause) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_STARTED,
+            FunctionToCall::CALL_PAUSE,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_PAUSED);
+}
+
+TEST(test_various, aaudio_state_lowlat_started_flush) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_STARTED,
+            FunctionToCall::CALL_FLUSH,
+            AAUDIO_ERROR_INVALID_STATE,
+            AAUDIO_STREAM_STATE_STARTED);
+}
+
+TEST(test_various, aaudio_state_none_started_flush) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_STARTED,
+            FunctionToCall::CALL_FLUSH,
+            AAUDIO_ERROR_INVALID_STATE,
+            AAUDIO_STREAM_STATE_STARTED);
+}
+
+// STOPPED =================================================================
+TEST(test_various, aaudio_state_lowlat_stopped_start) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_STOPPED,
+            FunctionToCall::CALL_START,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_STARTED);
+}
+
+TEST(test_various, aaudio_state_none_stopped_start) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_STOPPED,
+            FunctionToCall::CALL_START,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_STARTED);
+}
+
+TEST(test_various, aaudio_state_lowlat_stopped_stop) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_STOPPED,
+            FunctionToCall::CALL_STOP,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_STOPPED);
+}
+
+TEST(test_various, aaudio_state_none_stopped_stop) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_STOPPED,
+            FunctionToCall::CALL_STOP,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_STOPPED);
+}
+
+TEST(test_various, aaudio_state_lowlat_stopped_pause) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_STOPPED,
+            FunctionToCall::CALL_PAUSE,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_PAUSED);
+}
+
+TEST(test_various, aaudio_state_none_stopped_pause) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_STOPPED,
+            FunctionToCall::CALL_PAUSE,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_PAUSED);
+}
+
+TEST(test_various, aaudio_state_lowlat_stopped_flush) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+            AAUDIO_STREAM_STATE_STOPPED,
+            FunctionToCall::CALL_FLUSH,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_FLUSHED);
+}
+
+TEST(test_various, aaudio_state_none_stopped_flush) {
+    checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+            AAUDIO_STREAM_STATE_STOPPED,
+            FunctionToCall::CALL_FLUSH,
+            AAUDIO_OK,
+            AAUDIO_STREAM_STATE_FLUSHED);
+}
+
+// PAUSED =================================================================
+TEST(test_various, aaudio_state_lowlat_paused_start) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+        AAUDIO_STREAM_STATE_PAUSED,
+        FunctionToCall::CALL_START,
+        AAUDIO_OK,
+        AAUDIO_STREAM_STATE_STARTED);
+}
+
+TEST(test_various, aaudio_state_none_paused_start) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+        AAUDIO_STREAM_STATE_PAUSED,
+        FunctionToCall::CALL_START,
+        AAUDIO_OK,
+        AAUDIO_STREAM_STATE_STARTED);
+}
+
+TEST(test_various, aaudio_state_lowlat_paused_stop) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+        AAUDIO_STREAM_STATE_PAUSED,
+        FunctionToCall::CALL_STOP,
+        AAUDIO_OK,
+        AAUDIO_STREAM_STATE_STOPPED);
+}
+
+TEST(test_various, aaudio_state_none_paused_stop) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+        AAUDIO_STREAM_STATE_PAUSED,
+        FunctionToCall::CALL_STOP,
+        AAUDIO_OK,
+        AAUDIO_STREAM_STATE_STOPPED);
+}
+
+TEST(test_various, aaudio_state_lowlat_paused_pause) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+        AAUDIO_STREAM_STATE_PAUSED,
+        FunctionToCall::CALL_PAUSE,
+        AAUDIO_OK,
+        AAUDIO_STREAM_STATE_PAUSED);
+}
+
+TEST(test_various, aaudio_state_none_paused_pause) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+        AAUDIO_STREAM_STATE_PAUSED,
+        FunctionToCall::CALL_PAUSE,
+        AAUDIO_OK,
+        AAUDIO_STREAM_STATE_PAUSED);
+}
+
+TEST(test_various, aaudio_state_lowlat_paused_flush) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+        AAUDIO_STREAM_STATE_PAUSED,
+        FunctionToCall::CALL_FLUSH,
+        AAUDIO_OK,
+        AAUDIO_STREAM_STATE_FLUSHED);
+}
+
+TEST(test_various, aaudio_state_none_paused_flush) {
+checkStateTransition(AAUDIO_PERFORMANCE_MODE_NONE,
+        AAUDIO_STREAM_STATE_PAUSED,
+        FunctionToCall::CALL_FLUSH,
+        AAUDIO_OK,
+        AAUDIO_STREAM_STATE_FLUSHED);
+}
+
+// ==========================================================================
+TEST(test_various, aaudio_set_buffer_size) {
+
+    int32_t bufferCapacity;
+    int32_t framesPerBurst = 0;
+    int32_t actualSize = 0;
+
+    AAudioStreamBuilder *aaudioBuilder = nullptr;
+    AAudioStream *aaudioStream = nullptr;
+
+    // Use an AAudioStreamBuilder to contain requested parameters.
+    ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+
+    // Request stream properties.
+    AAudioStreamBuilder_setDataCallback(aaudioBuilder, NoopDataCallbackProc, nullptr);
+    AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+
+    // Create an AAudioStream using the Builder.
+    EXPECT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream));
+
+    // This is the number of frames that are read in one chunk by a DMA controller
+    // or a DSP or a mixer.
+    framesPerBurst = AAudioStream_getFramesPerBurst(aaudioStream);
+    bufferCapacity = AAudioStream_getBufferCapacityInFrames(aaudioStream);
+    printf("          bufferCapacity = %d, remainder = %d\n",
+           bufferCapacity, bufferCapacity % framesPerBurst);
+
+    actualSize = AAudioStream_setBufferSizeInFrames(aaudioStream, 0);
+    EXPECT_GT(actualSize, 0);
+    EXPECT_LE(actualSize, bufferCapacity);
+
+    actualSize = AAudioStream_setBufferSizeInFrames(aaudioStream, 2 * framesPerBurst);
+    EXPECT_GT(actualSize, framesPerBurst);
+    EXPECT_LE(actualSize, bufferCapacity);
+
+    actualSize = AAudioStream_setBufferSizeInFrames(aaudioStream, bufferCapacity - 1);
+    EXPECT_GT(actualSize, framesPerBurst);
+    EXPECT_LE(actualSize, bufferCapacity);
+
+    actualSize = AAudioStream_setBufferSizeInFrames(aaudioStream, bufferCapacity);
+    EXPECT_GT(actualSize, framesPerBurst);
+    EXPECT_LE(actualSize, bufferCapacity);
+
+    actualSize = AAudioStream_setBufferSizeInFrames(aaudioStream, bufferCapacity + 1);
+    EXPECT_GT(actualSize, framesPerBurst);
+    EXPECT_LE(actualSize, bufferCapacity);
+
+    actualSize = AAudioStream_setBufferSizeInFrames(aaudioStream, 1234567);
+    EXPECT_GT(actualSize, framesPerBurst);
+    EXPECT_LE(actualSize, bufferCapacity);
+
+    actualSize = AAudioStream_setBufferSizeInFrames(aaudioStream, INT32_MAX);
+    EXPECT_GT(actualSize, framesPerBurst);
+    EXPECT_LE(actualSize, bufferCapacity);
+
+    actualSize = AAudioStream_setBufferSizeInFrames(aaudioStream, INT32_MIN);
+    EXPECT_GT(actualSize, 0);
+    EXPECT_LE(actualSize, bufferCapacity);
+
+    AAudioStream_close(aaudioStream);
+    AAudioStreamBuilder_delete(aaudioBuilder);
+}
+
+// ************************************************************
+// Test to make sure that AAUDIO_CALLBACK_RESULT_STOP works.
+
+// Callback function that counts calls.
+aaudio_data_callback_result_t CallbackOnceProc(
+        AAudioStream *stream,
+        void *userData,
+        void *audioData,
+        int32_t numFrames
+) {
+    (void) stream;
+    (void) audioData;
+    (void) numFrames;
+
+    std::atomic<int32_t> *callbackCountPtr = (std::atomic<int32_t> *)userData;
+    (*callbackCountPtr)++;
+
+    return AAUDIO_CALLBACK_RESULT_STOP;
+}
+
+void checkCallbackOnce(aaudio_performance_mode_t perfMode) {
+
+    std::atomic<int32_t>   callbackCount{0};
+
+    AAudioStreamBuilder *aaudioBuilder = nullptr;
+    AAudioStream *aaudioStream = nullptr;
+
+    // Use an AAudioStreamBuilder to contain requested parameters.
+    ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+
+    // Request stream properties.
+    AAudioStreamBuilder_setDataCallback(aaudioBuilder, CallbackOnceProc, &callbackCount);
+    AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
+
+    // Create an AAudioStream using the Builder.
+    ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream));
+    AAudioStreamBuilder_delete(aaudioBuilder);
+
+    ASSERT_EQ(AAUDIO_OK, AAudioStream_requestStart(aaudioStream));
+
+    sleep(1); // Give callback a chance to run many times.
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_requestStop(aaudioStream));
+
+    EXPECT_EQ(1, callbackCount.load()); // should stop after first call
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream));
+}
+
+TEST(test_various, aaudio_callback_once_none) {
+    checkCallbackOnce(AAUDIO_PERFORMANCE_MODE_NONE);
+}
+
+TEST(test_various, aaudio_callback_once_lowlat) {
+    checkCallbackOnce(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+}
+
+// ************************************************************
+struct WakeUpCallbackData {
+    void wakeOther() {
+        // signal waiting test to wake up
+        {
+            std::lock_guard <std::mutex> lock(mutex);
+            finished = true;
+        }
+        conditionVariable.notify_one();
+    }
+
+    void waitForFinished() {
+        std::unique_lock <std::mutex> aLock(mutex);
+        conditionVariable.wait(aLock, [=] { return finished; });
+    }
+
+    // For signalling foreground test when callback finished
+    std::mutex              mutex;
+    std::condition_variable conditionVariable;
+    bool                    finished = false;
+};
+
+// Test to make sure we cannot call recursively into the system from a callback.
+struct DangerousData : public WakeUpCallbackData {
+    aaudio_result_t resultStart = AAUDIO_OK;
+    aaudio_result_t resultStop = AAUDIO_OK;
+    aaudio_result_t resultPause = AAUDIO_OK;
+    aaudio_result_t resultFlush = AAUDIO_OK;
+    aaudio_result_t resultClose = AAUDIO_OK;
+};
+
+// Callback function that tries to call back into the stream.
+aaudio_data_callback_result_t DangerousDataCallbackProc(
+        AAudioStream *stream,
+        void *userData,
+        void *audioData,
+        int32_t numFrames) {
+    (void) audioData;
+    (void) numFrames;
+
+    DangerousData *data = (DangerousData *)userData;
+    data->resultStart = AAudioStream_requestStart(stream);
+    data->resultStop = AAudioStream_requestStop(stream);
+    data->resultPause = AAudioStream_requestPause(stream);
+    data->resultFlush = AAudioStream_requestFlush(stream);
+    data->resultClose = AAudioStream_close(stream);
+
+    data->wakeOther();
+
+    return AAUDIO_CALLBACK_RESULT_STOP;
+}
+
+//int main() { // To fix Android Studio formatting when editing.
+void checkDangerousCallback(aaudio_performance_mode_t perfMode) {
+    DangerousData        dangerousData;
+    AAudioStreamBuilder *aaudioBuilder = nullptr;
+    AAudioStream        *aaudioStream = nullptr;
+
+    // Use an AAudioStreamBuilder to contain requested parameters.
+    ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+
+    // Request stream properties.
+    AAudioStreamBuilder_setDataCallback(aaudioBuilder, DangerousDataCallbackProc, &dangerousData);
+    AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
+
+    // Create an AAudioStream using the Builder.
+    ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream));
+    AAudioStreamBuilder_delete(aaudioBuilder);
+
+    ASSERT_EQ(AAUDIO_OK, AAudioStream_requestStart(aaudioStream));
+
+    dangerousData.waitForFinished();
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_requestStop(aaudioStream));
+
+    EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, dangerousData.resultStart);
+    EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, dangerousData.resultStop);
+    EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, dangerousData.resultPause);
+    EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, dangerousData.resultFlush);
+    EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, dangerousData.resultClose);
+
+    EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream));
+}
+
+//int main() { // To fix Android Studio formatting when editing.
+
+TEST(test_various, aaudio_callback_blockers_none) {
+    checkDangerousCallback(AAUDIO_PERFORMANCE_MODE_NONE);
+}
+
+TEST(test_various, aaudio_callback_blockers_lowlat) {
+    checkDangerousCallback(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+}

diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 61c946c..2df37a8 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp

@@ -6,7 +6,22 @@
 
 cc_library_shared {
     name: "libaudioclient",
+
+    aidl: {
+        export_aidl_headers: true,
+        local_include_dirs: ["aidl"],
+        include_dirs: [
+            "frameworks/av/media/libaudioclient/aidl",
+        ],
+    },
+
     srcs: [
+        // AIDL files for audioclient interfaces
+        // The headers for these interfaces will be available to any modules that
+        // include libaudioclient, at the path "aidl/package/path/BnFoo.h"
+        "aidl/android/media/IAudioRecord.aidl",
+        ":libaudioclient_aidl",
+
         "AudioEffect.cpp",
         "AudioPolicy.cpp",
         "AudioRecord.cpp",
@@ -17,7 +32,6 @@
         "IAudioFlingerClient.cpp",
         "IAudioPolicyService.cpp",
         "IAudioPolicyServiceClient.cpp",
-        "IAudioRecord.cpp",
         "IAudioTrack.cpp",
         "IEffect.cpp",
         "IEffectClient.cpp",
@@ -33,10 +47,12 @@
         "libdl",
         "libaudioutils",
         "libaudiomanager",
+        "libmedia_helper",
+        "libmediametrics",
     ],
     export_shared_lib_headers: ["libbinder"],
 
-    local_include_dirs: ["include/media"],
+    local_include_dirs: ["include/media", "aidl"],
     header_libs: ["libaudioclient_headers"],
     export_header_lib_headers: ["libaudioclient_headers"],
 
@@ -56,3 +72,11 @@
         ],
     },
 }
+
+// AIDL interface between libaudioclient and framework.jar
+filegroup {
+    name: "libaudioclient_aidl",
+    srcs: [
+        "aidl/android/media/IPlayer.aidl",
+    ],
+}

diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index ba4acc6..f9df5b1 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp

@@ -26,6 +26,8 @@
 #include <utils/Log.h>
 #include <private/media/AudioTrackShared.h>
 #include <media/IAudioFlinger.h>
+#include <media/MediaAnalyticsItem.h>
+#include <media/TypeConverter.h>
 
 #define WAIT_PERIOD_MS          10
 
@@ -65,12 +67,90 @@
 
 // ---------------------------------------------------------------------------
 
+static std::string audioFormatTypeString(audio_format_t value) {
+    std::string formatType;
+    if (FormatConverter::toString(value, formatType)) {
+        return formatType;
+    }
+    char rawbuffer[16];  // room for "%d"
+    snprintf(rawbuffer, sizeof(rawbuffer), "%d", value);
+    return rawbuffer;
+}
+
+static std::string audioSourceString(audio_source_t value) {
+    std::string source;
+    if (SourceTypeConverter::toString(value, source)) {
+        return source;
+    }
+    char rawbuffer[16];  // room for "%d"
+    snprintf(rawbuffer, sizeof(rawbuffer), "%d", value);
+    return rawbuffer;
+}
+
+void AudioRecord::MediaMetrics::gather(const AudioRecord *record)
+{
+    // key for media statistics is defined in the header
+    // attrs for media statistics
+    // NB: these are matched with public Java API constants defined
+    // in frameworks/base/media/java/android/media/AudioRecord.java
+    // These must be kept synchronized with the constants there.
+    static constexpr char kAudioRecordEncoding[] = "android.media.audiorecord.encoding";
+    static constexpr char kAudioRecordSource[] = "android.media.audiorecord.source";
+    static constexpr char kAudioRecordLatency[] = "android.media.audiorecord.latency";
+    static constexpr char kAudioRecordSampleRate[] = "android.media.audiorecord.samplerate";
+    static constexpr char kAudioRecordChannelCount[] = "android.media.audiorecord.channels";
+    static constexpr char kAudioRecordCreated[] = "android.media.audiorecord.createdMs";
+    static constexpr char kAudioRecordDuration[] = "android.media.audiorecord.durationMs";
+    static constexpr char kAudioRecordCount[] = "android.media.audiorecord.n";
+    static constexpr char kAudioRecordError[] = "android.media.audiorecord.errcode";
+    static constexpr char kAudioRecordErrorFunction[] = "android.media.audiorecord.errfunc";
+
+    // constructor guarantees mAnalyticsItem is valid
+
+    mAnalyticsItem->setInt32(kAudioRecordLatency, record->mLatency);
+    mAnalyticsItem->setInt32(kAudioRecordSampleRate, record->mSampleRate);
+    mAnalyticsItem->setInt32(kAudioRecordChannelCount, record->mChannelCount);
+    mAnalyticsItem->setCString(kAudioRecordEncoding,
+                               audioFormatTypeString(record->mFormat).c_str());
+    mAnalyticsItem->setCString(kAudioRecordSource,
+                               audioSourceString(record->mAttributes.source).c_str());
+
+    // log total duration recording, including anything currently running [and count].
+    nsecs_t active = 0;
+    if (mStartedNs != 0) {
+        active = systemTime() - mStartedNs;
+    }
+    mAnalyticsItem->setInt64(kAudioRecordDuration, (mDurationNs + active) / (1000 * 1000));
+    mAnalyticsItem->setInt32(kAudioRecordCount, mCount);
+
+    // XXX I don't know that this adds a lot of value, long term
+    if (mCreatedNs != 0) {
+        mAnalyticsItem->setInt64(kAudioRecordCreated, mCreatedNs / (1000 * 1000));
+    }
+
+    if (mLastError != NO_ERROR) {
+        mAnalyticsItem->setInt32(kAudioRecordError, mLastError);
+        mAnalyticsItem->setCString(kAudioRecordErrorFunction, mLastErrorFunc.c_str());
+    }
+}
+
+// hand the user a snapshot of the metrics.
+status_t AudioRecord::getMetrics(MediaAnalyticsItem * &item)
+{
+    mMediaMetrics.gather(this);
+    MediaAnalyticsItem *tmp = mMediaMetrics.dup();
+    if (tmp == nullptr) {
+        return BAD_VALUE;
+    }
+    item = tmp;
+    return NO_ERROR;
+}
+
 AudioRecord::AudioRecord(const String16 &opPackageName)
     : mActive(false), mStatus(NO_INIT), mOpPackageName(opPackageName),
       mSessionId(AUDIO_SESSION_ALLOCATE),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT),
-      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE), mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
-      mPortId(AUDIO_PORT_HANDLE_NONE)
+      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE), mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE)
 {
 }
 
@@ -89,24 +169,25 @@
         audio_input_flags_t flags,
         uid_t uid,
         pid_t pid,
-        const audio_attributes_t* pAttributes)
+        const audio_attributes_t* pAttributes,
+        audio_port_handle_t selectedDeviceId)
     : mActive(false),
       mStatus(NO_INIT),
       mOpPackageName(opPackageName),
       mSessionId(AUDIO_SESSION_ALLOCATE),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
       mPreviousSchedulingGroup(SP_DEFAULT),
-      mProxy(NULL),
-      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
-      mPortId(AUDIO_PORT_HANDLE_NONE)
+      mProxy(NULL)
 {
-    mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,
+    (void)set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,
             notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags,
-            uid, pid, pAttributes);
+            uid, pid, pAttributes, selectedDeviceId);
 }
 
 AudioRecord::~AudioRecord()
 {
+    mMediaMetrics.gather(this);
+
     if (mStatus == NO_ERROR) {
         // Make sure that callback function exits in the case where
         // it is looping on buffer empty condition in obtainBuffer().
@@ -148,14 +229,22 @@
         audio_input_flags_t flags,
         uid_t uid,
         pid_t pid,
-        const audio_attributes_t* pAttributes)
+        const audio_attributes_t* pAttributes,
+        audio_port_handle_t selectedDeviceId)
 {
+    status_t status = NO_ERROR;
+    uint32_t channelCount;
+    pid_t callingPid;
+    pid_t myPid;
+
     ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
           "notificationFrames %u, sessionId %d, transferType %d, flags %#x, opPackageName %s "
           "uid %d, pid %d",
           inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
           sessionId, transferType, flags, String8(mOpPackageName).string(), uid, pid);
 
+    mSelectedDeviceId = selectedDeviceId;
+
     switch (transferType) {
     case TRANSFER_DEFAULT:
         if (cbf == NULL || threadCanCallJava) {
@@ -167,7 +256,8 @@
     case TRANSFER_CALLBACK:
         if (cbf == NULL) {
             ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL");
-            return BAD_VALUE;
+            status = BAD_VALUE;
+            goto exit;
         }
         break;
     case TRANSFER_OBTAIN:
@@ -175,14 +265,16 @@
         break;
     default:
         ALOGE("Invalid transfer type %d", transferType);
-        return BAD_VALUE;
+        status = BAD_VALUE;
+        goto exit;
     }
     mTransfer = transferType;
 
     // invariant that mAudioRecord != 0 is true only after set() returns successfully
     if (mAudioRecord != 0) {
         ALOGE("Track already in use");
-        return INVALID_OPERATION;
+        status = INVALID_OPERATION;
+        goto exit;
     }
 
     if (pAttributes == NULL) {
@@ -206,16 +298,18 @@
     // AudioFlinger capture only supports linear PCM
     if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
         ALOGE("Format %#x is not linear pcm", format);
-        return BAD_VALUE;
+        status = BAD_VALUE;
+        goto exit;
     }
     mFormat = format;
 
     if (!audio_is_input_channel(channelMask)) {
         ALOGE("Invalid channel mask %#x", channelMask);
-        return BAD_VALUE;
+        status = BAD_VALUE;
+        goto exit;
     }
     mChannelMask = channelMask;
-    uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
+    channelCount = audio_channel_count_from_in_mask(channelMask);
     mChannelCount = channelCount;
 
     if (audio_is_linear_pcm(format)) {
@@ -224,28 +318,24 @@
         mFrameSize = sizeof(uint8_t);
     }
 
-    // mFrameCount is initialized in openRecord_l
+    // mFrameCount is initialized in createRecord_l
     mReqFrameCount = frameCount;
 
     mNotificationFramesReq = notificationFrames;
-    // mNotificationFramesAct is initialized in openRecord_l
+    // mNotificationFramesAct is initialized in createRecord_l
 
-    if (sessionId == AUDIO_SESSION_ALLOCATE) {
-        mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
-    } else {
-        mSessionId = sessionId;
-    }
+    mSessionId = sessionId;
     ALOGV("set(): mSessionId %d", mSessionId);
 
-    int callingpid = IPCThreadState::self()->getCallingPid();
-    int mypid = getpid();
-    if (uid == AUDIO_UID_INVALID || (callingpid != mypid)) {
+    callingPid = IPCThreadState::self()->getCallingPid();
+    myPid = getpid();
+    if (uid == AUDIO_UID_INVALID || (callingPid != myPid)) {
         mClientUid = IPCThreadState::self()->getCallingUid();
     } else {
         mClientUid = uid;
     }
-    if (pid == -1 || (callingpid != mypid)) {
-        mClientPid = callingpid;
+    if (pid == -1 || (callingPid != myPid)) {
+        mClientPid = callingPid;
     } else {
         mClientPid = pid;
     }
@@ -260,7 +350,7 @@
     }
 
     // create the IAudioRecord
-    status_t status = openRecord_l(0 /*epoch*/, mOpPackageName);
+    status = createRecord_l(0 /*epoch*/, mOpPackageName);
 
     if (status != NO_ERROR) {
         if (mAudioRecordThread != 0) {
@@ -268,10 +358,9 @@
             mAudioRecordThread->requestExitAndWait();
             mAudioRecordThread.clear();
         }
-        return status;
+        goto exit;
     }
 
-    mStatus = NO_ERROR;
     mUserData = user;
     // TODO: add audio hardware input latency here
     mLatency = (1000LL * mFrameCount) / mSampleRate;
@@ -286,7 +375,12 @@
     mFramesRead = 0;
     mFramesReadServerOffset = 0;
 
-    return NO_ERROR;
+exit:
+    mStatus = status;
+    if (status != NO_ERROR) {
+        mMediaMetrics.markError(status, __FUNCTION__);
+    }
+    return status;
 }
 
 // -------------------------------------------------------------------------
@@ -323,7 +417,7 @@
 
     status_t status = NO_ERROR;
     if (!(flags & CBLK_INVALID)) {
-        status = mAudioRecord->start(event, triggerSession);
+        status = mAudioRecord->start(event, triggerSession).transactionError();
         if (status == DEAD_OBJECT) {
             flags |= CBLK_INVALID;
         }
@@ -344,8 +438,14 @@
             get_sched_policy(0, &mPreviousSchedulingGroup);
             androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
         }
+
+        // we've successfully started, log that time
+        mMediaMetrics.logStart(systemTime());
     }
 
+    if (status != NO_ERROR) {
+        mMediaMetrics.markError(status, __FUNCTION__);
+    }
     return status;
 }
 
@@ -370,6 +470,9 @@
         setpriority(PRIO_PROCESS, 0, mPreviousPriority);
         set_sched_policy(0, mPreviousSchedulingGroup);
     }
+
+    // we've successfully started, log that time
+    mMediaMetrics.logStop(systemTime());
 }
 
 bool AudioRecord::stopped() const
@@ -489,6 +592,7 @@
                 mAudioRecord->stop();
             }
             android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
+            mProxy->interrupt();
         }
     }
     return NO_ERROR;
@@ -521,6 +625,27 @@
     return mRoutedDeviceId;
 }
 
+status_t AudioRecord::dump(int fd, const Vector<String16>& args __unused) const
+{
+    String8 result;
+
+    result.append(" AudioRecord::dump\n");
+    result.appendFormat("  status(%d), active(%d), session Id(%d)\n",
+                        mStatus, mActive, mSessionId);
+    result.appendFormat("  flags(%#x), req. flags(%#x), audio source(%d)\n",
+                        mFlags, mOrigFlags, mAttributes.source);
+    result.appendFormat("  format(%#x), channel mask(%#x), channel count(%u), sample rate(%u)\n",
+                  mFormat, mChannelMask, mChannelCount, mSampleRate);
+    result.appendFormat("  frame count(%zu), req. frame count(%zu)\n",
+                  mFrameCount, mReqFrameCount);
+    result.appendFormat("  notif. frame count(%u), req. notif. frame count(%u)\n",
+             mNotificationFramesAct, mNotificationFramesReq);
+    result.appendFormat("  input(%d), latency(%u), selected device Id(%d), routed device Id(%d)\n",
+                        mInput, mLatency, mSelectedDeviceId, mRoutedDeviceId);
+    ::write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
 // -------------------------------------------------------------------------
 // TODO Move this macro to a common header file for enum to string conversion in audio framework.
 #define MEDIA_CASE_ENUM(name) case name: return #name
@@ -536,70 +661,29 @@
 }
 
 // must be called with mLock held
-status_t AudioRecord::openRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName)
+status_t AudioRecord::createRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName)
 {
     const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
+    IAudioFlinger::CreateRecordInput input;
+    IAudioFlinger::CreateRecordOutput output;
+    audio_session_t originalSessionId;
+    sp<media::IAudioRecord> record;
+    void *iMemPointer;
+    audio_track_cblk_t* cblk;
+    status_t status;
+
     if (audioFlinger == 0) {
         ALOGE("Could not get audioflinger");
-        return NO_INIT;
+        status = NO_INIT;
+        goto exit;
     }
 
-    audio_io_handle_t input;
-
     // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
     // After fast request is denied, we will request again if IAudioRecord is re-created.
 
-    status_t status;
-
-    // Not a conventional loop, but a retry loop for at most two iterations total.
-    // Try first maybe with FAST flag then try again without FAST flag if that fails.
-    // Exits loop normally via a return at the bottom, or with error via a break.
-    // The sp<> references will be dropped when re-entering scope.
-    // The lack of indentation is deliberate, to reduce code churn and ease merges.
-    for (;;) {
-    audio_config_base_t config  = {
-            .sample_rate = mSampleRate,
-            .channel_mask = mChannelMask,
-            .format = mFormat
-        };
-    mRoutedDeviceId = mSelectedDeviceId;
-    status = AudioSystem::getInputForAttr(&mAttributes, &input,
-                                        mSessionId,
-                                        // FIXME compare to AudioTrack
-                                        mClientPid,
-                                        mClientUid,
-                                        &config,
-                                        mFlags, &mRoutedDeviceId, &mPortId);
-
-    if (status != NO_ERROR || input == AUDIO_IO_HANDLE_NONE) {
-        ALOGE("Could not get audio input for session %d, record source %d, sample rate %u, "
-              "format %#x, channel mask %#x, flags %#x",
-              mSessionId, mAttributes.source, mSampleRate, mFormat, mChannelMask, mFlags);
-        return BAD_VALUE;
-    }
-
     // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
     // we must release it ourselves if anything goes wrong.
 
-#if 0
-    size_t afFrameCount;
-    status = AudioSystem::getFrameCount(input, &afFrameCount);
-    if (status != NO_ERROR) {
-        ALOGE("getFrameCount(input=%d) status %d", input, status);
-        break;
-    }
-#endif
-
-    uint32_t afSampleRate;
-    status = AudioSystem::getSamplingRate(input, &afSampleRate);
-    if (status != NO_ERROR) {
-        ALOGE("getSamplingRate(input=%d) status %d", input, status);
-        break;
-    }
-    if (mSampleRate == 0) {
-        mSampleRate = afSampleRate;
-    }
-
     // Client can only express a preference for FAST.  Server will perform additional tests.
     if (mFlags & AUDIO_INPUT_FLAG_FAST) {
         bool useCaseAllowed =
@@ -618,66 +702,41 @@
         if (!useCaseAllowed) {
             ALOGW("AUDIO_INPUT_FLAG_FAST denied, incompatible transfer = %s",
                   convertTransferToText(mTransfer));
-        }
-
-        // sample rates must also match
-        bool sampleRateAllowed = mSampleRate == afSampleRate;
-        if (!sampleRateAllowed) {
-            ALOGW("AUDIO_INPUT_FLAG_FAST denied, rates do not match %u Hz, require %u Hz",
-                  mSampleRate, afSampleRate);
-        }
-
-        bool fastAllowed = useCaseAllowed && sampleRateAllowed;
-        if (!fastAllowed) {
             mFlags = (audio_input_flags_t) (mFlags & ~(AUDIO_INPUT_FLAG_FAST |
                     AUDIO_INPUT_FLAG_RAW));
-            AudioSystem::releaseInput(input, mSessionId);
-            continue;   // retry
         }
     }
 
-    // The notification frame count is the period between callbacks, as suggested by the client
-    // but moderated by the server.  For record, the calculations are done entirely on server side.
-    size_t notificationFrames = mNotificationFramesReq;
-    size_t frameCount = mReqFrameCount;
-
-    audio_input_flags_t flags = mFlags;
-
-    pid_t tid = -1;
+    input.attr = mAttributes;
+    input.config.sample_rate = mSampleRate;
+    input.config.channel_mask = mChannelMask;
+    input.config.format = mFormat;
+    input.clientInfo.clientUid = mClientUid;
+    input.clientInfo.clientPid = mClientPid;
+    input.clientInfo.clientTid = -1;
     if (mFlags & AUDIO_INPUT_FLAG_FAST) {
         if (mAudioRecordThread != 0) {
-            tid = mAudioRecordThread->getTid();
+            input.clientInfo.clientTid = mAudioRecordThread->getTid();
         }
     }
+    input.opPackageName = opPackageName;
 
-    size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,
-                                // but we will still need the original value also
-    audio_session_t originalSessionId = mSessionId;
+    input.flags = mFlags;
+    // The notification frame count is the period between callbacks, as suggested by the client
+    // but moderated by the server.  For record, the calculations are done entirely on server side.
+    input.frameCount = mReqFrameCount;
+    input.notificationFrameCount = mNotificationFramesReq;
+    input.selectedDeviceId = mSelectedDeviceId;
+    input.sessionId = mSessionId;
+    originalSessionId = mSessionId;
 
-    sp<IMemory> iMem;           // for cblk
-    sp<IMemory> bufferMem;
-    sp<IAudioRecord> record = audioFlinger->openRecord(input,
-                                                       mSampleRate,
-                                                       mFormat,
-                                                       mChannelMask,
-                                                       opPackageName,
-                                                       &temp,
-                                                       &flags,
-                                                       mClientPid,
-                                                       tid,
-                                                       mClientUid,
-                                                       &mSessionId,
-                                                       &notificationFrames,
-                                                       iMem,
-                                                       bufferMem,
-                                                       &status,
-                                                       mPortId);
-    ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
-            "session ID changed from %d to %d", originalSessionId, mSessionId);
+    record = audioFlinger->createRecord(input,
+                                                              output,
+                                                              &status);
 
     if (status != NO_ERROR) {
         ALOGE("AudioFlinger could not create record track, status: %d", status);
-        break;
+        goto exit;
     }
     ALOG_ASSERT(record != 0);
 
@@ -685,41 +744,41 @@
     // so we are no longer responsible for releasing it.
 
     mAwaitBoost = false;
-    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
-        if (flags & AUDIO_INPUT_FLAG_FAST) {
-            ALOGI("AUDIO_INPUT_FLAG_FAST successful; frameCount %zu -> %zu", frameCount, temp);
-            mAwaitBoost = true;
-        } else {
-            ALOGW("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %zu -> %zu", frameCount, temp);
-            mFlags = (audio_input_flags_t) (mFlags & ~(AUDIO_INPUT_FLAG_FAST |
-                    AUDIO_INPUT_FLAG_RAW));
-            continue;   // retry
-        }
+    if (output.flags & AUDIO_INPUT_FLAG_FAST) {
+        ALOGI("AUDIO_INPUT_FLAG_FAST successful; frameCount %zu -> %zu",
+              mReqFrameCount, output.frameCount);
+        mAwaitBoost = true;
     }
-    mFlags = flags;
+    mFlags = output.flags;
+    mRoutedDeviceId = output.selectedDeviceId;
+    mSessionId = output.sessionId;
+    mSampleRate = output.sampleRate;
 
-    if (iMem == 0) {
+    if (output.cblk == 0) {
         ALOGE("Could not get control block");
-        return NO_INIT;
+        status = NO_INIT;
+        goto exit;
     }
-    void *iMemPointer = iMem->pointer();
+    iMemPointer = output.cblk ->pointer();
     if (iMemPointer == NULL) {
         ALOGE("Could not get control block pointer");
-        return NO_INIT;
+        status = NO_INIT;
+        goto exit;
     }
-    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
+    cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
 
     // Starting address of buffers in shared memory.
     // The buffers are either immediately after the control block,
     // or in a separate area at discretion of server.
     void *buffers;
-    if (bufferMem == 0) {
+    if (output.buffers == 0) {
         buffers = cblk + 1;
     } else {
-        buffers = bufferMem->pointer();
+        buffers = output.buffers->pointer();
         if (buffers == NULL) {
             ALOGE("Could not get buffer pointer");
-            return NO_INIT;
+            status = NO_INIT;
+            goto exit;
         }
     }
 
@@ -729,43 +788,42 @@
         mDeathNotifier.clear();
     }
     mAudioRecord = record;
-    mCblkMemory = iMem;
-    mBufferMemory = bufferMem;
+    mCblkMemory = output.cblk;
+    mBufferMemory = output.buffers;
     IPCThreadState::self()->flushCommands();
 
     mCblk = cblk;
-    // note that temp is the (possibly revised) value of frameCount
-    if (temp < frameCount || (frameCount == 0 && temp == 0)) {
-        ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
+    // note that output.frameCount is the (possibly revised) value of mReqFrameCount
+    if (output.frameCount < mReqFrameCount || (mReqFrameCount == 0 && output.frameCount == 0)) {
+        ALOGW("Requested frameCount %zu but received frameCount %zu",
+              mReqFrameCount,  output.frameCount);
     }
-    frameCount = temp;
 
     // Make sure that application is notified with sufficient margin before overrun.
     // The computation is done on server side.
-    if (mNotificationFramesReq > 0 && notificationFrames != mNotificationFramesReq) {
+    if (mNotificationFramesReq > 0 && output.notificationFrameCount != mNotificationFramesReq) {
         ALOGW("Server adjusted notificationFrames from %u to %zu for frameCount %zu",
-                mNotificationFramesReq, notificationFrames, frameCount);
+                mNotificationFramesReq, output.notificationFrameCount, output.frameCount);
     }
-    mNotificationFramesAct = (uint32_t) notificationFrames;
-
+    mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
 
     //mInput != input includes the case where mInput == AUDIO_IO_HANDLE_NONE for first creation
-    if (mDeviceCallback != 0 && mInput != input) {
+    if (mDeviceCallback != 0 && mInput != output.inputId) {
         if (mInput != AUDIO_IO_HANDLE_NONE) {
             AudioSystem::removeAudioDeviceCallback(this, mInput);
         }
-        AudioSystem::addAudioDeviceCallback(this, input);
+        AudioSystem::addAudioDeviceCallback(this, output.inputId);
     }
 
     // We retain a copy of the I/O handle, but don't own the reference
-    mInput = input;
+    mInput = output.inputId;
     mRefreshRemaining = true;
 
-    mFrameCount = frameCount;
+    mFrameCount = output.frameCount;
     // If IAudioRecord is re-created, don't let the requested frameCount
     // decrease.  This can confuse clients that cache frameCount().
-    if (frameCount > mReqFrameCount) {
-        mReqFrameCount = frameCount;
+    if (mFrameCount > mReqFrameCount) {
+        mReqFrameCount = mFrameCount;
     }
 
     // update proxy
@@ -776,17 +834,9 @@
     mDeathNotifier = new DeathNotifier(this);
     IInterface::asBinder(mAudioRecord)->linkToDeath(mDeathNotifier, this);
 
-    return NO_ERROR;
-
-    // End of retry loop.
-    // The lack of indentation is deliberate, to reduce code churn and ease merges.
-    }
-
-// Arrive here on error, via a break
-    AudioSystem::releaseInput(input, mSessionId);
-    if (status == NO_ERROR) {
-        status = NO_INIT;
-    }
+exit:
+    mStatus = status;
+    // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
     return status;
 }
 
@@ -1216,22 +1266,43 @@
     ALOGW("dead IAudioRecord, creating a new one from %s()", from);
     ++mSequence;
 
+    const int INITIAL_RETRIES = 3;
+    int retries = INITIAL_RETRIES;
+retry:
+    if (retries < INITIAL_RETRIES) {
+        // refresh the audio configuration cache in this process to make sure we get new
+        // input parameters and new IAudioRecord in createRecord_l()
+        AudioSystem::clearAudioConfigCache();
+    }
     mFlags = mOrigFlags;
 
-    // if the new IAudioRecord is created, openRecord_l() will modify the
+    // if the new IAudioRecord is created, createRecord_l() will modify the
     // following member variables: mAudioRecord, mCblkMemory, mCblk, mBufferMemory.
     // It will also delete the strong references on previous IAudioRecord and IMemory
     Modulo<uint32_t> position(mProxy->getPosition());
     mNewPosition = position + mUpdatePeriod;
-    status_t result = openRecord_l(position, mOpPackageName);
-    if (result == NO_ERROR) {
+    status_t result = createRecord_l(position, mOpPackageName);
+
+    if (result != NO_ERROR) {
+        ALOGW("%s(): createRecord_l failed, do not retry", __func__);
+        retries = 0;
+    } else {
         if (mActive) {
             // callback thread or sync event hasn't changed
             // FIXME this fails if we have a new AudioFlinger instance
-            result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, AUDIO_SESSION_NONE);
+            result = mAudioRecord->start(
+                AudioSystem::SYNC_EVENT_SAME, AUDIO_SESSION_NONE).transactionError();
         }
         mFramesReadServerOffset = mFramesRead; // server resets to zero so we need an offset.
     }
+
+    if (result != NO_ERROR) {
+        ALOGW("%s() failed status %d, retries %d", __func__, result, retries);
+        if (--retries > 0) {
+            goto retry;
+        }
+    }
+
     if (result != NO_ERROR) {
         ALOGW("restoreRecord_l() failed status %d", result);
         mActive = false;
@@ -1303,6 +1374,14 @@
     }
 }
 
+// -------------------------------------------------------------------------
+
+status_t AudioRecord::getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones)
+{
+    AutoMutex lock(mLock);
+    return mAudioRecord->getActiveMicrophones(activeMicrophones).transactionError();
+}
+
 // =========================================================================
 
 void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused)

diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index cdc75ac..c072901 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp

@@ -20,6 +20,8 @@
 #include <utils/Log.h>
 #include <binder/IServiceManager.h>
 #include <binder/ProcessState.h>
+#include <binder/IPCThreadState.h>
+#include <media/AudioResamplerPublic.h>
 #include <media/AudioSystem.h>
 #include <media/IAudioFlinger.h>
 #include <media/IAudioPolicyService.h>
@@ -38,8 +40,7 @@
 sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
 audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
 dynamic_policy_callback AudioSystem::gDynPolicyCallback = NULL;
-record_config_callback  AudioSystem::gRecordConfigCallback = NULL;
-
+record_config_callback AudioSystem::gRecordConfigCallback = NULL;
 
 // establish binder interface to AudioFlinger service
 const sp<IAudioFlinger> AudioSystem::get_audio_flinger()
@@ -75,7 +76,9 @@
         af = gAudioFlinger;
     }
     if (afc != 0) {
+        int64_t token = IPCThreadState::self()->clearCallingIdentity();
         af->registerClient(afc);
+        IPCThreadState::self()->restoreCallingIdentity(token);
     }
     return af;
 }
@@ -253,6 +256,31 @@
     return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
 }
 
+/* static */ size_t AudioSystem::calculateMinFrameCount(
+        uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate,
+        uint32_t sampleRate, float speed /*, uint32_t notificationsPerBufferReq*/)
+{
+    // Ensure that buffer depth covers at least audio hardware latency
+    uint32_t minBufCount = afLatencyMs / ((1000 * afFrameCount) / afSampleRate);
+    if (minBufCount < 2) {
+        minBufCount = 2;
+    }
+#if 0
+    // The notificationsPerBufferReq parameter is not yet used for non-fast tracks,
+    // but keeping the code here to make it easier to add later.
+    if (minBufCount < notificationsPerBufferReq) {
+        minBufCount = notificationsPerBufferReq;
+    }
+#endif
+    ALOGV("calculateMinFrameCount afLatency %u  afFrameCount %u  afSampleRate %u  "
+            "sampleRate %u  speed %f  minBufCount: %u" /*"  notificationsPerBufferReq %u"*/,
+            afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount
+            /*, notificationsPerBufferReq*/);
+    return minBufCount * sourceFramesNeededWithTimestretch(
+            sampleRate, afFrameCount, afSampleRate, speed);
+}
+
+
 status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream_type_t streamType)
 {
     audio_io_handle_t output;
@@ -605,7 +633,7 @@
         || (channelMask != mInChannelMask)) {
         size_t inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
         if (inBuffSize == 0) {
-            ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %#x channelMask %x",
+            ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %#x channelMask %#x",
                     sampleRate, format, channelMask);
             return BAD_VALUE;
         }
@@ -742,7 +770,10 @@
         ap = gAudioPolicyService;
     }
     if (apc != 0) {
+        int64_t token = IPCThreadState::self()->clearCallingIdentity();
         ap->registerClient(apc);
+        ap->setAudioPortCallbacksEnabled(apc->isAudioPortCbEnabled());
+        IPCThreadState::self()->restoreCallingIdentity(token);
     }
 
     return ap;
@@ -822,22 +853,18 @@
 }
 
 
-audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream,
-                                    uint32_t samplingRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    audio_output_flags_t flags,
-                                    const audio_offload_info_t *offloadInfo)
+audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return 0;
-    return aps->getOutput(stream, samplingRate, format, channelMask, flags, offloadInfo);
+    return aps->getOutput(stream);
 }
 
 status_t AudioSystem::getOutputForAttr(const audio_attributes_t *attr,
                                         audio_io_handle_t *output,
                                         audio_session_t session,
                                         audio_stream_type_t *stream,
+                                        pid_t pid,
                                         uid_t uid,
                                         const audio_config_t *config,
                                         audio_output_flags_t flags,
@@ -846,7 +873,7 @@
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return NO_INIT;
-    return aps->getOutputForAttr(attr, output, session, stream, uid,
+    return aps->getOutputForAttr(attr, output, session, stream, pid, uid,
                                  config,
                                  flags, selectedDeviceId, portId);
 }
@@ -883,6 +910,7 @@
                                 audio_session_t session,
                                 pid_t pid,
                                 uid_t uid,
+                                const String16& opPackageName,
                                 const audio_config_base_t *config,
                                 audio_input_flags_t flags,
                                 audio_port_handle_t *selectedDeviceId,
@@ -891,32 +919,29 @@
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return NO_INIT;
     return aps->getInputForAttr(
-            attr, input, session, pid, uid,
+            attr, input, session, pid, uid, opPackageName,
             config, flags, selectedDeviceId, portId);
 }
 
-status_t AudioSystem::startInput(audio_io_handle_t input,
-                                 audio_session_t session)
+status_t AudioSystem::startInput(audio_port_handle_t portId, bool *silenced)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return PERMISSION_DENIED;
-    return aps->startInput(input, session);
+    return aps->startInput(portId, silenced);
 }
 
-status_t AudioSystem::stopInput(audio_io_handle_t input,
-                                audio_session_t session)
+status_t AudioSystem::stopInput(audio_port_handle_t portId)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return PERMISSION_DENIED;
-    return aps->stopInput(input, session);
+    return aps->stopInput(portId);
 }
 
-void AudioSystem::releaseInput(audio_io_handle_t input,
-                               audio_session_t session)
+void AudioSystem::releaseInput(audio_port_handle_t portId)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return;
-    aps->releaseInput(input, session);
+    aps->releaseInput(portId);
 }
 
 status_t AudioSystem::initStreamVolume(audio_stream_type_t stream,
@@ -1035,11 +1060,11 @@
     return af->getPrimaryOutputFrameCount();
 }
 
-status_t AudioSystem::setLowRamDevice(bool isLowRamDevice)
+status_t AudioSystem::setLowRamDevice(bool isLowRamDevice, int64_t totalMemory)
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
-    return af->setLowRamDevice(isLowRamDevice);
+    return af->setLowRamDevice(isLowRamDevice, totalMemory);
 }
 
 void AudioSystem::clearAudioConfigCache()
@@ -1254,6 +1279,31 @@
     return aps->getStreamVolumeDB(stream, index, device);
 }
 
+status_t AudioSystem::getMicrophones(std::vector<media::MicrophoneInfo> *microphones)
+{
+    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+    if (af == 0) return PERMISSION_DENIED;
+    return af->getMicrophones(microphones);
+}
+
+status_t AudioSystem::getSurroundFormats(unsigned int *numSurroundFormats,
+                                         audio_format_t *surroundFormats,
+                                         bool *surroundFormatsEnabled,
+                                         bool reported)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+    return aps->getSurroundFormats(
+            numSurroundFormats, surroundFormats, surroundFormatsEnabled, reported);
+}
+
+status_t AudioSystem::setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+    return aps->setSurroundFormatEnabled(audioFormat, enabled);
+}
+
 // ---------------------------------------------------------------------------
 
 int AudioSystem::AudioPolicyServiceClient::addAudioPortCallback(

diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 3529d2c..ab9efe8 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp

@@ -29,8 +29,11 @@
 #include <utils/Log.h>
 #include <private/media/AudioTrackShared.h>
 #include <media/IAudioFlinger.h>
+#include <media/AudioParameter.h>
 #include <media/AudioPolicyHelper.h>
 #include <media/AudioResamplerPublic.h>
+#include <media/MediaAnalyticsItem.h>
+#include <media/TypeConverter.h>
 
 #define WAIT_PERIOD_MS                  10
 #define WAIT_STREAM_END_TIMEOUT_SEC     120
@@ -39,6 +42,8 @@
 namespace android {
 // ---------------------------------------------------------------------------
 
+using media::VolumeShaper;
+
 // TODO: Move to a separate .h
 
 template <typename T>
@@ -51,8 +56,6 @@
     return x > y ? x : y;
 }
 
-static const int32_t NANOS_PER_SECOND = 1000000000;
-
 static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed)
 {
     return ((double)frames * 1000000000) / ((double)sampleRate * speed);
@@ -97,32 +100,6 @@
     return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch;
 }
 
-// Must match similar computation in createTrack_l in Threads.cpp.
-// TODO: Move to a common library
-static size_t calculateMinFrameCount(
-        uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate,
-        uint32_t sampleRate, float speed /*, uint32_t notificationsPerBufferReq*/)
-{
-    // Ensure that buffer depth covers at least audio hardware latency
-    uint32_t minBufCount = afLatencyMs / ((1000 * afFrameCount) / afSampleRate);
-    if (minBufCount < 2) {
-        minBufCount = 2;
-    }
-#if 0
-    // The notificationsPerBufferReq parameter is not yet used for non-fast tracks,
-    // but keeping the code here to make it easier to add later.
-    if (minBufCount < notificationsPerBufferReq) {
-        minBufCount = notificationsPerBufferReq;
-    }
-#endif
-    ALOGV("calculateMinFrameCount afLatency %u  afFrameCount %u  afSampleRate %u  "
-            "sampleRate %u  speed %f  minBufCount: %u" /*"  notificationsPerBufferReq %u"*/,
-            afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount
-            /*, notificationsPerBufferReq*/);
-    return minBufCount * sourceFramesNeededWithTimestretch(
-            sampleRate, afFrameCount, afSampleRate, speed);
-}
-
 // static
 status_t AudioTrack::getMinFrameCount(
         size_t* frameCount,
@@ -163,8 +140,8 @@
 
     // When called from createTrack, speed is 1.0f (normal speed).
     // This is rechecked again on setting playback rate (TODO: on setting sample rate, too).
-    *frameCount = calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, 1.0f
-            /*, 0 notificationsPerBufferReq*/);
+    *frameCount = AudioSystem::calculateMinFrameCount(afLatency, afFrameCount, afSampleRate,
+                                              sampleRate, 1.0f /*, 0 notificationsPerBufferReq*/);
 
     // The formula above should always produce a non-zero value under normal circumstances:
     // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX.
@@ -181,6 +158,88 @@
 
 // ---------------------------------------------------------------------------
 
+static std::string audioContentTypeString(audio_content_type_t value) {
+    std::string contentType;
+    if (AudioContentTypeConverter::toString(value, contentType)) {
+        return contentType;
+    }
+    char rawbuffer[16];  // room for "%d"
+    snprintf(rawbuffer, sizeof(rawbuffer), "%d", value);
+    return rawbuffer;
+}
+
+static std::string audioUsageString(audio_usage_t value) {
+    std::string usage;
+    if (UsageTypeConverter::toString(value, usage)) {
+        return usage;
+    }
+    char rawbuffer[16];  // room for "%d"
+    snprintf(rawbuffer, sizeof(rawbuffer), "%d", value);
+    return rawbuffer;
+}
+
+void AudioTrack::MediaMetrics::gather(const AudioTrack *track)
+{
+
+    // key for media statistics is defined in the header
+    // attrs for media statistics
+    // NB: these are matched with public Java API constants defined
+    // in frameworks/base/media/java/android/media/AudioTrack.java
+    // These must be kept synchronized with the constants there.
+    static constexpr char kAudioTrackStreamType[] = "android.media.audiotrack.streamtype";
+    static constexpr char kAudioTrackContentType[] = "android.media.audiotrack.type";
+    static constexpr char kAudioTrackUsage[] = "android.media.audiotrack.usage";
+    static constexpr char kAudioTrackSampleRate[] = "android.media.audiotrack.samplerate";
+    static constexpr char kAudioTrackChannelMask[] = "android.media.audiotrack.channelmask";
+
+    // NB: These are not yet exposed as public Java API constants.
+    static constexpr char kAudioTrackUnderrunFrames[] = "android.media.audiotrack.underrunframes";
+    static constexpr char kAudioTrackStartupGlitch[] = "android.media.audiotrack.glitch.startup";
+
+    // only if we're in a good state...
+    // XXX: shall we gather alternative info if failing?
+    const status_t lstatus = track->initCheck();
+    if (lstatus != NO_ERROR) {
+        ALOGD("no metrics gathered, track status=%d", (int) lstatus);
+        return;
+    }
+
+    // constructor guarantees mAnalyticsItem is valid
+
+    const int32_t underrunFrames = track->getUnderrunFrames();
+    if (underrunFrames != 0) {
+        mAnalyticsItem->setInt32(kAudioTrackUnderrunFrames, underrunFrames);
+    }
+
+    if (track->mTimestampStartupGlitchReported) {
+        mAnalyticsItem->setInt32(kAudioTrackStartupGlitch, 1);
+    }
+
+    if (track->mStreamType != -1) {
+        // deprecated, but this will tell us who still uses it.
+        mAnalyticsItem->setInt32(kAudioTrackStreamType, track->mStreamType);
+    }
+    // XXX: consider including from mAttributes: source type
+    mAnalyticsItem->setCString(kAudioTrackContentType,
+                               audioContentTypeString(track->mAttributes.content_type).c_str());
+    mAnalyticsItem->setCString(kAudioTrackUsage,
+                               audioUsageString(track->mAttributes.usage).c_str());
+    mAnalyticsItem->setInt32(kAudioTrackSampleRate, track->mSampleRate);
+    mAnalyticsItem->setInt64(kAudioTrackChannelMask, track->mChannelMask);
+}
+
+// hand the user a snapshot of the metrics.
+status_t AudioTrack::getMetrics(MediaAnalyticsItem * &item)
+{
+    mMediaMetrics.gather(this);
+    MediaAnalyticsItem *tmp = mMediaMetrics.dup();
+    if (tmp == nullptr) {
+        return BAD_VALUE;
+    }
+    item = tmp;
+    return NO_ERROR;
+}
+
 AudioTrack::AudioTrack()
     : mStatus(NO_INIT),
       mState(STATE_STOPPED),
@@ -188,8 +247,7 @@
       mPreviousSchedulingGroup(SP_DEFAULT),
       mPausedPosition(0),
       mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
-      mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
-      mPortId(AUDIO_PORT_HANDLE_NONE)
+      mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE)
 {
     mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
     mAttributes.usage = AUDIO_USAGE_UNKNOWN;
@@ -214,19 +272,18 @@
         pid_t pid,
         const audio_attributes_t* pAttributes,
         bool doNotReconnect,
-        float maxRequiredSpeed)
+        float maxRequiredSpeed,
+        audio_port_handle_t selectedDeviceId)
     : mStatus(NO_INIT),
       mState(STATE_STOPPED),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
       mPreviousSchedulingGroup(SP_DEFAULT),
-      mPausedPosition(0),
-      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
-      mPortId(AUDIO_PORT_HANDLE_NONE)
+      mPausedPosition(0)
 {
-    mStatus = set(streamType, sampleRate, format, channelMask,
+    (void)set(streamType, sampleRate, format, channelMask,
             frameCount, flags, cbf, user, notificationFrames,
             0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
-            offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
+            offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
 }
 
 AudioTrack::AudioTrack(
@@ -252,10 +309,9 @@
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
       mPreviousSchedulingGroup(SP_DEFAULT),
       mPausedPosition(0),
-      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
-      mPortId(AUDIO_PORT_HANDLE_NONE)
+      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
 {
-    mStatus = set(streamType, sampleRate, format, channelMask,
+    (void)set(streamType, sampleRate, format, channelMask,
             0 /*frameCount*/, flags, cbf, user, notificationFrames,
             sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
             uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
@@ -263,6 +319,9 @@
 
 AudioTrack::~AudioTrack()
 {
+    // pull together the numbers, before we clean up our structures
+    mMediaMetrics.gather(this);
+
     if (mStatus == NO_ERROR) {
         // Make sure that callback function exits in the case where
         // it is looping on buffer full condition in obtainBuffer().
@@ -308,14 +367,22 @@
         pid_t pid,
         const audio_attributes_t* pAttributes,
         bool doNotReconnect,
-        float maxRequiredSpeed)
+        float maxRequiredSpeed,
+        audio_port_handle_t selectedDeviceId)
 {
+    status_t status;
+    uint32_t channelCount;
+    pid_t callingPid;
+    pid_t myPid;
+
     ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
           "flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
           streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
           sessionId, transferType, uid, pid);
 
     mThreadCanCallJava = threadCanCallJava;
+    mSelectedDeviceId = selectedDeviceId;
+    mSessionId = sessionId;
 
     switch (transferType) {
     case TRANSFER_DEFAULT:
@@ -330,25 +397,29 @@
     case TRANSFER_CALLBACK:
         if (cbf == NULL || sharedBuffer != 0) {
             ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0");
-            return BAD_VALUE;
+            status = BAD_VALUE;
+            goto exit;
         }
         break;
     case TRANSFER_OBTAIN:
     case TRANSFER_SYNC:
         if (sharedBuffer != 0) {
             ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0");
-            return BAD_VALUE;
+            status = BAD_VALUE;
+            goto exit;
         }
         break;
     case TRANSFER_SHARED:
         if (sharedBuffer == 0) {
             ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0");
-            return BAD_VALUE;
+            status = BAD_VALUE;
+            goto exit;
         }
         break;
     default:
         ALOGE("Invalid transfer type %d", transferType);
-        return BAD_VALUE;
+        status = BAD_VALUE;
+        goto exit;
     }
     mSharedBuffer = sharedBuffer;
     mTransfer = transferType;
@@ -362,7 +433,8 @@
     // invariant that mAudioTrack != 0 is true only after set() returns successfully
     if (mAudioTrack != 0) {
         ALOGE("Track already in use");
-        return INVALID_OPERATION;
+        status = INVALID_OPERATION;
+        goto exit;
     }
 
     // handle default values first.
@@ -372,7 +444,8 @@
     if (pAttributes == NULL) {
         if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
             ALOGE("Invalid stream type %d", streamType);
-            return BAD_VALUE;
+            status = BAD_VALUE;
+            goto exit;
         }
         mStreamType = streamType;
 
@@ -404,16 +477,18 @@
     // validate parameters
     if (!audio_is_valid_format(format)) {
         ALOGE("Invalid format %#x", format);
-        return BAD_VALUE;
+        status = BAD_VALUE;
+        goto exit;
     }
     mFormat = format;
 
     if (!audio_is_output_channel(channelMask)) {
         ALOGE("Invalid channel mask %#x", channelMask);
-        return BAD_VALUE;
+        status = BAD_VALUE;
+        goto exit;
     }
     mChannelMask = channelMask;
-    uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
+    channelCount = audio_channel_count_from_out_mask(channelMask);
     mChannelCount = channelCount;
 
     // force direct flag if format is not linear PCM
@@ -448,7 +523,8 @@
 
     // sampling rate must be specified for direct outputs
     if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
-        return BAD_VALUE;
+        status = BAD_VALUE;
+        goto exit;
     }
     mSampleRate = sampleRate;
     mOriginalSampleRate = sampleRate;
@@ -479,12 +555,14 @@
         if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
             ALOGE("notificationFrames=%d not permitted for non-fast track",
                     notificationFrames);
-            return BAD_VALUE;
+            status = BAD_VALUE;
+            goto exit;
         }
         if (frameCount > 0) {
             ALOGE("notificationFrames=%d not permitted with non-zero frameCount=%zu",
                     notificationFrames, frameCount);
-            return BAD_VALUE;
+            status = BAD_VALUE;
+            goto exit;
         }
         mNotificationFramesReq = 0;
         const uint32_t minNotificationsPerBuffer = 1;
@@ -496,20 +574,15 @@
                 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
     }
     mNotificationFramesAct = 0;
-    if (sessionId == AUDIO_SESSION_ALLOCATE) {
-        mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
-    } else {
-        mSessionId = sessionId;
-    }
-    int callingpid = IPCThreadState::self()->getCallingPid();
-    int mypid = getpid();
-    if (uid == AUDIO_UID_INVALID || (callingpid != mypid)) {
+    callingPid = IPCThreadState::self()->getCallingPid();
+    myPid = getpid();
+    if (uid == AUDIO_UID_INVALID || (callingPid != myPid)) {
         mClientUid = IPCThreadState::self()->getCallingUid();
     } else {
         mClientUid = uid;
     }
-    if (pid == -1 || (callingpid != mypid)) {
-        mClientPid = callingpid;
+    if (pid == -1 || (callingPid != myPid)) {
+        mClientPid = callingPid;
     } else {
         mClientPid = pid;
     }
@@ -524,7 +597,7 @@
     }
 
     // create the IAudioTrack
-    status_t status = createTrack_l();
+    status = createTrack_l();
 
     if (status != NO_ERROR) {
         if (mAudioTrackThread != 0) {
@@ -532,10 +605,9 @@
             mAudioTrackThread->requestExitAndWait();
             mAudioTrackThread.clear();
         }
-        return status;
+        goto exit;
     }
 
-    mStatus = NO_ERROR;
     mUserData = user;
     mLoopCount = 0;
     mLoopStart = 0;
@@ -562,8 +634,11 @@
     mFramesWritten = 0;
     mFramesWrittenServerOffset = 0;
     mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
-    mVolumeHandler = new VolumeHandler();
-    return NO_ERROR;
+    mVolumeHandler = new media::VolumeHandler();
+
+exit:
+    mStatus = status;
+    return status;
 }
 
 // -------------------------------------------------------------------------
@@ -695,6 +770,7 @@
         mReleased = 0;
     }
 
+    mProxy->stop(); // notify server not to read beyond current client position until start().
     mProxy->interrupt();
     mAudioTrack->stop();
 
@@ -730,7 +806,7 @@
         return;
     }
     AutoMutex lock(mLock);
-    if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) {
+    if (mState == STATE_ACTIVE) {
         return;
     }
     flush_l();
@@ -1219,6 +1295,7 @@
         mSelectedDeviceId = deviceId;
         if (mStatus == NO_ERROR) {
             android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
+            mProxy->interrupt();
         }
     }
     return NO_ERROR;
@@ -1306,76 +1383,19 @@
 
 status_t AudioTrack::createTrack_l()
 {
+    status_t status;
+    bool callbackAdded = false;
+
     const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
     if (audioFlinger == 0) {
         ALOGE("Could not get audioflinger");
-        return NO_INIT;
+        status = NO_INIT;
+        goto exit;
     }
 
-    audio_io_handle_t output;
-    audio_stream_type_t streamType = mStreamType;
-    audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL;
-    bool callbackAdded = false;
-
+    {
     // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
     // After fast request is denied, we will request again if IAudioTrack is re-created.
-
-    status_t status;
-    audio_config_t config = AUDIO_CONFIG_INITIALIZER;
-    config.sample_rate = mSampleRate;
-    config.channel_mask = mChannelMask;
-    config.format = mFormat;
-    config.offload_info = mOffloadInfoCopy;
-    mRoutedDeviceId = mSelectedDeviceId;
-    status = AudioSystem::getOutputForAttr(attr, &output,
-                                           mSessionId, &streamType, mClientUid,
-                                           &config,
-                                           mFlags, &mRoutedDeviceId, &mPortId);
-
-    if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) {
-        ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u,"
-              " format %#x, channel mask %#x, flags %#x",
-              mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask,
-              mFlags);
-        return BAD_VALUE;
-    }
-    {
-    // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
-    // we must release it ourselves if anything goes wrong.
-
-    // Not all of these values are needed under all conditions, but it is easier to get them all
-    status = AudioSystem::getLatency(output, &mAfLatency);
-    if (status != NO_ERROR) {
-        ALOGE("getLatency(%d) failed status %d", output, status);
-        goto release;
-    }
-    ALOGV("createTrack_l() output %d afLatency %u", output, mAfLatency);
-
-    status = AudioSystem::getFrameCount(output, &mAfFrameCount);
-    if (status != NO_ERROR) {
-        ALOGE("getFrameCount(output=%d) status %d", output, status);
-        goto release;
-    }
-
-    // TODO consider making this a member variable if there are other uses for it later
-    size_t afFrameCountHAL;
-    status = AudioSystem::getFrameCountHAL(output, &afFrameCountHAL);
-    if (status != NO_ERROR) {
-        ALOGE("getFrameCountHAL(output=%d) status %d", output, status);
-        goto release;
-    }
-    ALOG_ASSERT(afFrameCountHAL > 0);
-
-    status = AudioSystem::getSamplingRate(output, &mAfSampleRate);
-    if (status != NO_ERROR) {
-        ALOGE("getSamplingRate(output=%d) status %d", output, status);
-        goto release;
-    }
-    if (mSampleRate == 0) {
-        mSampleRate = mAfSampleRate;
-        mOriginalSampleRate = mAfSampleRate;
-    }
-
     // Client can only express a preference for FAST.  Server will perform additional tests.
     if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
         // either of these use cases:
@@ -1389,130 +1409,81 @@
             // use case 4: synchronous write
             ((mTransfer == TRANSFER_SYNC) && mThreadCanCallJava);
 
-        bool useCaseAllowed = sharedBuffer || transferAllowed;
-        if (!useCaseAllowed) {
-            ALOGW("AUDIO_OUTPUT_FLAG_FAST denied, not shared buffer and transfer = %s",
-                  convertTransferToText(mTransfer));
-        }
-
-        // sample rates must also match
-        bool sampleRateAllowed = mSampleRate == mAfSampleRate;
-        if (!sampleRateAllowed) {
-            ALOGW("AUDIO_OUTPUT_FLAG_FAST denied, rates do not match %u Hz, require %u Hz",
-                  mSampleRate, mAfSampleRate);
-        }
-
-        bool fastAllowed = useCaseAllowed && sampleRateAllowed;
+        bool fastAllowed = sharedBuffer || transferAllowed;
         if (!fastAllowed) {
+            ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client, not shared buffer and transfer = %s",
+                  convertTransferToText(mTransfer));
             mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
         }
     }
 
-    mNotificationFramesAct = mNotificationFramesReq;
-
-    size_t frameCount = mReqFrameCount;
-    if (!audio_has_proportional_frames(mFormat)) {
-
-        if (mSharedBuffer != 0) {
-            // Same comment as below about ignoring frameCount parameter for set()
-            frameCount = mSharedBuffer->size();
-        } else if (frameCount == 0) {
-            frameCount = mAfFrameCount;
-        }
-        if (mNotificationFramesAct != frameCount) {
-            mNotificationFramesAct = frameCount;
-        }
-    } else if (mSharedBuffer != 0) {
-        // FIXME: Ensure client side memory buffers need
-        // not have additional alignment beyond sample
-        // (e.g. 16 bit stereo accessed as 32 bit frame).
-        size_t alignment = audio_bytes_per_sample(mFormat);
-        if (alignment & 1) {
-            // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
-            alignment = 1;
-        }
-        if (mChannelCount > 1) {
-            // More than 2 channels does not require stronger alignment than stereo
-            alignment <<= 1;
-        }
-        if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) {
-            ALOGE("Invalid buffer alignment: address %p, channel count %u",
-                    mSharedBuffer->pointer(), mChannelCount);
-            status = BAD_VALUE;
-            goto release;
-        }
-
-        // When initializing a shared buffer AudioTrack via constructors,
-        // there's no frameCount parameter.
-        // But when initializing a shared buffer AudioTrack via set(),
-        // there _is_ a frameCount parameter.  We silently ignore it.
-        frameCount = mSharedBuffer->size() / mFrameSize;
+    IAudioFlinger::CreateTrackInput input;
+    if (mStreamType != AUDIO_STREAM_DEFAULT) {
+        stream_type_to_audio_attributes(mStreamType, &input.attr);
     } else {
-        size_t minFrameCount = 0;
-        // For fast tracks the frame count calculations and checks are mostly done by server,
-        // but we try to respect the application's request for notifications per buffer.
-        if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
-            if (mNotificationsPerBufferReq > 0) {
-                // Avoid possible arithmetic overflow during multiplication.
-                // mNotificationsPerBuffer is clamped to a small integer earlier, so it is unlikely.
-                if (mNotificationsPerBufferReq > SIZE_MAX / afFrameCountHAL) {
-                    ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu",
-                            mNotificationsPerBufferReq, afFrameCountHAL);
-                } else {
-                    minFrameCount = afFrameCountHAL * mNotificationsPerBufferReq;
-                }
-            }
-        } else {
-            // for normal tracks precompute the frame count based on speed.
-            const float speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
-                            max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
-            minFrameCount = calculateMinFrameCount(
-                    mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate,
-                    speed /*, 0 mNotificationsPerBufferReq*/);
-        }
-        if (frameCount < minFrameCount) {
-            frameCount = minFrameCount;
-        }
+        input.attr = mAttributes;
     }
-
-    audio_output_flags_t flags = mFlags;
-
-    pid_t tid = -1;
+    input.config = AUDIO_CONFIG_INITIALIZER;
+    input.config.sample_rate = mSampleRate;
+    input.config.channel_mask = mChannelMask;
+    input.config.format = mFormat;
+    input.config.offload_info = mOffloadInfoCopy;
+    input.clientInfo.clientUid = mClientUid;
+    input.clientInfo.clientPid = mClientPid;
+    input.clientInfo.clientTid = -1;
     if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
         // It is currently meaningless to request SCHED_FIFO for a Java thread.  Even if the
         // application-level code follows all non-blocking design rules, the language runtime
         // doesn't also follow those rules, so the thread will not benefit overall.
         if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
-            tid = mAudioTrackThread->getTid();
+            input.clientInfo.clientTid = mAudioTrackThread->getTid();
         }
     }
+    input.sharedBuffer = mSharedBuffer;
+    input.notificationsPerBuffer = mNotificationsPerBufferReq;
+    input.speed = 1.0;
+    if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
+            (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
+        input.speed  = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
+                        max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
+    }
+    input.flags = mFlags;
+    input.frameCount = mReqFrameCount;
+    input.notificationFrameCount = mNotificationFramesReq;
+    input.selectedDeviceId = mSelectedDeviceId;
+    input.sessionId = mSessionId;
 
-    size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,
-                                // but we will still need the original value also
-    audio_session_t originalSessionId = mSessionId;
-    sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
-                                                      mSampleRate,
-                                                      mFormat,
-                                                      mChannelMask,
-                                                      &temp,
-                                                      &flags,
-                                                      mSharedBuffer,
+    IAudioFlinger::CreateTrackOutput output;
+
+    sp<IAudioTrack> track = audioFlinger->createTrack(input,
                                                       output,
-                                                      mClientPid,
-                                                      tid,
-                                                      &mSessionId,
-                                                      mClientUid,
-                                                      &status,
-                                                      mPortId);
-    ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
-            "session ID changed from %d to %d", originalSessionId, mSessionId);
+                                                      &status);
 
-    if (status != NO_ERROR) {
-        ALOGE("AudioFlinger could not create track, status: %d", status);
-        goto release;
+    if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
+        ALOGE("AudioFlinger could not create track, status: %d output %d", status, output.outputId);
+        if (status == NO_ERROR) {
+            status = NO_INIT;
+        }
+        goto exit;
     }
     ALOG_ASSERT(track != 0);
 
+    mFrameCount = output.frameCount;
+    mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
+    mRoutedDeviceId = output.selectedDeviceId;
+    mSessionId = output.sessionId;
+
+    mSampleRate = output.sampleRate;
+    if (mOriginalSampleRate == 0) {
+        mOriginalSampleRate = mSampleRate;
+    }
+
+    mAfFrameCount = output.afFrameCount;
+    mAfSampleRate = output.afSampleRate;
+    mAfLatency = output.afLatencyMs;
+
+    mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
+
     // AudioFlinger now owns the reference to the I/O handle,
     // so we are no longer responsible for releasing it.
 
@@ -1521,13 +1492,13 @@
     if (iMem == 0) {
         ALOGE("Could not get control block");
         status = NO_INIT;
-        goto release;
+        goto exit;
     }
     void *iMemPointer = iMem->pointer();
     if (iMemPointer == NULL) {
         ALOGE("Could not get control block pointer");
         status = NO_INIT;
-        goto release;
+        goto exit;
     }
     // invariant that mAudioTrack != 0 is true only after set() returns successfully
     if (mAudioTrack != 0) {
@@ -1540,66 +1511,33 @@
 
     audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
     mCblk = cblk;
-    // note that temp is the (possibly revised) value of frameCount
-    if (temp < frameCount || (frameCount == 0 && temp == 0)) {
-        // In current design, AudioTrack client checks and ensures frame count validity before
-        // passing it to AudioFlinger so AudioFlinger should not return a different value except
-        // for fast track as it uses a special method of assigning frame count.
-        ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
-    }
-    frameCount = temp;
 
     mAwaitBoost = false;
     if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
-        if (flags & AUDIO_OUTPUT_FLAG_FAST) {
-            ALOGI("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu", frameCount, temp);
+        if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
+            ALOGI("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
+                  mReqFrameCount, mFrameCount);
             if (!mThreadCanCallJava) {
                 mAwaitBoost = true;
             }
         } else {
-            ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu", frameCount,
-                    temp);
+            ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu", mReqFrameCount,
+                  mFrameCount);
         }
     }
-    mFlags = flags;
-
-    // Make sure that application is notified with sufficient margin before underrun.
-    // The client can divide the AudioTrack buffer into sub-buffers,
-    // and expresses its desire to server as the notification frame count.
-    if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) {
-        size_t maxNotificationFrames;
-        if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
-            // notify every HAL buffer, regardless of the size of the track buffer
-            maxNotificationFrames = afFrameCountHAL;
-        } else {
-            // For normal tracks, use at least double-buffering if no sample rate conversion,
-            // or at least triple-buffering if there is sample rate conversion
-            const int nBuffering = mOriginalSampleRate == mAfSampleRate ? 2 : 3;
-            maxNotificationFrames = frameCount / nBuffering;
-        }
-        if (mNotificationFramesAct == 0 || mNotificationFramesAct > maxNotificationFrames) {
-            if (mNotificationFramesAct == 0) {
-                ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu",
-                    maxNotificationFrames, frameCount);
-            } else {
-                ALOGW("Client adjusted notificationFrames from %u to %zu for frameCount %zu",
-                    mNotificationFramesAct, maxNotificationFrames, frameCount);
-            }
-            mNotificationFramesAct = (uint32_t) maxNotificationFrames;
-        }
-    }
+    mFlags = output.flags;
 
     //mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
-    if (mDeviceCallback != 0 && mOutput != output) {
+    if (mDeviceCallback != 0 && mOutput != output.outputId) {
         if (mOutput != AUDIO_IO_HANDLE_NONE) {
             AudioSystem::removeAudioDeviceCallback(this, mOutput);
         }
-        AudioSystem::addAudioDeviceCallback(this, output);
+        AudioSystem::addAudioDeviceCallback(this, output.outputId);
         callbackAdded = true;
     }
 
     // We retain a copy of the I/O handle, but don't own the reference
-    mOutput = output;
+    mOutput = output.outputId;
     mRefreshRemaining = true;
 
     // Starting address of buffers in shared memory.  If there is a shared buffer, buffers
@@ -1614,18 +1552,16 @@
         if (buffers == NULL) {
             ALOGE("Could not get buffer pointer");
             status = NO_INIT;
-            goto release;
+            goto exit;
         }
     }
 
     mAudioTrack->attachAuxEffect(mAuxEffectId);
-    mFrameCount = frameCount;
-    updateLatency_l();  // this refetches mAfLatency and sets mLatency
 
     // If IAudioTrack is re-created, don't let the requested frameCount
     // decrease.  This can confuse clients that cache frameCount().
-    if (frameCount > mReqFrameCount) {
-        mReqFrameCount = frameCount;
+    if (mFrameCount > mReqFrameCount) {
+        mReqFrameCount = mFrameCount;
     }
 
     // reset server position to 0 as we have new cblk.
@@ -1634,9 +1570,9 @@
     // update proxy
     if (mSharedBuffer == 0) {
         mStaticProxy.clear();
-        mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
+        mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
     } else {
-        mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
+        mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
         mProxy = mStaticProxy;
     }
 
@@ -1659,18 +1595,17 @@
     mDeathNotifier = new DeathNotifier(this);
     IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
 
-    return NO_ERROR;
     }
 
-release:
-    AudioSystem::releaseOutput(output, streamType, mSessionId);
-    if (callbackAdded) {
+exit:
+    if (status != NO_ERROR && callbackAdded) {
         // note: mOutput is always valid is callbackAdded is true
         AudioSystem::removeAudioDeviceCallback(this, mOutput);
     }
-    if (status == NO_ERROR) {
-        status = NO_INIT;
-    }
+
+    mStatus = status;
+
+    // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
     return status;
 }
 
@@ -2314,6 +2249,16 @@
         staticPosition = mStaticProxy->getPosition().unsignedValue();
     }
 
+    // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
+    // causes a lot of churn on the service side, and it can reject starting
+    // playback of a previously created track. May also apply to other cases.
+    const int INITIAL_RETRIES = 3;
+    int retries = INITIAL_RETRIES;
+retry:
+    if (retries < INITIAL_RETRIES) {
+        // See the comment for clearAudioConfigCache at the start of the function.
+        AudioSystem::clearAudioConfigCache();
+    }
     mFlags = mOrigFlags;
 
     // If a new IAudioTrack is successfully created, createTrack_l() will modify the
@@ -2322,7 +2267,10 @@
     // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
     status_t result = createTrack_l();
 
-    if (result == NO_ERROR) {
+    if (result != NO_ERROR) {
+        ALOGW("%s(): createTrack_l failed, do not retry", __func__);
+        retries = 0;
+    } else {
         // take the frames that will be lost by track recreation into account in saved position
         // For streaming tracks, this is the amount we obtained from the user/client
         // (not the number actually consumed at the server - those are already lost).
@@ -2367,7 +2315,10 @@
         mFramesWrittenAtRestore = mFramesWrittenServerOffset;
     }
     if (result != NO_ERROR) {
-        ALOGW("restoreTrack_l() failed status %d", result);
+        ALOGW("%s() failed status %d, retries %d", __func__, result, retries);
+        if (--retries > 0) {
+            goto retry;
+        }
         mState = STATE_STOPPED;
         mReleased = 0;
     }
@@ -2406,8 +2357,8 @@
         return true; // static tracks do not have issues with buffer sizing.
     }
     const size_t minFrameCount =
-            calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed
-                /*, 0 mNotificationsPerBufferReq*/);
+            AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
+                                            sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
     const bool allowed = mFrameCount >= minFrameCount;
     ALOGD_IF(!allowed,
             "isSampleRateSpeedAllowed_l denied "
@@ -2424,6 +2375,17 @@
     return mAudioTrack->setParameters(keyValuePairs);
 }
 
+status_t AudioTrack::selectPresentation(int presentationId, int programId)
+{
+    AutoMutex lock(mLock);
+    AudioParameter param = AudioParameter();
+    param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
+    param.addInt(String8(AudioParameter::keyProgramId), programId);
+    ALOGV("PresentationId/ProgramId[%s]",param.toString().string());
+
+    return mAudioTrack->setParameters(param.toString());
+}
+
 VolumeShaper::Status AudioTrack::applyVolumeShaper(
         const sp<VolumeShaper::Configuration>& configuration,
         const sp<VolumeShaper::Operation>& operation)
@@ -2823,23 +2785,28 @@
 
 status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
 {
-
-    const size_t SIZE = 256;
-    char buffer[SIZE];
     String8 result;
 
     result.append(" AudioTrack::dump\n");
-    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n", mStreamType,
-            mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
-    result.append(buffer);
-    snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%zu)\n", mFormat,
-            mChannelCount, mFrameCount);
-    result.append(buffer);
-    snprintf(buffer, 255, "  sample rate(%u), speed(%f), status(%d)\n",
-            mSampleRate, mPlaybackRate.mSpeed, mStatus);
-    result.append(buffer);
-    snprintf(buffer, 255, "  state(%d), latency (%d)\n", mState, mLatency);
-    result.append(buffer);
+    result.appendFormat("  status(%d), state(%d), session Id(%d), flags(%#x)\n",
+                        mStatus, mState, mSessionId, mFlags);
+    result.appendFormat("  stream type(%d), left - right volume(%f, %f)\n",
+                        (mStreamType == AUDIO_STREAM_DEFAULT) ?
+                                audio_attributes_to_stream_type(&mAttributes) : mStreamType,
+                        mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
+    result.appendFormat("  format(%#x), channel mask(%#x), channel count(%u)\n",
+                  mFormat, mChannelMask, mChannelCount);
+    result.appendFormat("  sample rate(%u), original sample rate(%u), speed(%f)\n",
+                  mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
+    result.appendFormat("  frame count(%zu), req. frame count(%zu)\n",
+                  mFrameCount, mReqFrameCount);
+    result.appendFormat("  notif. frame count(%u), req. notif. frame count(%u),"
+            " req. notif. per buff(%u)\n",
+             mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
+    result.appendFormat("  latency (%d), selected device Id(%d), routed device Id(%d)\n",
+                        mLatency, mSelectedDeviceId, mRoutedDeviceId);
+    result.appendFormat("  output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
+                        mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
     ::write(fd, result.string(), result.size());
     return NO_ERROR;
 }

diff --git a/media/libaudioclient/AudioTrackShared.cpp b/media/libaudioclient/AudioTrackShared.cpp
index 7bf4f99..dced3c4 100644
--- a/media/libaudioclient/AudioTrackShared.cpp
+++ b/media/libaudioclient/AudioTrackShared.cpp

@@ -393,19 +393,50 @@
 
 // ---------------------------------------------------------------------------
 
-__attribute__((no_sanitize("integer")))
 void AudioTrackClientProxy::flush()
 {
+    sendStreamingFlushStop(true /* flush */);
+}
+
+void AudioTrackClientProxy::stop()
+{
+    sendStreamingFlushStop(false /* flush */);
+}
+
+// Sets the client-written mFlush and mStop positions, which control server behavior.
+//
+// @param flush indicates whether the operation is a flush or stop.
+// A client stop sets mStop to the current write position;
+// the server will not read past this point until start() or subsequent flush().
+// A client flush sets both mStop and mFlush to the current write position.
+// This advances the server read limit (if previously set) and on the next
+// server read advances the server read position to this limit.
+//
+void AudioTrackClientProxy::sendStreamingFlushStop(bool flush)
+{
+    // TODO: Replace this by 64 bit counters - avoids wrap complication.
     // This works for mFrameCountP2 <= 2^30
-    size_t increment = mFrameCountP2 << 1;
-    size_t mask = increment - 1;
-    audio_track_cblk_t* cblk = mCblk;
     // mFlush is 32 bits concatenated as [ flush_counter ] [ newfront_offset ]
     // Should newFlush = cblk->u.mStreaming.mRear?  Only problem is
     // if you want to flush twice to the same rear location after a 32 bit wrap.
-    int32_t newFlush = (cblk->u.mStreaming.mRear & mask) |
-                        ((cblk->u.mStreaming.mFlush & ~mask) + increment);
-    android_atomic_release_store(newFlush, &cblk->u.mStreaming.mFlush);
+
+    const size_t increment = mFrameCountP2 << 1;
+    const size_t mask = increment - 1;
+    // No need for client atomic synchronization on mRear, mStop, mFlush
+    // as AudioTrack client only read/writes to them under client lock. Server only reads.
+    const int32_t rearMasked = mCblk->u.mStreaming.mRear & mask;
+
+    // update stop before flush so that the server front
+    // never advances beyond a (potential) previous stop's rear limit.
+    int32_t stopBits; // the following add can overflow
+    __builtin_add_overflow(mCblk->u.mStreaming.mStop & ~mask, increment, &stopBits);
+    android_atomic_release_store(rearMasked | stopBits, &mCblk->u.mStreaming.mStop);
+
+    if (flush) {
+        int32_t flushBits; // the following add can overflow
+        __builtin_add_overflow(mCblk->u.mStreaming.mFlush & ~mask, increment, &flushBits);
+        android_atomic_release_store(rearMasked | flushBits, &mCblk->u.mStreaming.mFlush);
+    }
 }
 
 bool AudioTrackClientProxy::clearStreamEndDone() {
@@ -540,6 +571,11 @@
     LOG_ALWAYS_FATAL("static flush");
 }
 
+void StaticAudioTrackClientProxy::stop()
+{
+    ; // no special handling required for static tracks.
+}
+
 void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int loopCount)
 {
     // This can only happen on a 64-bit client
@@ -638,6 +674,7 @@
     if (flush != mFlush) {
         ALOGV("ServerProxy::flushBufferIfNeeded() mStreaming.mFlush = 0x%x, mFlush = 0x%0x",
                 flush, mFlush);
+        // shouldn't matter, but for range safety use mRear instead of getRear().
         int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
         int32_t front = cblk->u.mStreaming.mFront;
 
@@ -677,6 +714,46 @@
 }
 
 __attribute__((no_sanitize("integer")))
+int32_t AudioTrackServerProxy::getRear() const
+{
+    const int32_t stop = android_atomic_acquire_load(&mCblk->u.mStreaming.mStop);
+    const int32_t rear = android_atomic_acquire_load(&mCblk->u.mStreaming.mRear);
+    const int32_t stopLast = mStopLast.load(std::memory_order_acquire);
+    if (stop != stopLast) {
+        const int32_t front = mCblk->u.mStreaming.mFront;
+        const size_t overflowBit = mFrameCountP2 << 1;
+        const size_t mask = overflowBit - 1;
+        int32_t newRear = (rear & ~mask) | (stop & mask);
+        ssize_t filled = newRear - front;
+        // overflowBit is unsigned, so cast to signed for comparison.
+        if (filled >= (ssize_t)overflowBit) {
+            // front and rear offsets span the overflow bit of the p2 mask
+            // so rebasing newRear on the rear offset is off by the overflow bit.
+            ALOGV("stop wrap: filled %zx >= overflowBit %zx", filled, overflowBit);
+            newRear -= overflowBit;
+            filled -= overflowBit;
+        }
+        if (0 <= filled && (size_t) filled <= mFrameCount) {
+            // we're stopped, return the stop level as newRear
+            return newRear;
+        }
+
+        // A corrupt stop. Log error and ignore.
+        ALOGE("mStopLast %#x -> stop %#x, front %#x, rear %#x, mask %#x, newRear %#x, "
+                "filled %zd=%#x",
+                stopLast, stop, front, rear,
+                (unsigned)mask, newRear, filled, (unsigned)filled);
+        // Don't reset mStopLast as this is const.
+    }
+    return rear;
+}
+
+void AudioTrackServerProxy::start()
+{
+    mStopLast = android_atomic_acquire_load(&mCblk->u.mStreaming.mStop);
+}
+
+__attribute__((no_sanitize("integer")))
 status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
 {
     LOG_ALWAYS_FATAL_IF(buffer == NULL || buffer->mFrameCount == 0,
@@ -693,7 +770,7 @@
     // See notes on barriers at ClientProxy::obtainBuffer()
     if (mIsOut) {
         flushBufferIfNeeded(); // might modify mFront
-        rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
+        rear = getRear();
         front = cblk->u.mStreaming.mFront;
     } else {
         front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
@@ -825,8 +902,7 @@
         // FIXME should return an accurate value, but over-estimate is better than under-estimate
         return mFrameCount;
     }
-    // the acquire might not be necessary since not doing a subsequent read
-    int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
+    const int32_t rear = getRear();
     ssize_t filled = rear - cblk->u.mStreaming.mFront;
     // pipe should not already be overfull
     if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
@@ -852,7 +928,7 @@
     if (flush != mFlush) {
         return mFrameCount;
     }
-    const int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
+    const int32_t rear = getRear();
     const ssize_t filled = rear - cblk->u.mStreaming.mFront;
     if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
         return 0; // error condition, silently return 0.
@@ -1149,6 +1225,12 @@
     }
 }
 
+int32_t StaticAudioTrackServerProxy::getRear() const
+{
+    LOG_ALWAYS_FATAL("getRear() not permitted for static tracks");
+    return 0;
+}
+
 // ---------------------------------------------------------------------------
 
 }   // namespace android

diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 14feada..9f3b742 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp

@@ -22,7 +22,11 @@
 #include <stdint.h>
 #include <sys/types.h>
 
+#include <binder/IPCThreadState.h>
 #include <binder/Parcel.h>
+#include <cutils/multiuser.h>
+#include <media/TimeCheck.h>
+#include <private/android_filesystem_config.h>
 
 #include "IAudioFlinger.h"
 
@@ -30,7 +34,7 @@
 
 enum {
     CREATE_TRACK = IBinder::FIRST_CALL_TRANSACTION,
-    OPEN_RECORD,
+    CREATE_RECORD,
     SAMPLE_RATE,
     RESERVED,   // obsolete, was CHANNEL_COUNT
     FORMAT,
@@ -47,6 +51,7 @@
     SET_MODE,
     SET_MIC_MUTE,
     GET_MIC_MUTE,
+    SET_RECORD_SILENCED,
     SET_PARAMETERS,
     GET_PARAMETERS,
     REGISTER_CLIENT,
@@ -83,6 +88,7 @@
     GET_AUDIO_HW_SYNC_FOR_SESSION,
     SYSTEM_READY,
     FRAME_COUNT_HAL,
+    GET_MICROPHONES,
 };
 
 #define MAX_ITEMS_PER_LIST 1024
@@ -95,182 +101,74 @@
     {
     }
 
-    virtual sp<IAudioTrack> createTrack(
-                                audio_stream_type_t streamType,
-                                uint32_t sampleRate,
-                                audio_format_t format,
-                                audio_channel_mask_t channelMask,
-                                size_t *pFrameCount,
-                                audio_output_flags_t *flags,
-                                const sp<IMemory>& sharedBuffer,
-                                audio_io_handle_t output,
-                                pid_t pid,
-                                pid_t tid,
-                                audio_session_t *sessionId,
-                                int clientUid,
-                                status_t *status,
-                                audio_port_handle_t portId)
+    virtual sp<IAudioTrack> createTrack(const CreateTrackInput& input,
+                                        CreateTrackOutput& output,
+                                        status_t *status)
     {
         Parcel data, reply;
         sp<IAudioTrack> track;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeInt32((int32_t) streamType);
-        data.writeInt32(sampleRate);
-        data.writeInt32(format);
-        data.writeInt32(channelMask);
-        size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
-        data.writeInt64(frameCount);
-        audio_output_flags_t lFlags = flags != NULL ? *flags : AUDIO_OUTPUT_FLAG_NONE;
-        data.writeInt32(lFlags);
-        // haveSharedBuffer
-        if (sharedBuffer != 0) {
-            data.writeInt32(true);
-            data.writeStrongBinder(IInterface::asBinder(sharedBuffer));
-        } else {
-            data.writeInt32(false);
+
+        if (status == nullptr) {
+            return track;
         }
-        data.writeInt32((int32_t) output);
-        data.writeInt32((int32_t) pid);
-        data.writeInt32((int32_t) tid);
-        audio_session_t lSessionId = AUDIO_SESSION_ALLOCATE;
-        if (sessionId != NULL) {
-            lSessionId = *sessionId;
-        }
-        data.writeInt32(lSessionId);
-        data.writeInt32(clientUid);
-        data.writeInt32(portId);
+
+        input.writeToParcel(&data);
+
         status_t lStatus = remote()->transact(CREATE_TRACK, data, &reply);
         if (lStatus != NO_ERROR) {
-            ALOGE("createTrack error: %s", strerror(-lStatus));
-        } else {
-            frameCount = reply.readInt64();
-            if (pFrameCount != NULL) {
-                *pFrameCount = frameCount;
-            }
-            lFlags = (audio_output_flags_t)reply.readInt32();
-            if (flags != NULL) {
-                *flags = lFlags;
-            }
-            lSessionId = (audio_session_t) reply.readInt32();
-            if (sessionId != NULL) {
-                *sessionId = lSessionId;
-            }
-            lStatus = reply.readInt32();
-            track = interface_cast<IAudioTrack>(reply.readStrongBinder());
-            if (lStatus == NO_ERROR) {
-                if (track == 0) {
-                    ALOGE("createTrack should have returned an IAudioTrack");
-                    lStatus = UNKNOWN_ERROR;
-                }
-            } else {
-                if (track != 0) {
-                    ALOGE("createTrack returned an IAudioTrack but with status %d", lStatus);
-                    track.clear();
-                }
-            }
+            ALOGE("createTrack transaction error %d", lStatus);
+            *status = DEAD_OBJECT;
+            return track;
         }
-        if (status != NULL) {
-            *status = lStatus;
+        *status = reply.readInt32();
+        if (*status != NO_ERROR) {
+            ALOGE("createTrack returned error %d", *status);
+            return track;
         }
+        track = interface_cast<IAudioTrack>(reply.readStrongBinder());
+        if (track == 0) {
+            ALOGE("createTrack returned an NULL IAudioTrack with status OK");
+            *status = DEAD_OBJECT;
+            return track;
+        }
+        output.readFromParcel(&reply);
         return track;
     }
 
-    virtual sp<IAudioRecord> openRecord(
-                                audio_io_handle_t input,
-                                uint32_t sampleRate,
-                                audio_format_t format,
-                                audio_channel_mask_t channelMask,
-                                const String16& opPackageName,
-                                size_t *pFrameCount,
-                                audio_input_flags_t *flags,
-                                pid_t pid,
-                                pid_t tid,
-                                int clientUid,
-                                audio_session_t *sessionId,
-                                size_t *notificationFrames,
-                                sp<IMemory>& cblk,
-                                sp<IMemory>& buffers,
-                                status_t *status,
-                                audio_port_handle_t portId)
+    virtual sp<media::IAudioRecord> createRecord(const CreateRecordInput& input,
+                                                 CreateRecordOutput& output,
+                                                 status_t *status)
     {
         Parcel data, reply;
-        sp<IAudioRecord> record;
+        sp<media::IAudioRecord> record;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeInt32((int32_t) input);
-        data.writeInt32(sampleRate);
-        data.writeInt32(format);
-        data.writeInt32(channelMask);
-        data.writeString16(opPackageName);
-        size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
-        data.writeInt64(frameCount);
-        audio_input_flags_t lFlags = flags != NULL ? *flags : AUDIO_INPUT_FLAG_NONE;
-        data.writeInt32(lFlags);
-        data.writeInt32((int32_t) pid);
-        data.writeInt32((int32_t) tid);
-        data.writeInt32((int32_t) clientUid);
-        audio_session_t lSessionId = AUDIO_SESSION_ALLOCATE;
-        if (sessionId != NULL) {
-            lSessionId = *sessionId;
+
+        if (status == nullptr) {
+            return record;
         }
-        data.writeInt32(lSessionId);
-        data.writeInt64(notificationFrames != NULL ? *notificationFrames : 0);
-        data.writeInt32(portId);
-        cblk.clear();
-        buffers.clear();
-        status_t lStatus = remote()->transact(OPEN_RECORD, data, &reply);
+
+        input.writeToParcel(&data);
+
+        status_t lStatus = remote()->transact(CREATE_RECORD, data, &reply);
         if (lStatus != NO_ERROR) {
-            ALOGE("openRecord error: %s", strerror(-lStatus));
-        } else {
-            frameCount = reply.readInt64();
-            if (pFrameCount != NULL) {
-                *pFrameCount = frameCount;
-            }
-            lFlags = (audio_input_flags_t)reply.readInt32();
-            if (flags != NULL) {
-                *flags = lFlags;
-            }
-            lSessionId = (audio_session_t) reply.readInt32();
-            if (sessionId != NULL) {
-                *sessionId = lSessionId;
-            }
-            size_t lNotificationFrames = (size_t) reply.readInt64();
-            if (notificationFrames != NULL) {
-                *notificationFrames = lNotificationFrames;
-            }
-            lStatus = reply.readInt32();
-            record = interface_cast<IAudioRecord>(reply.readStrongBinder());
-            cblk = interface_cast<IMemory>(reply.readStrongBinder());
-            if (cblk != 0 && cblk->pointer() == NULL) {
-                cblk.clear();
-            }
-            buffers = interface_cast<IMemory>(reply.readStrongBinder());
-            if (buffers != 0 && buffers->pointer() == NULL) {
-                buffers.clear();
-            }
-            if (lStatus == NO_ERROR) {
-                if (record == 0) {
-                    ALOGE("openRecord should have returned an IAudioRecord");
-                    lStatus = UNKNOWN_ERROR;
-                } else if (cblk == 0) {
-                    ALOGE("openRecord should have returned a cblk");
-                    lStatus = NO_MEMORY;
-                }
-                // buffers is permitted to be 0
-            } else {
-                if (record != 0 || cblk != 0 || buffers != 0) {
-                    ALOGE("openRecord returned an IAudioRecord, cblk, "
-                          "or buffers but with status %d", lStatus);
-                }
-            }
-            if (lStatus != NO_ERROR) {
-                record.clear();
-                cblk.clear();
-                buffers.clear();
-            }
+            ALOGE("createRecord transaction error %d", lStatus);
+            *status = DEAD_OBJECT;
+            return record;
         }
-        if (status != NULL) {
-            *status = lStatus;
+        *status = reply.readInt32();
+        if (*status != NO_ERROR) {
+            ALOGE("createRecord returned error %d", *status);
+            return record;
         }
+
+        record = interface_cast<media::IAudioRecord>(reply.readStrongBinder());
+        if (record == 0) {
+            ALOGE("createRecord returned a NULL IAudioRecord with status OK");
+            *status = DEAD_OBJECT;
+            return record;
+        }
+        output.readFromParcel(&reply);
         return record;
     }
 
@@ -413,6 +311,15 @@
         return reply.readInt32();
     }
 
+    virtual void setRecordSilenced(uid_t uid, bool silenced)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        data.writeInt32(uid);
+        data.writeInt32(silenced ? 1 : 0);
+        remote()->transact(SET_RECORD_SILENCED, data, &reply);
+    }
+
     virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
     {
         Parcel data, reply;
@@ -804,14 +711,18 @@
         return reply.readInt64();
     }
 
-    virtual status_t setLowRamDevice(bool isLowRamDevice)
+    virtual status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory) override
     {
         Parcel data, reply;
-        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeInt32((int) isLowRamDevice);
-        remote()->transact(SET_LOW_RAM_DEVICE, data, &reply);
-        return reply.readInt32();
+
+        static_assert(NO_ERROR == 0, "NO_ERROR must be 0");
+        return data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor())
+                ?: data.writeInt32((int) isLowRamDevice)
+                ?: data.writeInt64(totalMemory)
+                ?: remote()->transact(SET_LOW_RAM_DEVICE, data, &reply)
+                ?: reply.readInt32();
     }
+
     virtual status_t listAudioPorts(unsigned int *num_ports,
                                     struct audio_port *ports)
     {
@@ -935,6 +846,18 @@
         }
         return reply.readInt64();
     }
+    virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+        status_t status = remote()->transact(GET_MICROPHONES, data, &reply);
+        if (status != NO_ERROR ||
+                (status = (status_t)reply.readInt32()) != NO_ERROR) {
+            return status;
+        }
+        status = reply.readParcelableVector(microphones);
+        return status;
+    }
 };
 
 IMPLEMENT_META_INTERFACE(AudioFlinger, "android.media.IAudioFlinger");
@@ -944,21 +867,81 @@
 status_t BnAudioFlinger::onTransact(
     uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
 {
+    // make sure transactions reserved to AudioPolicyManager do not come from other processes
+    switch (code) {
+        case SET_STREAM_VOLUME:
+        case SET_STREAM_MUTE:
+        case OPEN_OUTPUT:
+        case OPEN_DUPLICATE_OUTPUT:
+        case CLOSE_OUTPUT:
+        case SUSPEND_OUTPUT:
+        case RESTORE_OUTPUT:
+        case OPEN_INPUT:
+        case CLOSE_INPUT:
+        case INVALIDATE_STREAM:
+        case SET_VOICE_VOLUME:
+        case MOVE_EFFECTS:
+        case LOAD_HW_MODULE:
+        case LIST_AUDIO_PORTS:
+        case GET_AUDIO_PORT:
+        case CREATE_AUDIO_PATCH:
+        case RELEASE_AUDIO_PATCH:
+        case LIST_AUDIO_PATCHES:
+        case SET_AUDIO_PORT_CONFIG:
+        case SET_RECORD_SILENCED:
+            ALOGW("%s: transaction %d received from PID %d",
+                  __func__, code, IPCThreadState::self()->getCallingPid());
+            // return status only for non void methods
+            switch (code) {
+                case SET_RECORD_SILENCED:
+                    break;
+                default:
+                    reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
+                    break;
+            }
+            return OK;
+        default:
+            break;
+    }
+
+    // make sure the following transactions come from system components
+    switch (code) {
+        case SET_MASTER_VOLUME:
+        case SET_MASTER_MUTE:
+        case SET_MODE:
+        case SET_MIC_MUTE:
+        case SET_LOW_RAM_DEVICE:
+        case SYSTEM_READY: {
+            if (multiuser_get_app_id(IPCThreadState::self()->getCallingUid()) >= AID_APP_START) {
+                ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
+                      __func__, code, IPCThreadState::self()->getCallingPid(),
+                      IPCThreadState::self()->getCallingUid());
+                // return status only for non void methods
+                switch (code) {
+                    case SYSTEM_READY:
+                        break;
+                    default:
+                        reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
+                        break;
+                }
+                return OK;
+            }
+        } break;
+        default:
+            break;
+    }
+
     // Whitelist of relevant events to trigger log merging.
     // Log merging should activate during audio activity of any kind. This are considered the
     // most relevant events.
     // TODO should select more wisely the items from the list
     switch (code) {
         case CREATE_TRACK:
-        case OPEN_RECORD:
+        case CREATE_RECORD:
         case SET_MASTER_VOLUME:
         case SET_MASTER_MUTE:
-        case SET_STREAM_VOLUME:
-        case SET_STREAM_MUTE:
         case SET_MIC_MUTE:
         case SET_PARAMETERS:
-        case OPEN_INPUT:
-        case SET_VOICE_VOLUME:
         case CREATE_EFFECT:
         case SYSTEM_READY: {
             requestLogMerge();
@@ -967,77 +950,60 @@
         default:
             break;
     }
+
+    char timeCheckString[64];
+    snprintf(timeCheckString, sizeof(timeCheckString), "IAudioFlinger: %d", code);
+    TimeCheck check(timeCheckString);
+
     switch (code) {
         case CREATE_TRACK: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            int streamType = data.readInt32();
-            uint32_t sampleRate = data.readInt32();
-            audio_format_t format = (audio_format_t) data.readInt32();
-            audio_channel_mask_t channelMask = data.readInt32();
-            size_t frameCount = data.readInt64();
-            audio_output_flags_t flags = (audio_output_flags_t) data.readInt32();
-            bool haveSharedBuffer = data.readInt32() != 0;
-            sp<IMemory> buffer;
-            if (haveSharedBuffer) {
-                buffer = interface_cast<IMemory>(data.readStrongBinder());
+
+            CreateTrackInput input;
+            if (input.readFromParcel((Parcel*)&data) != NO_ERROR) {
+                reply->writeInt32(DEAD_OBJECT);
+                return NO_ERROR;
             }
-            audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
-            pid_t pid = (pid_t) data.readInt32();
-            pid_t tid = (pid_t) data.readInt32();
-            audio_session_t sessionId = (audio_session_t) data.readInt32();
-            int clientUid = data.readInt32();
-            audio_port_handle_t portId = (audio_port_handle_t) data.readInt32();
-            status_t status = NO_ERROR;
-            sp<IAudioTrack> track;
-            if ((haveSharedBuffer && (buffer == 0)) ||
-                    ((buffer != 0) && (buffer->pointer() == NULL))) {
-                ALOGW("CREATE_TRACK: cannot retrieve shared memory");
-                status = DEAD_OBJECT;
-            } else {
-                track = createTrack(
-                        (audio_stream_type_t) streamType, sampleRate, format,
-                        channelMask, &frameCount, &flags, buffer, output, pid, tid,
-                        &sessionId, clientUid, &status, portId);
-                LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
-            }
-            reply->writeInt64(frameCount);
-            reply->writeInt32(flags);
-            reply->writeInt32(sessionId);
+
+            status_t status;
+            CreateTrackOutput output;
+
+            sp<IAudioTrack> track= createTrack(input,
+                                               output,
+                                               &status);
+
+            LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
             reply->writeInt32(status);
+            if (status != NO_ERROR) {
+                return NO_ERROR;
+            }
             reply->writeStrongBinder(IInterface::asBinder(track));
+            output.writeToParcel(reply);
             return NO_ERROR;
         } break;
-        case OPEN_RECORD: {
+        case CREATE_RECORD: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            audio_io_handle_t input = (audio_io_handle_t) data.readInt32();
-            uint32_t sampleRate = data.readInt32();
-            audio_format_t format = (audio_format_t) data.readInt32();
-            audio_channel_mask_t channelMask = data.readInt32();
-            const String16& opPackageName = data.readString16();
-            size_t frameCount = data.readInt64();
-            audio_input_flags_t flags = (audio_input_flags_t) data.readInt32();
-            pid_t pid = (pid_t) data.readInt32();
-            pid_t tid = (pid_t) data.readInt32();
-            int clientUid = data.readInt32();
-            audio_session_t sessionId = (audio_session_t) data.readInt32();
-            size_t notificationFrames = data.readInt64();
-            audio_port_handle_t portId = (audio_port_handle_t) data.readInt32();
-            sp<IMemory> cblk;
-            sp<IMemory> buffers;
-            status_t status = NO_ERROR;
-            sp<IAudioRecord> record = openRecord(input,
-                    sampleRate, format, channelMask, opPackageName, &frameCount, &flags,
-                    pid, tid, clientUid, &sessionId, &notificationFrames, cblk, buffers,
-                    &status, portId);
+
+            CreateRecordInput input;
+            if (input.readFromParcel((Parcel*)&data) != NO_ERROR) {
+                reply->writeInt32(DEAD_OBJECT);
+                return NO_ERROR;
+            }
+
+            status_t status;
+            CreateRecordOutput output;
+
+            sp<media::IAudioRecord> record = createRecord(input,
+                                                          output,
+                                                          &status);
+
             LOG_ALWAYS_FATAL_IF((record != 0) != (status == NO_ERROR));
-            reply->writeInt64(frameCount);
-            reply->writeInt32(flags);
-            reply->writeInt32(sessionId);
-            reply->writeInt64(notificationFrames);
             reply->writeInt32(status);
+            if (status != NO_ERROR) {
+                return NO_ERROR;
+            }
             reply->writeStrongBinder(IInterface::asBinder(record));
-            reply->writeStrongBinder(IInterface::asBinder(cblk));
-            reply->writeStrongBinder(IInterface::asBinder(buffers));
+            output.writeToParcel(reply);
             return NO_ERROR;
         } break;
         case SAMPLE_RATE: {
@@ -1127,6 +1093,15 @@
             reply->writeInt32( getMicMute() );
             return NO_ERROR;
         } break;
+        case SET_RECORD_SILENCED: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            uid_t uid = data.readInt32();
+            audio_source_t source;
+            data.read(&source, sizeof(audio_source_t));
+            bool silenced = data.readInt32() == 1;
+            setRecordSilenced(uid, silenced);
+            return NO_ERROR;
+        } break;
         case SET_PARAMETERS: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
@@ -1364,8 +1339,13 @@
         } break;
         case SET_LOW_RAM_DEVICE: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            bool isLowRamDevice = data.readInt32() != 0;
-            reply->writeInt32(setLowRamDevice(isLowRamDevice));
+            int32_t isLowRamDevice;
+            int64_t totalMemory;
+            const status_t status =
+                    data.readInt32(&isLowRamDevice) ?:
+                    data.readInt64(&totalMemory) ?:
+                    setLowRamDevice(isLowRamDevice != 0, totalMemory);
+            (void)reply->writeInt32(status);
             return NO_ERROR;
         } break;
         case LIST_AUDIO_PORTS: {
@@ -1481,6 +1461,16 @@
             reply->writeInt64( frameCountHAL((audio_io_handle_t) data.readInt32()) );
             return NO_ERROR;
         } break;
+        case GET_MICROPHONES: {
+            CHECK_INTERFACE(IAudioFlinger, data, reply);
+            std::vector<media::MicrophoneInfo> microphones;
+            status_t status = getMicrophones(&microphones);
+            reply->writeInt32(status);
+            if (status == NO_ERROR) {
+                reply->writeParcelableVector(microphones);
+            }
+            return NO_ERROR;
+        }
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }

diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index f071a02..8cd4a85 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp

@@ -22,11 +22,13 @@
 #include <math.h>
 #include <sys/types.h>
 
+#include <binder/IPCThreadState.h>
 #include <binder/Parcel.h>
-
+#include <cutils/multiuser.h>
 #include <media/AudioEffect.h>
 #include <media/IAudioPolicyService.h>
-
+#include <media/TimeCheck.h>
+#include <private/android_filesystem_config.h>
 #include <system/audio.h>
 
 namespace android {
@@ -78,7 +80,9 @@
     SET_AUDIO_PORT_CALLBACK_ENABLED,
     SET_MASTER_MONO,
     GET_MASTER_MONO,
-    GET_STREAM_VOLUME_DB
+    GET_STREAM_VOLUME_DB,
+    GET_SURROUND_FORMATS,
+    SET_SURROUND_FORMAT_ENABLED
 };
 
 #define MAX_ITEMS_PER_LIST 1024
@@ -160,28 +164,11 @@
         return static_cast <audio_policy_forced_cfg_t> (reply.readInt32());
     }
 
-    virtual audio_io_handle_t getOutput(
-                                        audio_stream_type_t stream,
-                                        uint32_t samplingRate,
-                                        audio_format_t format,
-                                        audio_channel_mask_t channelMask,
-                                        audio_output_flags_t flags,
-                                        const audio_offload_info_t *offloadInfo)
+    virtual audio_io_handle_t getOutput(audio_stream_type_t stream)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
         data.writeInt32(static_cast <uint32_t>(stream));
-        data.writeInt32(samplingRate);
-        data.writeInt32(static_cast <uint32_t>(format));
-        data.writeInt32(channelMask);
-        data.writeInt32(static_cast <uint32_t>(flags));
-        // hasOffloadInfo
-        if (offloadInfo == NULL) {
-            data.writeInt32(0);
-        } else {
-            data.writeInt32(1);
-            data.write(offloadInfo, sizeof(audio_offload_info_t));
-        }
         remote()->transact(GET_OUTPUT, data, &reply);
         return static_cast <audio_io_handle_t> (reply.readInt32());
     }
@@ -190,6 +177,7 @@
                                         audio_io_handle_t *output,
                                         audio_session_t session,
                                         audio_stream_type_t *stream,
+                                        pid_t pid,
                                         uid_t uid,
                                         const audio_config_t *config,
                                         audio_output_flags_t flags,
@@ -233,6 +221,7 @@
                 data.writeInt32(1);
                 data.writeInt32(*stream);
             }
+            data.writeInt32(pid);
             data.writeInt32(uid);
             data.write(config, sizeof(audio_config_t));
             data.writeInt32(static_cast <uint32_t>(flags));
@@ -299,6 +288,7 @@
                                      audio_session_t session,
                                      pid_t pid,
                                      uid_t uid,
+                                     const String16& opPackageName,
                                      const audio_config_base_t *config,
                                      audio_input_flags_t flags,
                                      audio_port_handle_t *selectedDeviceId,
@@ -327,6 +317,7 @@
         data.writeInt32(session);
         data.writeInt32(pid);
         data.writeInt32(uid);
+        data.writeString16(opPackageName);
         data.write(config, sizeof(audio_config_base_t));
         data.writeInt32(flags);
         data.writeInt32(*selectedDeviceId);
@@ -345,35 +336,33 @@
         return NO_ERROR;
     }
 
-    virtual status_t startInput(audio_io_handle_t input,
-                                audio_session_t session)
+    virtual status_t startInput(audio_port_handle_t portId,
+                                bool *silenced)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
-        data.writeInt32(input);
-        data.writeInt32(session);
+        data.writeInt32(portId);
+        data.writeInt32(*silenced ? 1 : 0);
         remote()->transact(START_INPUT, data, &reply);
-        return static_cast <status_t> (reply.readInt32());
+        status_t status = static_cast <status_t> (reply.readInt32());
+        *silenced = reply.readInt32() == 1;
+        return status;
     }
 
-    virtual status_t stopInput(audio_io_handle_t input,
-                               audio_session_t session)
+    virtual status_t stopInput(audio_port_handle_t portId)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
-        data.writeInt32(input);
-        data.writeInt32(session);
+        data.writeInt32(portId);
         remote()->transact(STOP_INPUT, data, &reply);
         return static_cast <status_t> (reply.readInt32());
     }
 
-    virtual void releaseInput(audio_io_handle_t input,
-                              audio_session_t session)
+    virtual void releaseInput(audio_port_handle_t portId)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
-        data.writeInt32(input);
-        data.writeInt32(session);
+        data.writeInt32(portId);
         remote()->transact(RELEASE_INPUT, data, &reply);
     }
 
@@ -842,16 +831,127 @@
         }
         return reply.readFloat();
     }
+
+    virtual status_t getSurroundFormats(unsigned int *numSurroundFormats,
+                                        audio_format_t *surroundFormats,
+                                        bool *surroundFormatsEnabled,
+                                        bool reported)
+    {
+        if (numSurroundFormats == NULL || (*numSurroundFormats != 0 &&
+                (surroundFormats == NULL || surroundFormatsEnabled == NULL))) {
+            return BAD_VALUE;
+        }
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        unsigned int numSurroundFormatsReq = *numSurroundFormats;
+        data.writeUint32(numSurroundFormatsReq);
+        data.writeBool(reported);
+        status_t status = remote()->transact(GET_SURROUND_FORMATS, data, &reply);
+        if (status == NO_ERROR && (status = (status_t)reply.readInt32()) == NO_ERROR) {
+            *numSurroundFormats = reply.readUint32();
+        }
+        if (status == NO_ERROR) {
+            if (numSurroundFormatsReq > *numSurroundFormats) {
+                numSurroundFormatsReq = *numSurroundFormats;
+            }
+            if (numSurroundFormatsReq > 0) {
+                status = reply.read(surroundFormats,
+                                    numSurroundFormatsReq * sizeof(audio_format_t));
+                if (status != NO_ERROR) {
+                    return status;
+                }
+                status = reply.read(surroundFormatsEnabled,
+                                    numSurroundFormatsReq * sizeof(bool));
+            }
+        }
+        return status;
+    }
+
+    virtual status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(audioFormat);
+        data.writeBool(enabled);
+        status_t status = remote()->transact(SET_SURROUND_FORMAT_ENABLED, data, &reply);
+        if (status != NO_ERROR) {
+            return status;
+        }
+        return reply.readInt32();
+    }
 };
 
 IMPLEMENT_META_INTERFACE(AudioPolicyService, "android.media.IAudioPolicyService");
 
 // ----------------------------------------------------------------------
 
-
 status_t BnAudioPolicyService::onTransact(
     uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
 {
+    // make sure transactions reserved to AudioFlinger do not come from other processes
+    switch (code) {
+        case START_OUTPUT:
+        case STOP_OUTPUT:
+        case RELEASE_OUTPUT:
+        case GET_INPUT_FOR_ATTR:
+        case START_INPUT:
+        case STOP_INPUT:
+        case RELEASE_INPUT:
+        case GET_STRATEGY_FOR_STREAM:
+        case GET_OUTPUT_FOR_EFFECT:
+        case REGISTER_EFFECT:
+        case UNREGISTER_EFFECT:
+        case SET_EFFECT_ENABLED:
+        case GET_OUTPUT_FOR_ATTR:
+        case ACQUIRE_SOUNDTRIGGER_SESSION:
+        case RELEASE_SOUNDTRIGGER_SESSION:
+            ALOGW("%s: transaction %d received from PID %d",
+                  __func__, code, IPCThreadState::self()->getCallingPid());
+            // return status only for non void methods
+            switch (code) {
+                case RELEASE_OUTPUT:
+                case RELEASE_INPUT:
+                    break;
+                default:
+                    reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
+                    break;
+            }
+            return OK;
+        default:
+            break;
+    }
+
+    // make sure the following transactions come from system components
+    switch (code) {
+        case SET_DEVICE_CONNECTION_STATE:
+        case HANDLE_DEVICE_CONFIG_CHANGE:
+        case SET_PHONE_STATE:
+//FIXME: Allow SET_FORCE_USE calls from system apps until a better use case routing API is available
+//      case SET_FORCE_USE:
+        case INIT_STREAM_VOLUME:
+        case SET_STREAM_VOLUME:
+        case REGISTER_POLICY_MIXES:
+        case SET_MASTER_MONO:
+        case START_AUDIO_SOURCE:
+        case STOP_AUDIO_SOURCE:
+        case GET_SURROUND_FORMATS:
+        case SET_SURROUND_FORMAT_ENABLED: {
+            if (multiuser_get_app_id(IPCThreadState::self()->getCallingUid()) >= AID_APP_START) {
+                ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
+                      __func__, code, IPCThreadState::self()->getCallingPid(),
+                      IPCThreadState::self()->getCallingUid());
+                reply->writeInt32(static_cast<int32_t> (INVALID_OPERATION));
+                return OK;
+            }
+        } break;
+        default:
+            break;
+    }
+
+    char timeCheckString[64];
+    snprintf(timeCheckString, sizeof(timeCheckString), "IAudioPolicyService: %d", code);
+    TimeCheck check(timeCheckString);
+
     switch (code) {
         case SET_DEVICE_CONNECTION_STATE: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
@@ -934,22 +1034,7 @@
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
             audio_stream_type_t stream =
                     static_cast <audio_stream_type_t>(data.readInt32());
-            uint32_t samplingRate = data.readInt32();
-            audio_format_t format = (audio_format_t) data.readInt32();
-            audio_channel_mask_t channelMask = data.readInt32();
-            audio_output_flags_t flags =
-                    static_cast <audio_output_flags_t>(data.readInt32());
-            bool hasOffloadInfo = data.readInt32() != 0;
-            audio_offload_info_t offloadInfo = {};
-            if (hasOffloadInfo) {
-                data.read(&offloadInfo, sizeof(audio_offload_info_t));
-            }
-            audio_io_handle_t output = getOutput(stream,
-                                                 samplingRate,
-                                                 format,
-                                                 channelMask,
-                                                 flags,
-                                                 hasOffloadInfo ? &offloadInfo : NULL);
+            audio_io_handle_t output = getOutput(stream);
             reply->writeInt32(static_cast <int>(output));
             return NO_ERROR;
         } break;
@@ -968,6 +1053,7 @@
             if (hasStream) {
                 stream = (audio_stream_type_t)data.readInt32();
             }
+            pid_t pid = (pid_t)data.readInt32();
             uid_t uid = (uid_t)data.readInt32();
             audio_config_t config;
             memset(&config, 0, sizeof(audio_config_t));
@@ -978,7 +1064,7 @@
             audio_port_handle_t portId = (audio_port_handle_t)data.readInt32();
             audio_io_handle_t output = 0;
             status_t status = getOutputForAttr(hasAttributes ? &attr : NULL,
-                    &output, session, &stream, uid,
+                    &output, session, &stream, pid, uid,
                     &config,
                     flags, &selectedDeviceId, &portId);
             reply->writeInt32(status);
@@ -1031,6 +1117,7 @@
             audio_session_t session = (audio_session_t)data.readInt32();
             pid_t pid = (pid_t)data.readInt32();
             uid_t uid = (uid_t)data.readInt32();
+            const String16 opPackageName = data.readString16();
             audio_config_base_t config;
             memset(&config, 0, sizeof(audio_config_base_t));
             data.read(&config, sizeof(audio_config_base_t));
@@ -1038,7 +1125,7 @@
             audio_port_handle_t selectedDeviceId = (audio_port_handle_t) data.readInt32();
             audio_port_handle_t portId = (audio_port_handle_t)data.readInt32();
             status_t status = getInputForAttr(&attr, &input, session, pid, uid,
-                                              &config,
+                                              opPackageName, &config,
                                               flags, &selectedDeviceId, &portId);
             reply->writeInt32(status);
             if (status == NO_ERROR) {
@@ -1051,25 +1138,25 @@
 
         case START_INPUT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_io_handle_t input = static_cast <audio_io_handle_t>(data.readInt32());
-            audio_session_t session = static_cast <audio_session_t>(data.readInt32());
-            reply->writeInt32(static_cast <uint32_t>(startInput(input, session)));
+            audio_port_handle_t portId = static_cast <audio_port_handle_t>(data.readInt32());
+            bool silenced = data.readInt32() == 1;
+            status_t status = startInput(portId, &silenced);
+            reply->writeInt32(static_cast <uint32_t>(status));
+            reply->writeInt32(silenced ? 1 : 0);
             return NO_ERROR;
         } break;
 
         case STOP_INPUT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_io_handle_t input = static_cast <audio_io_handle_t>(data.readInt32());
-            audio_session_t session = static_cast <audio_session_t>(data.readInt32());
-            reply->writeInt32(static_cast <uint32_t>(stopInput(input, session)));
+            audio_port_handle_t portId = static_cast <audio_port_handle_t>(data.readInt32());
+            reply->writeInt32(static_cast <uint32_t>(stopInput(portId)));
             return NO_ERROR;
         } break;
 
         case RELEASE_INPUT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_io_handle_t input = static_cast <audio_io_handle_t>(data.readInt32());
-            audio_session_t session = static_cast <audio_session_t>(data.readInt32());
-            releaseInput(input, session);
+            audio_port_handle_t portId = static_cast <audio_port_handle_t>(data.readInt32());
+            releaseInput(portId);
             return NO_ERROR;
         } break;
 
@@ -1456,6 +1543,50 @@
             return NO_ERROR;
         }
 
+        case GET_SURROUND_FORMATS: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            unsigned int numSurroundFormatsReq = data.readUint32();
+            if (numSurroundFormatsReq > MAX_ITEMS_PER_LIST) {
+                numSurroundFormatsReq = MAX_ITEMS_PER_LIST;
+            }
+            bool reported = data.readBool();
+            unsigned int numSurroundFormats = numSurroundFormatsReq;
+            audio_format_t *surroundFormats = (audio_format_t *)calloc(
+                    numSurroundFormats, sizeof(audio_format_t));
+            bool *surroundFormatsEnabled = (bool *)calloc(numSurroundFormats, sizeof(bool));
+            if (numSurroundFormatsReq > 0 &&
+                    (surroundFormats == NULL || surroundFormatsEnabled == NULL)) {
+                free(surroundFormats);
+                free(surroundFormatsEnabled);
+                reply->writeInt32(NO_MEMORY);
+                return NO_ERROR;
+            }
+            status_t status = getSurroundFormats(
+                    &numSurroundFormats, surroundFormats, surroundFormatsEnabled, reported);
+            reply->writeInt32(status);
+
+            if (status == NO_ERROR) {
+                reply->writeUint32(numSurroundFormats);
+                if (numSurroundFormatsReq > numSurroundFormats) {
+                    numSurroundFormatsReq = numSurroundFormats;
+                }
+                reply->write(surroundFormats, numSurroundFormatsReq * sizeof(audio_format_t));
+                reply->write(surroundFormatsEnabled, numSurroundFormatsReq * sizeof(bool));
+            }
+            free(surroundFormats);
+            free(surroundFormatsEnabled);
+            return NO_ERROR;
+        }
+
+        case SET_SURROUND_FORMAT_ENABLED: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            audio_format_t audioFormat = (audio_format_t) data.readInt32();
+            bool enabled = data.readBool();
+            status_t status = setSurroundFormatEnabled(audioFormat, enabled);
+            reply->writeInt32(status);
+            return NO_ERROR;
+        }
+
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }

diff --git a/media/libaudioclient/IAudioRecord.cpp b/media/libaudioclient/IAudioRecord.cpp
deleted file mode 100644
index 1331c0d..0000000
--- a/media/libaudioclient/IAudioRecord.cpp
+++ /dev/null

@@ -1,94 +0,0 @@
-/*
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#define LOG_TAG "IAudioRecord"
-//#define LOG_NDEBUG 0
-#include <utils/Log.h>
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <binder/Parcel.h>
-
-#include <media/IAudioRecord.h>
-
-namespace android {
-
-enum {
-    UNUSED_WAS_GET_CBLK = IBinder::FIRST_CALL_TRANSACTION,
-    START,
-    STOP
-};
-
-class BpAudioRecord : public BpInterface<IAudioRecord>
-{
-public:
-    explicit BpAudioRecord(const sp<IBinder>& impl)
-        : BpInterface<IAudioRecord>(impl)
-    {
-    }
-
-    virtual status_t start(int /*AudioSystem::sync_event_t*/ event, audio_session_t triggerSession)
-    {
-        Parcel data, reply;
-        data.writeInterfaceToken(IAudioRecord::getInterfaceDescriptor());
-        data.writeInt32(event);
-        data.writeInt32(triggerSession);
-        status_t status = remote()->transact(START, data, &reply);
-        if (status == NO_ERROR) {
-            status = reply.readInt32();
-        } else {
-            ALOGW("start() error: %s", strerror(-status));
-        }
-        return status;
-    }
-
-    virtual void stop()
-    {
-        Parcel data, reply;
-        data.writeInterfaceToken(IAudioRecord::getInterfaceDescriptor());
-        remote()->transact(STOP, data, &reply);
-    }
-
-};
-
-IMPLEMENT_META_INTERFACE(AudioRecord, "android.media.IAudioRecord");
-
-// ----------------------------------------------------------------------
-
-status_t BnAudioRecord::onTransact(
-    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
-    switch (code) {
-        case START: {
-            CHECK_INTERFACE(IAudioRecord, data, reply);
-            int /*AudioSystem::sync_event_t*/ event = data.readInt32();
-            audio_session_t triggerSession = (audio_session_t) data.readInt32();
-            reply->writeInt32(start(event, triggerSession));
-            return NO_ERROR;
-        } break;
-        case STOP: {
-            CHECK_INTERFACE(IAudioRecord, data, reply);
-            stop();
-            return NO_ERROR;
-        } break;
-        default:
-            return BBinder::onTransact(code, data, reply, flags);
-    }
-}
-
-} // namespace android

diff --git a/media/libaudioclient/IAudioTrack.cpp b/media/libaudioclient/IAudioTrack.cpp
index 79e864d..adff057 100644
--- a/media/libaudioclient/IAudioTrack.cpp
+++ b/media/libaudioclient/IAudioTrack.cpp

@@ -28,6 +28,8 @@
 
 namespace android {
 
+using media::VolumeShaper;
+
 enum {
     GET_CBLK = IBinder::FIRST_CALL_TRANSACTION,
     START,
@@ -185,7 +187,7 @@
             return nullptr;
         }
         sp<VolumeShaper::State> state = new VolumeShaper::State;
-        status = state->readFromParcel(reply);
+        status = state->readFromParcel(&reply);
         if (status != NO_ERROR) {
             return nullptr;
         }
@@ -263,12 +265,12 @@
             status_t status = data.readInt32(&present);
             if (status == NO_ERROR && present != 0) {
                 configuration = new VolumeShaper::Configuration();
-                status = configuration->readFromParcel(data);
+                status = configuration->readFromParcel(&data);
             }
             status = status ?: data.readInt32(&present);
             if (status == NO_ERROR && present != 0) {
                 operation = new VolumeShaper::Operation();
-                status = operation->readFromParcel(data);
+                status = operation->readFromParcel(&data);
             }
             if (status == NO_ERROR) {
                 status = (status_t)applyVolumeShaper(configuration, operation);

diff --git a/media/libaudioclient/PlayerBase.cpp b/media/libaudioclient/PlayerBase.cpp
index 7868318..b0c68e5 100644
--- a/media/libaudioclient/PlayerBase.cpp
+++ b/media/libaudioclient/PlayerBase.cpp

@@ -22,6 +22,8 @@
 
 namespace android {
 
+using media::VolumeShaper;
+
 //--------------------------------------------------------------------------------------------------
 PlayerBase::PlayerBase() : BnPlayer(),
         mPanMultiplierL(1.0f), mPanMultiplierR(1.0f),
@@ -117,23 +119,26 @@
 
 //------------------------------------------------------------------------------
 // Implementation of IPlayer
-void PlayerBase::start() {
+binder::Status PlayerBase::start() {
     ALOGD("PlayerBase::start() from IPlayer");
     (void)startWithStatus();
+    return binder::Status::ok();
 }
 
-void PlayerBase::pause() {
+binder::Status PlayerBase::pause() {
     ALOGD("PlayerBase::pause() from IPlayer");
     (void)pauseWithStatus();
+    return binder::Status::ok();
 }
 
 
-void PlayerBase::stop() {
+binder::Status PlayerBase::stop() {
     ALOGD("PlayerBase::stop() from IPlayer");
     (void)stopWithStatus();
+    return binder::Status::ok();
 }
 
-void PlayerBase::setVolume(float vol) {
+binder::Status PlayerBase::setVolume(float vol) {
     ALOGD("PlayerBase::setVolume() from IPlayer");
     {
         Mutex::Autolock _l(mSettingsLock);
@@ -144,9 +149,10 @@
     if (status != NO_ERROR) {
         ALOGW("PlayerBase::setVolume() error %d", status);
     }
+    return binder::Status::fromStatusT(status);
 }
 
-void PlayerBase::setPan(float pan) {
+binder::Status PlayerBase::setPan(float pan) {
     ALOGD("PlayerBase::setPan() from IPlayer");
     {
         Mutex::Autolock _l(mSettingsLock);
@@ -163,22 +169,19 @@
     if (status != NO_ERROR) {
         ALOGW("PlayerBase::setPan() error %d", status);
     }
+    return binder::Status::fromStatusT(status);
 }
 
-void PlayerBase::setStartDelayMs(int32_t delayMs __unused) {
+binder::Status PlayerBase::setStartDelayMs(int32_t delayMs __unused) {
     ALOGW("setStartDelay() is not supported");
+    return binder::Status::ok();
 }
 
-void PlayerBase::applyVolumeShaper(
-        const sp<VolumeShaper::Configuration>& configuration  __unused,
-        const sp<VolumeShaper::Operation>& operation __unused) {
+binder::Status PlayerBase::applyVolumeShaper(
+            const VolumeShaper::Configuration& configuration __unused,
+            const VolumeShaper::Operation& operation __unused) {
     ALOGW("applyVolumeShaper() is not supported");
-}
-
-status_t PlayerBase::onTransact(
-    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
-    return BnPlayer::onTransact(code, data, reply, flags);
+    return binder::Status::ok();
 }
 
 } // namespace android

diff --git a/media/libaudioclient/ToneGenerator.cpp b/media/libaudioclient/ToneGenerator.cpp
index 9bc2594..d846d79 100644
--- a/media/libaudioclient/ToneGenerator.cpp
+++ b/media/libaudioclient/ToneGenerator.cpp

@@ -20,6 +20,7 @@
 #include <math.h>
 #include <utils/Log.h>
 #include <cutils/properties.h>
+#include <media/AudioPolicyHelper.h>
 #include "media/ToneGenerator.h"
 
 
@@ -740,6 +741,18 @@
                         { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
           .repeatCnt = ToneGenerator::TONEGEN_INF,
           .repeatSegment = 0 },                              // TONE_JAPAN_RADIO_ACK
+        { .segments = { { .duration = 375, .waveFreq = { 400, 0 }, 0, 0 },
+                        { .duration = 375, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_GB_BUSY
+        { .segments = { { .duration = 400, .waveFreq = { 400, 0 }, 0, 0 },
+                        { .duration = 350, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 225, .waveFreq = { 400, 0 }, 0, 0 },
+                        { .duration = 525, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_GB_CONGESTION
         { .segments = { { .duration = 400, .waveFreq = { 400, 450, 0 }, 0, 0 },
                         { .duration = 200, .waveFreq = { 0 }, 0, 0 },
                         { .duration = 400, .waveFreq = { 400, 450, 0 }, 0, 0 },
@@ -747,6 +760,10 @@
                         { .duration = 0, .waveFreq = { 0 }, 0, 0}},
           .repeatCnt = ToneGenerator::TONEGEN_INF,
           .repeatSegment = 0 },                              // TONE_GB_RINGTONE
+        { .segments = { { .duration = ToneGenerator::TONEGEN_INF, .waveFreq = { 400, 425, 450, 0 }, 0, 0 },
+                        { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_AUSTRALIA_DIAL
         { .segments = { { .duration = 400, .waveFreq = { 400, 450, 0 }, 0, 0 },
                         { .duration = 200, .waveFreq = { 0 }, 0, 0 },
                         { .duration = 400, .waveFreq = { 400, 450, 0 }, 0, 0 },
@@ -771,6 +788,72 @@
                         { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
           .repeatCnt = ToneGenerator::TONEGEN_INF,
           .repeatSegment = 0 },                              // TONE_AUSTRALIA_CONGESTION
+        { .segments = { { .duration = 750, .waveFreq = { 425, 0 }, 0, 0 },
+                        { .duration = 750, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_SG_BUSY
+        { .segments = { { .duration = 400, .waveFreq = { 401, 425, 449, 0 }, 0, 0 },
+                        { .duration = 200, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 400, .waveFreq = { 401, 425, 449, 0 }, 0, 0 },
+                        { .duration = 2000, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_SG_RINGTONE
+        { .segments = { { .duration = 500, .waveFreq = { 480, 620, 0 }, 0, 0 },
+                        { .duration = 500, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_HK_BUSY
+        { .segments = { { .duration = 400, .waveFreq = { 440, 480, 0 }, 0, 0 },
+                        { .duration = 200, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 400, .waveFreq = { 440, 480, 0 }, 0, 0 },
+                        { .duration = 3000, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_HK_RINGTONE
+        { .segments = { { .duration = 400, .waveFreq = { 400, 450, 0 }, 0, 0 },
+                        { .duration = 200, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 400, .waveFreq = { 400, 450, 0 }, 0, 0 },
+                        { .duration = 2000, .waveFreq = { 0 }, 0, 0},
+                        { .duration = 0, .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_IE_RINGTONE
+        { .segments = { { .duration = 180, .waveFreq = { 425, 0 }, 0, 0 },
+                        { .duration = 200, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 200, .waveFreq = { 425, 0 }, 0, 0 },
+                        { .duration = 4500, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_IE_CALL_WAITING
+        { .segments = { { .duration = ToneGenerator::TONEGEN_INF, .waveFreq = { 375, 400, 425, 0 }, 0, 0 },
+                        { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_INDIA_DIAL
+        { .segments = { { .duration = 750, .waveFreq = { 400, 0 }, 0, 0 },
+                      { .duration = 750, .waveFreq = { 0 }, 0, 0 },
+                      { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                               // TONE_INDIA_BUSY
+        { .segments = { { .duration = 250, .waveFreq = { 400, 0 }, 0, 0 },
+                        { .duration = 250, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_INDIA_CONGESTION
+        { .segments = { { .duration = 200, .waveFreq = { 400, 0 }, 0, 0 },
+                        { .duration = 100, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 200, .waveFreq = { 400, 0 }, 0, 0 },
+                        { .duration = 7500, .waveFreq = { 0 }, 0, 0 },
+                        { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_INDIA_CALL_WAITING
+        { .segments = { { .duration = 400, .waveFreq = { 375, 400, 425, 0 }, 0, 0 },
+                      { .duration = 200, .waveFreq = { 0 }, 0, 0 },
+                      { .duration = 400, .waveFreq = { 375, 400, 425, 0 }, 0, 0 },
+                      { .duration = 2000, .waveFreq = { 0 }, 0, 0 },
+                      { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                               // TONE_INDIA_RINGTONE
 };
 
 // Used by ToneGenerator::getToneForRegion() to convert user specified supervisory tone type
@@ -797,9 +880,9 @@
             TONE_SUP_RINGTONE            // TONE_SUP_RINGTONE
         },
         {   // GB
-            TONE_SUP_DIAL,               // TONE_SUP_DIAL
-            TONE_SUP_BUSY,               // TONE_SUP_BUSY
-            TONE_SUP_CONGESTION,         // TONE_SUP_CONGESTION
+            TONE_ANSI_DIAL,              // TONE_SUP_DIAL
+            TONE_GB_BUSY,                // TONE_SUP_BUSY
+            TONE_GB_CONGESTION,          // TONE_SUP_CONGESTION
             TONE_SUP_RADIO_ACK,          // TONE_SUP_RADIO_ACK
             TONE_SUP_RADIO_NOTAVAIL,     // TONE_SUP_RADIO_NOTAVAIL
             TONE_SUP_ERROR,              // TONE_SUP_ERROR
@@ -807,7 +890,7 @@
             TONE_GB_RINGTONE             // TONE_SUP_RINGTONE
         },
         {   // AUSTRALIA
-            TONE_ANSI_DIAL,             // TONE_SUP_DIAL
+            TONE_AUSTRALIA_DIAL,        // TONE_SUP_DIAL
             TONE_AUSTRALIA_BUSY,        // TONE_SUP_BUSY
             TONE_AUSTRALIA_CONGESTION,  // TONE_SUP_CONGESTION
             TONE_SUP_RADIO_ACK,         // TONE_SUP_RADIO_ACK
@@ -815,6 +898,46 @@
             TONE_SUP_ERROR,             // TONE_SUP_ERROR
             TONE_AUSTRALIA_CALL_WAITING,// TONE_SUP_CALL_WAITING
             TONE_AUSTRALIA_RINGTONE     // TONE_SUP_RINGTONE
+        },
+        {   // SINGAPORE
+            TONE_SUP_DIAL,               // TONE_SUP_DIAL
+            TONE_SG_BUSY,                // TONE_SUP_BUSY
+            TONE_SUP_CONGESTION,         // TONE_SUP_CONGESTION
+            TONE_SUP_RADIO_ACK,          // TONE_SUP_RADIO_ACK
+            TONE_SUP_RADIO_NOTAVAIL,     // TONE_SUP_RADIO_NOTAVAIL
+            TONE_SUP_ERROR,              // TONE_SUP_ERROR
+            TONE_SUP_CALL_WAITING,       // TONE_SUP_CALL_WAITING
+            TONE_SG_RINGTONE             // TONE_SUP_RINGTONE
+        },
+        {   // HONGKONG
+            TONE_SUP_DIAL,               // TONE_SUP_DIAL
+            TONE_HK_BUSY,                // TONE_SUP_BUSY
+            TONE_SUP_CONGESTION,         // TONE_SUP_CONGESTION
+            TONE_SUP_RADIO_ACK,          // TONE_SUP_RADIO_ACK
+            TONE_SUP_RADIO_NOTAVAIL,     // TONE_SUP_RADIO_NOTAVAIL
+            TONE_SUP_ERROR,              // TONE_SUP_ERROR
+            TONE_SUP_CALL_WAITING,       // TONE_SUP_CALL_WAITING
+            TONE_HK_RINGTONE             // TONE_SUP_RINGTONE
+        },
+        {   // IRELAND
+            TONE_SUP_DIAL,               // TONE_SUP_DIAL
+            TONE_SUP_BUSY,               // TONE_SUP_BUSY
+            TONE_SUP_CONGESTION,         // TONE_SUP_CONGESTION
+            TONE_SUP_RADIO_ACK,          // TONE_SUP_RADIO_ACK
+            TONE_SUP_RADIO_NOTAVAIL,     // TONE_SUP_RADIO_NOTAVAIL
+            TONE_SUP_ERROR,              // TONE_SUP_ERROR
+            TONE_IE_CALL_WAITING,        // TONE_SUP_CALL_WAITING
+            TONE_IE_RINGTONE             // TONE_SUP_RINGTONE
+        },
+        {   // INDIA
+            TONE_INDIA_DIAL,             // TONE_SUP_DIAL
+            TONE_INDIA_BUSY,             // TONE_SUP_BUSY
+            TONE_INDIA_CONGESTION,       // TONE_SUP_CONGESTION
+            TONE_SUP_RADIO_ACK,          // TONE_SUP_RADIO_ACK
+            TONE_SUP_RADIO_NOTAVAIL,     // TONE_SUP_RADIO_NOTAVAIL
+            TONE_SUP_ERROR,              // TONE_SUP_ERROR
+            TONE_INDIA_CALL_WAITING,     // TONE_SUP_CALL_WAITING
+            TONE_INDIA_RINGTONE          // TONE_SUP_RINGTONE
         }
 };
 
@@ -864,6 +987,13 @@
     if (property_get("gsm.operator.iso-country", value, "") == 0) {
         property_get("gsm.sim.operator.iso-country", value, "");
     }
+    // If dual sim device has two SIM cards inserted and is not registerd to any network,
+    // "," is set to "gsm.operator.iso-country" prop.
+    // In this case, "gsm.sim.operator.iso-country" prop should be used.
+    if (strlen(value) == 1 && strstr(value, ",") != NULL) {
+        property_get("gsm.sim.operator.iso-country", value, "");
+    }
+
     if (strstr(value, "us") != NULL ||
         strstr(value, "ca") != NULL) {
         mRegion = ANSI;
@@ -873,6 +1003,14 @@
         mRegion = GB;
     } else if (strstr(value, "au") != NULL) {
         mRegion = AUSTRALIA;
+    } else if (strstr(value, "sg") != NULL) {
+        mRegion = SINGAPORE;
+    } else if (strstr(value, "hk") != NULL) {
+        mRegion = HONGKONG;
+    } else if (strstr(value, "ie") != NULL) {
+        mRegion = IRELAND;
+    } else if (strstr(value, "in") != NULL) {
+        mRegion = INDIA;
     } else {
         mRegion = CEPT;
     }
@@ -932,7 +1070,7 @@
     bool lResult = false;
     status_t lStatus;
 
-    if ((toneType < 0) || (toneType >= NUM_TONES))
+    if (toneType >= NUM_TONES)
         return lResult;
 
     toneType = getToneForRegion(toneType);
@@ -947,7 +1085,7 @@
         }
     }
 
-    ALOGV("startTone");
+    ALOGV("startTone toneType %d", toneType);
 
     mLock.lock();
 
@@ -1099,9 +1237,16 @@
     mpAudioTrack = new AudioTrack();
     ALOGV("AudioTrack(%p) created", mpAudioTrack.get());
 
+    audio_attributes_t attr;
+    audio_stream_type_t streamType = mStreamType;
+    if (mStreamType == AUDIO_STREAM_VOICE_CALL) {
+        streamType = AUDIO_STREAM_DTMF;
+    }
+    stream_type_to_audio_attributes(streamType, &attr);
+
     const size_t frameCount = mProcessSize;
     status_t status = mpAudioTrack->set(
-            mStreamType,
+            AUDIO_STREAM_DEFAULT,
             0,    // sampleRate
             AUDIO_FORMAT_PCM_16_BIT,
             AUDIO_CHANNEL_OUT_MONO,
@@ -1113,7 +1258,11 @@
             0,    // sharedBuffer
             mThreadCanCallJava,
             AUDIO_SESSION_ALLOCATE,
-            AudioTrack::TRANSFER_CALLBACK);
+            AudioTrack::TRANSFER_CALLBACK,
+            nullptr,
+            AUDIO_UID_INVALID,
+            -1,
+            &attr);
 
     if (status != NO_ERROR) {
         ALOGE("AudioTrack(%p) set failed with error %d", mpAudioTrack.get(), status);

diff --git a/media/libaudioclient/TrackPlayerBase.cpp b/media/libaudioclient/TrackPlayerBase.cpp
index 48cd803..0a914fc 100644
--- a/media/libaudioclient/TrackPlayerBase.cpp
+++ b/media/libaudioclient/TrackPlayerBase.cpp

@@ -18,6 +18,8 @@
 
 namespace android {
 
+using media::VolumeShaper;
+
 //--------------------------------------------------------------------------------------------------
 TrackPlayerBase::TrackPlayerBase() : PlayerBase(),
         mPlayerVolumeL(1.0f), mPlayerVolumeR(1.0f)
@@ -103,18 +105,24 @@
 }
 
 
-void TrackPlayerBase::applyVolumeShaper(
-        const sp<VolumeShaper::Configuration>& configuration,
-        const sp<VolumeShaper::Operation>& operation) {
+binder::Status TrackPlayerBase::applyVolumeShaper(
+        const VolumeShaper::Configuration& configuration,
+        const VolumeShaper::Operation& operation) {
+
+    sp<VolumeShaper::Configuration> spConfiguration = new VolumeShaper::Configuration(configuration);
+    sp<VolumeShaper::Operation> spOperation = new VolumeShaper::Operation(operation);
+
     if (mAudioTrack != 0) {
         ALOGD("TrackPlayerBase::applyVolumeShaper() from IPlayer");
-        VolumeShaper::Status status = mAudioTrack->applyVolumeShaper(configuration, operation);
+        VolumeShaper::Status status = mAudioTrack->applyVolumeShaper(spConfiguration, spOperation);
         if (status < 0) { // a non-negative value is the volume shaper id.
             ALOGE("TrackPlayerBase::applyVolumeShaper() failed with status %d", status);
         }
+        return binder::Status::fromStatusT(status);
     } else {
         ALOGD("TrackPlayerBase::applyVolumeShaper()"
-                " no AudioTrack for volume control from IPlayer");
+              " no AudioTrack for volume control from IPlayer");
+        return binder::Status::ok();
     }
 }
 

diff --git a/media/libaudioclient/aidl/android/media/IAudioRecord.aidl b/media/libaudioclient/aidl/android/media/IAudioRecord.aidl
new file mode 100644
index 0000000..01e0a71
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/IAudioRecord.aidl

@@ -0,0 +1,39 @@
+/*
+ * Copyright 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+import android.media.MicrophoneInfo;
+
+/* Native code must specify namespace media (media::IAudioRecord) when referring to this class */
+interface IAudioRecord {
+
+  /* After it's created the track is not active. Call start() to
+   * make it active.
+   */
+  void start(int /*AudioSystem::sync_event_t*/ event,
+             int /*audio_session_t*/ triggerSession);
+
+  /* Stop a track. If set, the callback will cease being called and
+   * obtainBuffer will return an error. Buffers that are already released
+   * will be processed, unless flush() is called.
+   */
+  void stop();
+
+  /* Get a list of current active microphones.
+   */
+  void getActiveMicrophones(out MicrophoneInfo[] activeMicrophones);
+}

diff --git a/media/libaudioclient/aidl/android/media/IPlayer.aidl b/media/libaudioclient/aidl/android/media/IPlayer.aidl
new file mode 100644
index 0000000..a90fcdd
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/IPlayer.aidl

@@ -0,0 +1,34 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+import android.media.VolumeShaper.Configuration;
+import android.media.VolumeShaper.Operation;
+
+/**
+ * @hide
+ */
+interface IPlayer {
+    oneway void start();
+    oneway void pause();
+    oneway void stop();
+    oneway void setVolume(float vol);
+    oneway void setPan(float pan);
+    oneway void setStartDelayMs(int delayMs);
+    oneway void applyVolumeShaper(in Configuration configuration,
+                                  in Operation operation);
+}

diff --git a/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl b/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl
new file mode 100644
index 0000000..d6e46cb
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/MicrophoneInfo.aidl

@@ -0,0 +1,19 @@
+/*
+ * Copyright 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+parcelable MicrophoneInfo cpp_header "media/MicrophoneInfo.h";

diff --git a/media/libaudioclient/aidl/android/media/VolumeShaper/Configuration.aidl b/media/libaudioclient/aidl/android/media/VolumeShaper/Configuration.aidl
new file mode 100644
index 0000000..fd0e60f
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/VolumeShaper/Configuration.aidl

@@ -0,0 +1,19 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media.VolumeShaper;
+
+parcelable Configuration cpp_header "media/VolumeShaper.h";

diff --git a/media/libaudioclient/aidl/android/media/VolumeShaper/Operation.aidl b/media/libaudioclient/aidl/android/media/VolumeShaper/Operation.aidl
new file mode 100644
index 0000000..4290d9d
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/VolumeShaper/Operation.aidl

@@ -0,0 +1,19 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media.VolumeShaper;
+
+parcelable Operation cpp_header "media/VolumeShaper.h";

diff --git a/media/libaudioclient/aidl/android/media/VolumeShaper/State.aidl b/media/libaudioclient/aidl/android/media/VolumeShaper/State.aidl
new file mode 100644
index 0000000..f6a22b8
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/VolumeShaper/State.aidl

@@ -0,0 +1,19 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media.VolumeShaper;
+
+parcelable State cpp_header "media/VolumeShaper.h";

diff --git a/media/libaudioclient/include/media/AudioClient.h b/media/libaudioclient/include/media/AudioClient.h
index 9efd76d..247af9e 100644
--- a/media/libaudioclient/include/media/AudioClient.h
+++ b/media/libaudioclient/include/media/AudioClient.h

@@ -18,19 +18,38 @@
 #ifndef ANDROID_AUDIO_CLIENT_H
 #define ANDROID_AUDIO_CLIENT_H
 
+#include <binder/Parcel.h>
+#include <binder/Parcelable.h>
 #include <system/audio.h>
 #include <utils/String16.h>
 
 namespace android {
 
-class AudioClient {
+class AudioClient : public Parcelable {
  public:
     AudioClient() :
-        clientUid(-1), clientPid(-1), packageName("") {}
+        clientUid(-1), clientPid(-1), clientTid(-1), packageName("") {}
 
     uid_t clientUid;
     pid_t clientPid;
+    pid_t clientTid;
     String16 packageName;
+
+    status_t readFromParcel(const Parcel *parcel) override {
+        clientUid = parcel->readInt32();
+        clientPid = parcel->readInt32();
+        clientTid = parcel->readInt32();
+        packageName = parcel->readString16();
+        return NO_ERROR;
+    }
+
+    status_t writeToParcel(Parcel *parcel) const override {
+        parcel->writeInt32(clientUid);
+        parcel->writeInt32(clientPid);
+        parcel->writeInt32(clientTid);
+        parcel->writeString16(packageName);
+        return NO_ERROR;
+    }
 };
 
 }; // namespace android

diff --git a/media/libaudioclient/include/media/AudioMixer.h b/media/libaudioclient/include/media/AudioMixer.h
index 2bd2d01..cf7d90f 100644
--- a/media/libaudioclient/include/media/AudioMixer.h
+++ b/media/libaudioclient/include/media/AudioMixer.h

@@ -18,14 +18,17 @@
 #ifndef ANDROID_AUDIO_MIXER_H
 #define ANDROID_AUDIO_MIXER_H
 
+#include <pthread.h>
+#include <sstream>
 #include <stdint.h>
 #include <sys/types.h>
+#include <unordered_map>
 
 #include <media/AudioBufferProvider.h>
 #include <media/AudioResampler.h>
 #include <media/AudioResamplerPublic.h>
 #include <media/BufferProviders.h>
-#include <media/nbaio/NBLog.h>
+#include <media/nblog/NBLog.h>
 #include <system/audio.h>
 #include <utils/Compat.h>
 #include <utils/threads.h>
@@ -33,6 +36,9 @@
 // FIXME This is actually unity gain, which might not be max in future, expressed in U.12
 #define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
 
+// This must match frameworks/av/services/audioflinger/Configuration.h
+#define FLOAT_AUX
+
 namespace android {
 
 // ----------------------------------------------------------------------------
@@ -40,20 +46,10 @@
 class AudioMixer
 {
 public:
-                            AudioMixer(size_t frameCount, uint32_t sampleRate,
-                                       uint32_t maxNumTracks = MAX_NUM_TRACKS);
-
-    /*virtual*/             ~AudioMixer();  // non-virtual saves a v-table, restore if sub-classed
-
-
-    // This mixer has a hard-coded upper limit of 32 active track inputs.
-    // Adding support for > 32 tracks would require more than simply changing this value.
-    static const uint32_t MAX_NUM_TRACKS = 32;
-    // maximum number of channels supported by the mixer
-
+    // Do not change these unless underlying code changes.
     // This mixer has a hard-coded upper limit of 8 channels for output.
-    static const uint32_t MAX_NUM_CHANNELS = 8;
-    static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only
+    static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8;
+    static constexpr uint32_t MAX_NUM_VOLUMES = FCC_2; // stereo volume only
     // maximum number of channels supported for the content
     static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
 
@@ -61,12 +57,6 @@
     static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
 
     enum { // names
-
-        // track names (MAX_NUM_TRACKS units)
-        TRACK0          = 0x1000,
-
-        // 0x2000 is unused
-
         // setParameter targets
         TRACK           = 0x3000,
         RESAMPLE        = 0x3001,
@@ -105,17 +95,33 @@
                                   // parameter 'value' is a pointer to the new playback rate.
     };
 
+    AudioMixer(size_t frameCount, uint32_t sampleRate)
+        : mSampleRate(sampleRate)
+        , mFrameCount(frameCount) {
+        pthread_once(&sOnceControl, &sInitRoutine);
+    }
 
-    // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
+    // Create a new track in the mixer.
+    //
+    // \param name        a unique user-provided integer associated with the track.
+    //                    If name already exists, the function will abort.
+    // \param channelMask output channel mask.
+    // \param format      PCM format
+    // \param sessionId   Session id for the track. Tracks with the same
+    //                    session id will be submixed together.
+    //
+    // \return OK        on success.
+    //         BAD_VALUE if the format does not satisfy isValidFormat()
+    //                   or the channelMask does not satisfy isValidChannelMask().
+    status_t    create(
+            int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId);
 
-    // Allocate a track name.  Returns new track name if successful, -1 on failure.
-    // The failure could be because of an invalid channelMask or format, or that
-    // the track capacity of the mixer is exceeded.
-    int         getTrackName(audio_channel_mask_t channelMask,
-                             audio_format_t format, int sessionId);
+    bool        exists(int name) const {
+        return mTracks.count(name) > 0;
+    }
 
-    // Free an allocated track by name
-    void        deleteTrackName(int name);
+    // Free an allocated track by name.
+    void        destroy(int name);
 
     // Enable or disable an allocated track by name
     void        enable(int name);
@@ -124,13 +130,26 @@
     void        setParameter(int name, int target, int param, void *value);
 
     void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
-    void        process();
 
-    uint32_t    trackNames() const { return mTrackNames; }
+    void        process() {
+        (this->*mHook)();
+    }
 
     size_t      getUnreleasedFrames(int name) const;
 
-    static inline bool isValidPcmTrackFormat(audio_format_t format) {
+    std::string trackNames() const {
+        std::stringstream ss;
+        for (const auto &pair : mTracks) {
+            ss << pair.first << " ";
+        }
+        return ss.str();
+    }
+
+    void        setNBLogWriter(NBLog::Writer *logWriter) {
+        mNBLogWriter = logWriter;
+    }
+
+    static inline bool isValidFormat(audio_format_t format) {
         switch (format) {
         case AUDIO_FORMAT_PCM_8_BIT:
         case AUDIO_FORMAT_PCM_16_BIT:
@@ -143,8 +162,23 @@
         }
     }
 
+    static inline bool isValidChannelMask(audio_channel_mask_t channelMask) {
+        return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
+    }
+
 private:
 
+    /* For multi-format functions (calls template functions
+     * in AudioMixerOps.h).  The template parameters are as follows:
+     *
+     *   MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+     *   USEFLOATVOL (set to true if float volume is used)
+     *   ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
+     *   TO: int32_t (Q4.27) or float
+     *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+     *   TA: int32_t (Q4.27)
+     */
+
     enum {
         // FIXME this representation permits up to 8 channels
         NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
@@ -161,14 +195,67 @@
         NEEDS_AUX                   = 0x00010000,
     };
 
-    struct state_t;
-    struct track_t;
+    // hook types
+    enum {
+        PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere
+    };
 
-    typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp,
-                           int32_t* aux);
-    static const int BLOCKSIZE = 16; // 4 cache lines
+    enum {
+        TRACKTYPE_NOP,
+        TRACKTYPE_RESAMPLE,
+        TRACKTYPE_NORESAMPLE,
+        TRACKTYPE_NORESAMPLEMONO,
+    };
 
-    struct track_t {
+    // process hook functionality
+    using process_hook_t = void(AudioMixer::*)();
+
+    struct Track;
+    using hook_t = void(Track::*)(int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
+
+    struct Track {
+        Track()
+            : bufferProvider(nullptr)
+        {
+            // TODO: move additional initialization here.
+        }
+
+        ~Track()
+        {
+            // bufferProvider, mInputBufferProvider need not be deleted.
+            mResampler.reset(nullptr);
+            // Ensure the order of destruction of buffer providers as they
+            // release the upstream provider in the destructor.
+            mTimestretchBufferProvider.reset(nullptr);
+            mPostDownmixReformatBufferProvider.reset(nullptr);
+            mDownmixerBufferProvider.reset(nullptr);
+            mReformatBufferProvider.reset(nullptr);
+        }
+
+        bool        needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
+        bool        setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
+        bool        doesResample() const { return mResampler.get() != nullptr; }
+        void        resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); }
+        void        adjustVolumeRamp(bool aux, bool useFloat = false);
+        size_t      getUnreleasedFrames() const { return mResampler.get() != nullptr ?
+                                                    mResampler->getUnreleasedFrames() : 0; };
+
+        status_t    prepareForDownmix();
+        void        unprepareForDownmix();
+        status_t    prepareForReformat();
+        void        unprepareForReformat();
+        bool        setPlaybackRate(const AudioPlaybackRate &playbackRate);
+        void        reconfigureBufferProviders();
+
+        static hook_t getTrackHook(int trackType, uint32_t channelCount,
+                audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+
+        void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+
+        template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+            typename TO, typename TI, typename TA>
+        void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp);
+
         uint32_t    needs;
 
         // TODO: Eventually remove legacy integer volume settings
@@ -178,16 +265,11 @@
         };
 
         int32_t     prevVolume[MAX_NUM_VOLUMES];
-
-        // 16-byte boundary
-
         int32_t     volumeInc[MAX_NUM_VOLUMES];
         int32_t     auxInc;
         int32_t     prevAuxLevel;
-
-        // 16-byte boundary
-
         int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
+
         uint16_t    frameCount;
 
         uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
@@ -199,22 +281,16 @@
         //  for how the Track buffer provider is wrapped by another one when dowmixing is required
         AudioBufferProvider*                bufferProvider;
 
-        // 16-byte boundary
-
         mutable AudioBufferProvider::Buffer buffer; // 8 bytes
 
         hook_t      hook;
-        const void* in;             // current location in buffer
+        const void  *mIn;             // current location in buffer
 
-        // 16-byte boundary
-
-        AudioResampler*     resampler;
+        std::unique_ptr<AudioResampler> mResampler;
         uint32_t            sampleRate;
         int32_t*           mainBuffer;
         int32_t*           auxBuffer;
 
-        // 16-byte boundary
-
         /* Buffer providers are constructed to translate the track input data as needed.
          *
          * TODO: perhaps make a single PlaybackConverterProvider class to move
@@ -225,17 +301,17 @@
          *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
          *    requires reformat. For example, it may convert floating point input to
          *    PCM_16_bit if that's required by the downmixer.
-         * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match
+         * 3) mDownmixerBufferProvider: If not NULL, performs the channel remixing to match
          *    the number of channels required by the mixer sink.
          * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
          *    the downmixer requirements to the mixer engine input requirements.
          * 5) mTimestretchBufferProvider: Adds timestretching for playback rate
          */
         AudioBufferProvider*     mInputBufferProvider;    // externally provided buffer provider.
-        PassthruBufferProvider*  mReformatBufferProvider; // provider wrapper for reformatting.
-        PassthruBufferProvider*  downmixerBufferProvider; // wrapper for channel conversion.
-        PassthruBufferProvider*  mPostDownmixReformatBufferProvider;
-        PassthruBufferProvider*  mTimestretchBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mReformatBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mDownmixerBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mPostDownmixReformatBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mTimestretchBufferProvider;
 
         int32_t     sessionId;
 
@@ -260,129 +336,74 @@
 
         AudioPlaybackRate    mPlaybackRate;
 
-        bool        needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
-        bool        setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
-        bool        doesResample() const { return resampler != NULL; }
-        void        resetResampler() { if (resampler != NULL) resampler->reset(); }
-        void        adjustVolumeRamp(bool aux, bool useFloat = false);
-        size_t      getUnreleasedFrames() const { return resampler != NULL ?
-                                                    resampler->getUnreleasedFrames() : 0; };
+    private:
+        // hooks
+        void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+        void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+        void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
 
-        status_t    prepareForDownmix();
-        void        unprepareForDownmix();
-        status_t    prepareForReformat();
-        void        unprepareForReformat();
-        bool        setPlaybackRate(const AudioPlaybackRate &playbackRate);
-        void        reconfigureBufferProviders();
+        void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+        void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+
+        // multi-format track hooks
+        template <int MIXTYPE, typename TO, typename TI, typename TA>
+        void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
+        template <int MIXTYPE, typename TO, typename TI, typename TA>
+        void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
     };
 
-    typedef void (*process_hook_t)(state_t* state);
-
-    // pad to 32-bytes to fill cache line
-    struct state_t {
-        uint32_t        enabledTracks;
-        uint32_t        needsChanged;
-        size_t          frameCount;
-        process_hook_t  hook;   // one of process__*, never NULL
-        int32_t         *outputTemp;
-        int32_t         *resampleTemp;
-        NBLog::Writer*  mNBLogWriter;   // associated NBLog::Writer or &mDummyLog
-        int32_t         reserved[1];
-        // FIXME allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
-        track_t         tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
-    };
-
-    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
-    uint32_t        mTrackNames;
-
-    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
-    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
-    const uint32_t  mConfiguredNames;
-
-    const uint32_t  mSampleRate;
-
-    NBLog::Writer   mDummyLogWriter;
-public:
-    // Called by FastMixer to inform AudioMixer of it's associated NBLog::Writer.
-    // FIXME It would be safer to use TLS for this, so we don't accidentally use wrong one.
-    void            setNBLogWriter(NBLog::Writer* log);
-private:
-    state_t         mState __attribute__((aligned(32)));
-
-    // Call after changing either the enabled status of a track, or parameters of an enabled track.
-    // OK to call more often than that, but unnecessary.
-    void invalidateState(uint32_t mask);
+    // TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
+    static constexpr int BLOCKSIZE = 16;
 
     bool setChannelMasks(int name,
             audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
 
-    static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
-            int32_t* aux);
-    static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-    static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
-            int32_t* aux);
-    static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
-            int32_t* aux);
-    static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
-            int32_t* aux);
-    static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
-            int32_t* aux);
+    // Called when track info changes and a new process hook should be determined.
+    void invalidate() {
+        mHook = &AudioMixer::process__validate;
+    }
 
-    static void process__validate(state_t* state);
-    static void process__nop(state_t* state);
-    static void process__genericNoResampling(state_t* state);
-    static void process__genericResampling(state_t* state);
-    static void process__OneTrack16BitsStereoNoResampling(state_t* state);
+    void process__validate();
+    void process__nop();
+    void process__genericNoResampling();
+    void process__genericResampling();
+    void process__oneTrack16BitsStereoNoResampling();
 
-    static pthread_once_t   sOnceControl;
-    static void             sInitRoutine();
-
-    /* multi-format volume mixing function (calls template functions
-     * in AudioMixerOps.h).  The template parameters are as follows:
-     *
-     *   MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
-     *   USEFLOATVOL (set to true if float volume is used)
-     *   ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
-     *   TO: int32_t (Q4.27) or float
-     *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
-     *   TA: int32_t (Q4.27)
-     */
-    template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
-        typename TO, typename TI, typename TA>
-    static void volumeMix(TO *out, size_t outFrames,
-            const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t);
-
-    // multi-format process hooks
     template <int MIXTYPE, typename TO, typename TI, typename TA>
-    static void process_NoResampleOneTrack(state_t* state);
+    void process__noResampleOneTrack();
 
-    // multi-format track hooks
-    template <int MIXTYPE, typename TO, typename TI, typename TA>
-    static void track__Resample(track_t* t, TO* out, size_t frameCount,
-            TO* temp __unused, TA* aux);
-    template <int MIXTYPE, typename TO, typename TI, typename TA>
-    static void track__NoResample(track_t* t, TO* out, size_t frameCount,
-            TO* temp __unused, TA* aux);
+    static process_hook_t getProcessHook(int processType, uint32_t channelCount,
+            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
 
     static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
             void *in, audio_format_t mixerInFormat, size_t sampleCount);
 
-    // hook types
-    enum {
-        PROCESSTYPE_NORESAMPLEONETRACK,
-    };
-    enum {
-        TRACKTYPE_NOP,
-        TRACKTYPE_RESAMPLE,
-        TRACKTYPE_NORESAMPLE,
-        TRACKTYPE_NORESAMPLEMONO,
-    };
+    static void sInitRoutine();
 
-    // functions for determining the proper process and track hooks.
-    static process_hook_t getProcessHook(int processType, uint32_t channelCount,
-            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
-    static hook_t getTrackHook(int trackType, uint32_t channelCount,
-            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+    // initialization constants
+    const uint32_t mSampleRate;
+    const size_t mFrameCount;
+
+    NBLog::Writer *mNBLogWriter = nullptr;   // associated NBLog::Writer
+
+    process_hook_t mHook = &AudioMixer::process__nop;   // one of process__*, never nullptr
+
+    // the size of the type (int32_t) should be the largest of all types supported
+    // by the mixer.
+    std::unique_ptr<int32_t[]> mOutputTemp;
+    std::unique_ptr<int32_t[]> mResampleTemp;
+
+    // track names grouped by main buffer, in no particular order of main buffer.
+    // however names for a particular main buffer are in order (by construction).
+    std::unordered_map<void * /* mainBuffer */, std::vector<int /* name */>> mGroups;
+
+    // track names that are enabled, in increasing order (by construction).
+    std::vector<int /* name */> mEnabled;
+
+    // track smart pointers, by name, in increasing order of name.
+    std::map<int /* name */, std::shared_ptr<Track>> mTracks;
+
+    static pthread_once_t sOnceControl; // initialized in constructor by first new
 };
 
 // ----------------------------------------------------------------------------

diff --git a/media/libaudioclient/include/media/AudioParameter.h b/media/libaudioclient/include/media/AudioParameter.h
index 1ace607..967d895 100644
--- a/media/libaudioclient/include/media/AudioParameter.h
+++ b/media/libaudioclient/include/media/AudioParameter.h

@@ -58,6 +58,12 @@
     static const char * const keyMonoOutput;
     static const char * const keyStreamHwAvSync;
 
+    //  keys for presentation selection
+    //  keyPresentationId: Audio presentation identifier
+    //  keyProgramId: Audio presentation program identifier
+    static const char * const keyPresentationId;
+    static const char * const keyProgramId;
+
     //  keyStreamConnect / Disconnect: value is an int in audio_devices_t
     static const char * const keyStreamConnect;
     static const char * const keyStreamDisconnect;
@@ -75,6 +81,11 @@
 
     static const char * const valueListSeparator;
 
+    // keyReconfigA2dp: Ask HwModule to reconfigure A2DP offloaded codec
+    // keyReconfigA2dpSupported: Query if HwModule supports A2DP offload codec config
+    static const char * const keyReconfigA2dp;
+    static const char * const keyReconfigA2dpSupported;
+
     String8 toString() const { return toStringImpl(true); }
     String8 keysToString() const { return toStringImpl(false); }
 

diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index dd72170..cf446a5 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h

@@ -17,12 +17,18 @@
 #ifndef ANDROID_AUDIORECORD_H
 #define ANDROID_AUDIORECORD_H
 
+#include <binder/IMemory.h>
 #include <cutils/sched_policy.h>
 #include <media/AudioSystem.h>
 #include <media/AudioTimestamp.h>
-#include <media/IAudioRecord.h>
+#include <media/MediaAnalyticsItem.h>
 #include <media/Modulo.h>
+#include <media/MicrophoneInfo.h>
+#include <utils/RefBase.h>
 #include <utils/threads.h>
+#include <vector>
+
+#include "android/media/IAudioRecord.h"
 
 namespace android {
 
@@ -182,7 +188,8 @@
                                     audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
                                     uid_t uid = AUDIO_UID_INVALID,
                                     pid_t pid = -1,
-                                    const audio_attributes_t* pAttributes = NULL);
+                                    const audio_attributes_t* pAttributes = NULL,
+                                    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
 
     /* Terminates the AudioRecord and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioRecord.
@@ -220,7 +227,8 @@
                             audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
                             uid_t uid = AUDIO_UID_INVALID,
                             pid_t pid = -1,
-                            const audio_attributes_t* pAttributes = NULL);
+                            const audio_attributes_t* pAttributes = NULL,
+                            audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
 
     /* Result of constructing the AudioRecord. This must be checked for successful initialization
      * before using any AudioRecord API (except for set()), because using
@@ -250,6 +258,11 @@
      */
             uint32_t    getNotificationPeriodInFrames() const { return mNotificationFramesAct; }
 
+    /*
+     * return metrics information for the current instance.
+     */
+            status_t getMetrics(MediaAnalyticsItem * &item);
+
     /* After it's created the track is not active. Call start() to
      * make it active. If set, the callback will start being called.
      * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until
@@ -516,6 +529,16 @@
     /* Get the flags */
             audio_input_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; }
 
+    /* Get active microphones. A empty vector of MicrophoneInfo will be passed as a parameter,
+     * the data will be filled when querying the hal.
+     */
+            status_t    getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
+
+    /*
+     * Dumps the state of an audio record.
+     */
+            status_t    dump(int fd, const Vector<String16>& args) const;
+
 private:
     /* copying audio record objects is not allowed */
                         AudioRecord(const AudioRecord& other);
@@ -565,7 +588,7 @@
 
             // caller must hold lock on mLock for all _l methods
 
-            status_t openRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName);
+            status_t createRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName);
 
             // FIXME enum is faster than strcmp() for parameter 'from'
             status_t restoreRecord_l(const char *from);
@@ -635,7 +658,7 @@
 
     // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0
     // provided the initial set() was successful
-    sp<IAudioRecord>        mAudioRecord;
+    sp<media::IAudioRecord> mAudioRecord;
     sp<IMemory>             mCblkMemory;
     audio_track_cblk_t*     mCblk;              // re-load after mLock.unlock()
     sp<IMemory>             mBufferMemory;
@@ -677,8 +700,40 @@
                                               // May not match the app selection depending on other
                                               // activity and connected devices
     wp<AudioSystem::AudioDeviceCallback> mDeviceCallback;
-    audio_port_handle_t    mPortId;  // unique ID allocated by audio policy
 
+private:
+    class MediaMetrics {
+      public:
+        MediaMetrics() : mAnalyticsItem(new MediaAnalyticsItem("audiorecord")),
+                         mCreatedNs(systemTime(SYSTEM_TIME_REALTIME)),
+                         mStartedNs(0), mDurationNs(0), mCount(0),
+                         mLastError(NO_ERROR) {
+        }
+        ~MediaMetrics() {
+            // mAnalyticsItem alloc failure will be flagged in the constructor
+            // don't log empty records
+            if (mAnalyticsItem->count() > 0) {
+                mAnalyticsItem->selfrecord();
+            }
+        }
+        void gather(const AudioRecord *record);
+        MediaAnalyticsItem *dup() { return mAnalyticsItem->dup(); }
+
+        void logStart(nsecs_t when) { mStartedNs = when; mCount++; }
+        void logStop(nsecs_t when) { mDurationNs += (when-mStartedNs); mStartedNs = 0;}
+        void markError(status_t errcode, const char *func)
+                 { mLastError = errcode; mLastErrorFunc = func;}
+      private:
+        std::unique_ptr<MediaAnalyticsItem> mAnalyticsItem;
+        nsecs_t mCreatedNs;     // XXX: perhaps not worth it in production
+        nsecs_t mStartedNs;
+        nsecs_t mDurationNs;
+        int32_t mCount;
+
+        status_t mLastError;
+        std::string mLastErrorFunc;
+    };
+    MediaMetrics mMediaMetrics;
 };
 
 }; // namespace android

diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 5a81d83..4c0f796 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h

@@ -23,11 +23,13 @@
 #include <media/AudioIoDescriptor.h>
 #include <media/IAudioFlingerClient.h>
 #include <media/IAudioPolicyServiceClient.h>
+#include <media/MicrophoneInfo.h>
 #include <system/audio.h>
 #include <system/audio_effect.h>
 #include <system/audio_policy.h>
 #include <utils/Errors.h>
 #include <utils/Mutex.h>
+#include <vector>
 
 namespace android {
 
@@ -106,6 +108,9 @@
 
     static float linearToLog(int volume);
     static int logToLinear(float volume);
+    static size_t calculateMinFrameCount(
+            uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate,
+            uint32_t sampleRate, float speed /*, uint32_t notificationsPerBufferReq*/);
 
     // Returned samplingRate and frameCount output values are guaranteed
     // to be non-zero if status == NO_ERROR
@@ -209,18 +214,11 @@
     static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
     static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
 
-    // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
-    // or release it with releaseOutput().
-    static audio_io_handle_t getOutput(audio_stream_type_t stream,
-                                        uint32_t samplingRate = 0,
-                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
-                                        audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
-                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
-                                        const audio_offload_info_t *offloadInfo = NULL);
     static status_t getOutputForAttr(const audio_attributes_t *attr,
                                      audio_io_handle_t *output,
                                      audio_session_t session,
                                      audio_stream_type_t *stream,
+                                     pid_t pid,
                                      uid_t uid,
                                      const audio_config_t *config,
                                      audio_output_flags_t flags,
@@ -236,24 +234,23 @@
                               audio_stream_type_t stream,
                               audio_session_t session);
 
-    // Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
+    // Client must successfully hand off the handle reference to AudioFlinger via createRecord(),
     // or release it with releaseInput().
     static status_t getInputForAttr(const audio_attributes_t *attr,
                                     audio_io_handle_t *input,
                                     audio_session_t session,
                                     pid_t pid,
                                     uid_t uid,
+                                    const String16& opPackageName,
                                     const audio_config_base_t *config,
                                     audio_input_flags_t flags,
                                     audio_port_handle_t *selectedDeviceId,
                                     audio_port_handle_t *portId);
 
-    static status_t startInput(audio_io_handle_t input,
-                               audio_session_t session);
-    static status_t stopInput(audio_io_handle_t input,
-                              audio_session_t session);
-    static void releaseInput(audio_io_handle_t input,
-                             audio_session_t session);
+    static status_t startInput(audio_port_handle_t portId,
+                               bool *silenced);
+    static status_t stopInput(audio_port_handle_t portId);
+    static void releaseInput(audio_port_handle_t portId);
     static status_t initStreamVolume(audio_stream_type_t stream,
                                       int indexMin,
                                       int indexMax);
@@ -286,7 +283,7 @@
     static uint32_t getPrimaryOutputSamplingRate();
     static size_t getPrimaryOutputFrameCount();
 
-    static status_t setLowRamDevice(bool isLowRamDevice);
+    static status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory);
 
     // Check if hw offload is possible for given format, stream type, sample rate,
     // bit rate, duration, video and streaming or offload property is enabled
@@ -341,6 +338,17 @@
     static float    getStreamVolumeDB(
             audio_stream_type_t stream, int index, audio_devices_t device);
 
+    static status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+
+    // numSurroundFormats holds the maximum number of formats and bool value allowed in the array.
+    // When numSurroundFormats is 0, surroundFormats and surroundFormatsEnabled will not be
+    // populated. The actual number of surround formats should be returned at numSurroundFormats.
+    static status_t getSurroundFormats(unsigned int *numSurroundFormats,
+                                       audio_format_t *surroundFormats,
+                                       bool *surroundFormatsEnabled,
+                                       bool reported);
+    static status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled);
+
     // ----------------------------------------------------------------------------
 
     class AudioPortCallback : public RefBase
@@ -432,6 +440,7 @@
 
         int addAudioPortCallback(const sp<AudioPortCallback>& callback);
         int removeAudioPortCallback(const sp<AudioPortCallback>& callback);
+        bool isAudioPortCbEnabled() const { return (mAudioPortCallbacks.size() != 0); }
 
         // DeathRecipient
         virtual void binderDied(const wp<IBinder>& who);
@@ -450,6 +459,7 @@
         Vector <sp <AudioPortCallback> >    mAudioPortCallbacks;
     };
 
+    static audio_io_handle_t getOutput(audio_stream_type_t stream);
     static const sp<AudioFlingerClient> getAudioFlingerClient();
     static sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle);
 

diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index 47d87e9..3eb627d 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h

@@ -22,6 +22,7 @@
 #include <media/AudioTimestamp.h>
 #include <media/IAudioTrack.h>
 #include <media/AudioResamplerPublic.h>
+#include <media/MediaAnalyticsItem.h>
 #include <media/Modulo.h>
 #include <utils/threads.h>
 
@@ -218,6 +219,8 @@
      *                     maxRequiredSpeed playback. Values less than 1.0f and greater than
      *                     AUDIO_TIMESTRETCH_SPEED_MAX will be clamped.  For non-PCM tracks
      *                     and direct or offloaded tracks, this parameter is ignored.
+     * selectedDeviceId:   Selected device id of the app which initially requested the AudioTrack
+     *                     to open with a specific device.
      * threadCanCallJava:  Not present in parameter list, and so is fixed at false.
      */
 
@@ -237,7 +240,8 @@
                                     pid_t pid = -1,
                                     const audio_attributes_t* pAttributes = NULL,
                                     bool doNotReconnect = false,
-                                    float maxRequiredSpeed = 1.0f);
+                                    float maxRequiredSpeed = 1.0f,
+                                    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
 
     /* Creates an audio track and registers it with AudioFlinger.
      * With this constructor, the track is configured for static buffer mode.
@@ -313,7 +317,8 @@
                             pid_t pid = -1,
                             const audio_attributes_t* pAttributes = NULL,
                             bool doNotReconnect = false,
-                            float maxRequiredSpeed = 1.0f);
+                            float maxRequiredSpeed = 1.0f,
+                            audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
 
     /* Result of constructing the AudioTrack. This must be checked for successful initialization
      * before using any AudioTrack API (except for set()), because using
@@ -381,6 +386,11 @@
     /* Return the static buffer specified in constructor or set(), or 0 for streaming mode */
             sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
 
+    /*
+     * return metrics information for the current track.
+     */
+            status_t getMetrics(MediaAnalyticsItem * &item);
+
     /* After it's created the track is not active. Call start() to
      * make it active. If set, the callback will start being called.
      * If the track was previously paused, volume is ramped up over the first mix buffer.
@@ -748,12 +758,15 @@
             status_t    setParameters(const String8& keyValuePairs);
 
     /* Sets the volume shaper object */
-            VolumeShaper::Status applyVolumeShaper(
-                    const sp<VolumeShaper::Configuration>& configuration,
-                    const sp<VolumeShaper::Operation>& operation);
+            media::VolumeShaper::Status applyVolumeShaper(
+                    const sp<media::VolumeShaper::Configuration>& configuration,
+                    const sp<media::VolumeShaper::Operation>& operation);
 
     /* Gets the volume shaper state */
-            sp<VolumeShaper::State> getVolumeShaperState(int id);
+            sp<media::VolumeShaper::State> getVolumeShaperState(int id);
+
+    /* Selects the presentation (if available) */
+            status_t    selectPresentation(int presentationId, int programId);
 
     /* Get parameters */
             String8     getParameters(const String8& keys);
@@ -990,7 +1003,7 @@
     sp<IAudioTrack>         mAudioTrack;
     sp<IMemory>             mCblkMemory;
     audio_track_cblk_t*     mCblk;                  // re-load after mLock.unlock()
-    audio_io_handle_t       mOutput;                // returned by AudioSystem::getOutput()
+    audio_io_handle_t       mOutput;                // returned by AudioSystem::getOutputForAttr()
 
     sp<AudioTrackThread>    mAudioTrackThread;
     bool                    mThreadCanCallJava;
@@ -1160,7 +1173,7 @@
                                               // May not match the app selection depending on other
                                               // activity and connected devices.
 
-    sp<VolumeHandler>       mVolumeHandler;
+    sp<media::VolumeHandler>       mVolumeHandler;
 
 private:
     class DeathNotifier : public IBinder::DeathRecipient {
@@ -1178,7 +1191,25 @@
     pid_t                   mClientPid;
 
     wp<AudioSystem::AudioDeviceCallback> mDeviceCallback;
-    audio_port_handle_t     mPortId;  // unique ID allocated by audio policy
+
+private:
+    class MediaMetrics {
+      public:
+        MediaMetrics() : mAnalyticsItem(new MediaAnalyticsItem("audiotrack")) {
+        }
+        ~MediaMetrics() {
+            // mAnalyticsItem alloc failure will be flagged in the constructor
+            // don't log empty records
+            if (mAnalyticsItem->count() > 0) {
+                mAnalyticsItem->selfrecord();
+            }
+        }
+        void gather(const AudioTrack *track);
+        MediaAnalyticsItem *dup() { return mAnalyticsItem->dup(); }
+      private:
+        std::unique_ptr<MediaAnalyticsItem> mAnalyticsItem;
+    };
+    MediaMetrics mMediaMetrics;
 };
 
 }; // namespace android

diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 0ad4231..e6bf72f 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h

@@ -24,8 +24,10 @@
 #include <utils/RefBase.h>
 #include <utils/Errors.h>
 #include <binder/IInterface.h>
+#include <binder/Parcel.h>
+#include <binder/Parcelable.h>
+#include <media/AudioClient.h>
 #include <media/IAudioTrack.h>
-#include <media/IAudioRecord.h>
 #include <media/IAudioFlingerClient.h>
 #include <system/audio.h>
 #include <system/audio_effect.h>
@@ -33,6 +35,10 @@
 #include <media/IEffect.h>
 #include <media/IEffectClient.h>
 #include <utils/String8.h>
+#include <media/MicrophoneInfo.h>
+#include <vector>
+
+#include "android/media/IAudioRecord.h"
 
 namespace android {
 
@@ -43,6 +49,271 @@
 public:
     DECLARE_META_INTERFACE(AudioFlinger);
 
+    /* CreateTrackInput contains all input arguments sent by AudioTrack to AudioFlinger
+     * when calling createTrack() including arguments that will be updated by AudioFlinger
+     * and returned in CreateTrackOutput object
+     */
+    class CreateTrackInput : public Parcelable {
+    public:
+        status_t readFromParcel(const Parcel *parcel) override {
+            /* input arguments*/
+            memset(&attr, 0, sizeof(audio_attributes_t));
+            if (parcel->read(&attr, sizeof(audio_attributes_t)) != NO_ERROR) {
+                return DEAD_OBJECT;
+            }
+            attr.tags[AUDIO_ATTRIBUTES_TAGS_MAX_SIZE -1] = '\0';
+            memset(&config, 0, sizeof(audio_config_t));
+            if (parcel->read(&config, sizeof(audio_config_t)) != NO_ERROR) {
+                return DEAD_OBJECT;
+            }
+            if (clientInfo.readFromParcel(parcel) != NO_ERROR) {
+                return DEAD_OBJECT;
+            }
+            if (parcel->readInt32() != 0) {
+                sharedBuffer = interface_cast<IMemory>(parcel->readStrongBinder());
+                if (sharedBuffer == 0 || sharedBuffer->pointer() == NULL) {
+                    return BAD_VALUE;
+                }
+            }
+            notificationsPerBuffer = parcel->readInt32();
+            speed = parcel->readFloat();
+
+            /* input/output arguments*/
+            (void)parcel->read(&flags, sizeof(audio_output_flags_t));
+            frameCount = parcel->readInt64();
+            notificationFrameCount = parcel->readInt64();
+            (void)parcel->read(&selectedDeviceId, sizeof(audio_port_handle_t));
+            (void)parcel->read(&sessionId, sizeof(audio_session_t));
+            return NO_ERROR;
+        }
+
+        status_t writeToParcel(Parcel *parcel) const override {
+            /* input arguments*/
+            (void)parcel->write(&attr, sizeof(audio_attributes_t));
+            (void)parcel->write(&config, sizeof(audio_config_t));
+            (void)clientInfo.writeToParcel(parcel);
+            if (sharedBuffer != 0) {
+                (void)parcel->writeInt32(1);
+                (void)parcel->writeStrongBinder(IInterface::asBinder(sharedBuffer));
+            } else {
+                (void)parcel->writeInt32(0);
+            }
+            (void)parcel->writeInt32(notificationsPerBuffer);
+            (void)parcel->writeFloat(speed);
+
+            /* input/output arguments*/
+            (void)parcel->write(&flags, sizeof(audio_output_flags_t));
+            (void)parcel->writeInt64(frameCount);
+            (void)parcel->writeInt64(notificationFrameCount);
+            (void)parcel->write(&selectedDeviceId, sizeof(audio_port_handle_t));
+            (void)parcel->write(&sessionId, sizeof(audio_session_t));
+            return NO_ERROR;
+        }
+
+        /* input */
+        audio_attributes_t attr;
+        audio_config_t config;
+        AudioClient clientInfo;
+        sp<IMemory> sharedBuffer;
+        uint32_t notificationsPerBuffer;
+        float speed;
+
+        /* input/output */
+        audio_output_flags_t flags;
+        size_t frameCount;
+        size_t notificationFrameCount;
+        audio_port_handle_t selectedDeviceId;
+        audio_session_t sessionId;
+    };
+
+    /* CreateTrackOutput contains all output arguments returned by AudioFlinger to AudioTrack
+     * when calling createTrack() including arguments that were passed as I/O for update by
+     * CreateTrackInput.
+     */
+    class CreateTrackOutput : public Parcelable {
+    public:
+        status_t readFromParcel(const Parcel *parcel) override {
+            /* input/output arguments*/
+            (void)parcel->read(&flags, sizeof(audio_output_flags_t));
+            frameCount = parcel->readInt64();
+            notificationFrameCount = parcel->readInt64();
+            (void)parcel->read(&selectedDeviceId, sizeof(audio_port_handle_t));
+            (void)parcel->read(&sessionId, sizeof(audio_session_t));
+
+            /* output arguments*/
+            sampleRate = parcel->readUint32();
+            afFrameCount = parcel->readInt64();
+            afSampleRate = parcel->readInt64();
+            afLatencyMs = parcel->readInt32();
+            (void)parcel->read(&outputId, sizeof(audio_io_handle_t));
+            return NO_ERROR;
+        }
+
+        status_t writeToParcel(Parcel *parcel) const override {
+            /* input/output arguments*/
+            (void)parcel->write(&flags, sizeof(audio_output_flags_t));
+            (void)parcel->writeInt64(frameCount);
+            (void)parcel->writeInt64(notificationFrameCount);
+            (void)parcel->write(&selectedDeviceId, sizeof(audio_port_handle_t));
+            (void)parcel->write(&sessionId, sizeof(audio_session_t));
+
+            /* output arguments*/
+            (void)parcel->writeUint32(sampleRate);
+            (void)parcel->writeInt64(afFrameCount);
+            (void)parcel->writeInt64(afSampleRate);
+            (void)parcel->writeInt32(afLatencyMs);
+            (void)parcel->write(&outputId, sizeof(audio_io_handle_t));
+            return NO_ERROR;
+        }
+
+        /* input/output */
+        audio_output_flags_t flags;
+        size_t frameCount;
+        size_t notificationFrameCount;
+        audio_port_handle_t selectedDeviceId;
+        audio_session_t sessionId;
+
+        /* output */
+        uint32_t sampleRate;
+        size_t   afFrameCount;
+        uint32_t afSampleRate;
+        uint32_t afLatencyMs;
+        audio_io_handle_t outputId;
+    };
+
+    /* CreateRecordInput contains all input arguments sent by AudioRecord to AudioFlinger
+     * when calling createRecord() including arguments that will be updated by AudioFlinger
+     * and returned in CreateRecordOutput object
+     */
+    class CreateRecordInput : public Parcelable {
+    public:
+        status_t readFromParcel(const Parcel *parcel) override {
+            /* input arguments*/
+            memset(&attr, 0, sizeof(audio_attributes_t));
+            if (parcel->read(&attr, sizeof(audio_attributes_t)) != NO_ERROR) {
+                return DEAD_OBJECT;
+            }
+            attr.tags[AUDIO_ATTRIBUTES_TAGS_MAX_SIZE -1] = '\0';
+            memset(&config, 0, sizeof(audio_config_base_t));
+            if (parcel->read(&config, sizeof(audio_config_base_t)) != NO_ERROR) {
+                return DEAD_OBJECT;
+            }
+            if (clientInfo.readFromParcel(parcel) != NO_ERROR) {
+                return DEAD_OBJECT;
+            }
+            opPackageName = parcel->readString16();
+
+            /* input/output arguments*/
+            (void)parcel->read(&flags, sizeof(audio_input_flags_t));
+            frameCount = parcel->readInt64();
+            notificationFrameCount = parcel->readInt64();
+            (void)parcel->read(&selectedDeviceId, sizeof(audio_port_handle_t));
+            (void)parcel->read(&sessionId, sizeof(audio_session_t));
+            return NO_ERROR;
+        }
+
+        status_t writeToParcel(Parcel *parcel) const override {
+            /* input arguments*/
+            (void)parcel->write(&attr, sizeof(audio_attributes_t));
+            (void)parcel->write(&config, sizeof(audio_config_base_t));
+            (void)clientInfo.writeToParcel(parcel);
+            (void)parcel->writeString16(opPackageName);
+
+            /* input/output arguments*/
+            (void)parcel->write(&flags, sizeof(audio_input_flags_t));
+            (void)parcel->writeInt64(frameCount);
+            (void)parcel->writeInt64(notificationFrameCount);
+            (void)parcel->write(&selectedDeviceId, sizeof(audio_port_handle_t));
+            (void)parcel->write(&sessionId, sizeof(audio_session_t));
+            return NO_ERROR;
+        }
+
+        /* input */
+        audio_attributes_t attr;
+        audio_config_base_t config;
+        AudioClient clientInfo;
+        String16 opPackageName;
+
+        /* input/output */
+        audio_input_flags_t flags;
+        size_t frameCount;
+        size_t notificationFrameCount;
+        audio_port_handle_t selectedDeviceId;
+        audio_session_t sessionId;
+    };
+
+    /* CreateRecordOutput contains all output arguments returned by AudioFlinger to AudioRecord
+     * when calling createRecord() including arguments that were passed as I/O for update by
+     * CreateRecordInput.
+     */
+    class CreateRecordOutput : public Parcelable {
+    public:
+        status_t readFromParcel(const Parcel *parcel) override {
+            /* input/output arguments*/
+            (void)parcel->read(&flags, sizeof(audio_input_flags_t));
+            frameCount = parcel->readInt64();
+            notificationFrameCount = parcel->readInt64();
+            (void)parcel->read(&selectedDeviceId, sizeof(audio_port_handle_t));
+            (void)parcel->read(&sessionId, sizeof(audio_session_t));
+
+            /* output arguments*/
+            sampleRate = parcel->readUint32();
+            (void)parcel->read(&inputId, sizeof(audio_io_handle_t));
+            if (parcel->readInt32() != 0) {
+                cblk = interface_cast<IMemory>(parcel->readStrongBinder());
+                if (cblk == 0 || cblk->pointer() == NULL) {
+                    return BAD_VALUE;
+                }
+            }
+            if (parcel->readInt32() != 0) {
+                buffers = interface_cast<IMemory>(parcel->readStrongBinder());
+                if (buffers == 0 || buffers->pointer() == NULL) {
+                    return BAD_VALUE;
+                }
+            }
+            return NO_ERROR;
+        }
+
+        status_t writeToParcel(Parcel *parcel) const override {
+            /* input/output arguments*/
+            (void)parcel->write(&flags, sizeof(audio_input_flags_t));
+            (void)parcel->writeInt64(frameCount);
+            (void)parcel->writeInt64(notificationFrameCount);
+            (void)parcel->write(&selectedDeviceId, sizeof(audio_port_handle_t));
+            (void)parcel->write(&sessionId, sizeof(audio_session_t));
+
+            /* output arguments*/
+            (void)parcel->writeUint32(sampleRate);
+            (void)parcel->write(&inputId, sizeof(audio_io_handle_t));
+            if (cblk != 0) {
+                (void)parcel->writeInt32(1);
+                (void)parcel->writeStrongBinder(IInterface::asBinder(cblk));
+            } else {
+                (void)parcel->writeInt32(0);
+            }
+            if (buffers != 0) {
+                (void)parcel->writeInt32(1);
+                (void)parcel->writeStrongBinder(IInterface::asBinder(buffers));
+            } else {
+                (void)parcel->writeInt32(0);
+            }
+
+            return NO_ERROR;
+        }
+
+        /* input/output */
+        audio_input_flags_t flags;
+        size_t frameCount;
+        size_t notificationFrameCount;
+        audio_port_handle_t selectedDeviceId;
+        audio_session_t sessionId;
+
+        /* output */
+        uint32_t sampleRate;
+        audio_io_handle_t inputId;
+        sp<IMemory> cblk;
+        sp<IMemory> buffers;
+    };
 
     // invariant on exit for all APIs that return an sp<>:
     //   (return value != 0) == (*status == NO_ERROR)
@@ -50,45 +321,13 @@
     /* create an audio track and registers it with AudioFlinger.
      * return null if the track cannot be created.
      */
-    virtual sp<IAudioTrack> createTrack(
-                                audio_stream_type_t streamType,
-                                uint32_t sampleRate,
-                                audio_format_t format,
-                                audio_channel_mask_t channelMask,
-                                size_t *pFrameCount,
-                                audio_output_flags_t *flags,
-                                const sp<IMemory>& sharedBuffer,
-                                // On successful return, AudioFlinger takes over the handle
-                                // reference and will release it when the track is destroyed.
-                                // However on failure, the client is responsible for release.
-                                audio_io_handle_t output,
-                                pid_t pid,
-                                pid_t tid,  // -1 means unused, otherwise must be valid non-0
-                                audio_session_t *sessionId,
-                                int clientUid,
-                                status_t *status,
-                                audio_port_handle_t portId) = 0;
+    virtual sp<IAudioTrack> createTrack(const CreateTrackInput& input,
+                                        CreateTrackOutput& output,
+                                        status_t *status) = 0;
 
-    virtual sp<IAudioRecord> openRecord(
-                                // On successful return, AudioFlinger takes over the handle
-                                // reference and will release it when the track is destroyed.
-                                // However on failure, the client is responsible for release.
-                                audio_io_handle_t input,
-                                uint32_t sampleRate,
-                                audio_format_t format,
-                                audio_channel_mask_t channelMask,
-                                const String16& callingPackage,
-                                size_t *pFrameCount,
-                                audio_input_flags_t *flags,
-                                pid_t pid,
-                                pid_t tid,  // -1 means unused, otherwise must be valid non-0
-                                int clientUid,
-                                audio_session_t *sessionId,
-                                size_t *notificationFrames,
-                                sp<IMemory>& cblk,
-                                sp<IMemory>& buffers,   // return value 0 means it follows cblk
-                                status_t *status,
-                                audio_port_handle_t portId) = 0;
+    virtual sp<media::IAudioRecord> createRecord(const CreateRecordInput& input,
+                                        CreateRecordOutput& output,
+                                        status_t *status) = 0;
 
     // FIXME Surprisingly, format/latency don't work for input handles
 
@@ -131,6 +370,7 @@
     // mic mute/state
     virtual     status_t    setMicMute(bool state) = 0;
     virtual     bool        getMicMute() const = 0;
+    virtual     void        setRecordSilenced(uid_t uid, bool silenced) = 0;
 
     virtual     status_t    setParameters(audio_io_handle_t ioHandle,
                                     const String8& keyValuePairs) = 0;
@@ -216,8 +456,9 @@
 
     // Intended for AudioService to inform AudioFlinger of device's low RAM attribute,
     // and should be called at most once.  For a definition of what "low RAM" means, see
-    // android.app.ActivityManager.isLowRamDevice().
-    virtual status_t setLowRamDevice(bool isLowRamDevice) = 0;
+    // android.app.ActivityManager.isLowRamDevice().  The totalMemory parameter
+    // is obtained from android.app.ActivityManager.MemoryInfo.totalMem.
+    virtual status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory) = 0;
 
     /* List available audio ports and their attributes */
     virtual status_t listAudioPorts(unsigned int *num_ports,
@@ -247,6 +488,9 @@
 
     // Returns the number of frames per audio HAL buffer.
     virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const = 0;
+
+    /* List available microphones and their characteristics */
+    virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones) = 0;
 };
 
 

diff --git a/media/libaudioclient/include/media/IAudioPolicyService.h b/media/libaudioclient/include/media/IAudioPolicyService.h
index eec3e88..c3876af 100644
--- a/media/libaudioclient/include/media/IAudioPolicyService.h
+++ b/media/libaudioclient/include/media/IAudioPolicyService.h

@@ -55,16 +55,12 @@
     virtual status_t setForceUse(audio_policy_force_use_t usage,
                                     audio_policy_forced_cfg_t config) = 0;
     virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) = 0;
-    virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
-                                        uint32_t samplingRate = 0,
-                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
-                                        audio_channel_mask_t channelMask = 0,
-                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
-                                        const audio_offload_info_t *offloadInfo = NULL) = 0;
+    virtual audio_io_handle_t getOutput(audio_stream_type_t stream) = 0;
     virtual status_t getOutputForAttr(const audio_attributes_t *attr,
                                       audio_io_handle_t *output,
                                       audio_session_t session,
                                       audio_stream_type_t *stream,
+                                      pid_t pid,
                                       uid_t uid,
                                       const audio_config_t *config,
                                       audio_output_flags_t flags,
@@ -84,16 +80,15 @@
                               audio_session_t session,
                               pid_t pid,
                               uid_t uid,
+                              const String16& opPackageName,
                               const audio_config_base_t *config,
                               audio_input_flags_t flags,
                               audio_port_handle_t *selectedDeviceId,
                               audio_port_handle_t *portId) = 0;
-    virtual status_t startInput(audio_io_handle_t input,
-                                audio_session_t session) = 0;
-    virtual status_t stopInput(audio_io_handle_t input,
-                               audio_session_t session) = 0;
-    virtual void releaseInput(audio_io_handle_t input,
-                              audio_session_t session) = 0;
+    virtual status_t startInput(audio_port_handle_t portId,
+                                bool *silenced) = 0;
+    virtual status_t stopInput(audio_port_handle_t portId) = 0;
+    virtual void releaseInput(audio_port_handle_t portId) = 0;
     virtual status_t initStreamVolume(audio_stream_type_t stream,
                                       int indexMin,
                                       int indexMax) = 0;
@@ -171,6 +166,12 @@
     virtual status_t getMasterMono(bool *mono) = 0;
     virtual float    getStreamVolumeDB(
             audio_stream_type_t stream, int index, audio_devices_t device) = 0;
+
+    virtual status_t getSurroundFormats(unsigned int *numSurroundFormats,
+                                        audio_format_t *surroundFormats,
+                                        bool *surroundFormatsEnabled,
+                                        bool reported) = 0;
+    virtual status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled) = 0;
 };
 
 

diff --git a/media/libaudioclient/include/media/IAudioRecord.h b/media/libaudioclient/include/media/IAudioRecord.h
deleted file mode 100644
index 7768176..0000000
--- a/media/libaudioclient/include/media/IAudioRecord.h
+++ /dev/null

@@ -1,66 +0,0 @@
-/*
- * Copyright (C) 2007 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef IAUDIORECORD_H_
-#define IAUDIORECORD_H_
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <utils/RefBase.h>
-#include <utils/Errors.h>
-#include <binder/IInterface.h>
-#include <binder/IMemory.h>
-#include <system/audio.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-class IAudioRecord : public IInterface
-{
-public:
-    DECLARE_META_INTERFACE(AudioRecord);
-
-    /* After it's created the track is not active. Call start() to
-     * make it active.
-     */
-    virtual status_t    start(int /*AudioSystem::sync_event_t*/ event,
-                              audio_session_t triggerSession) = 0;
-
-    /* Stop a track. If set, the callback will cease being called and
-     * obtainBuffer will return an error. Buffers that are already released
-     * will be processed, unless flush() is called.
-     */
-    virtual void        stop() = 0;
-};
-
-// ----------------------------------------------------------------------------
-
-class BnAudioRecord : public BnInterface<IAudioRecord>
-{
-public:
-    virtual status_t    onTransact( uint32_t code,
-                                    const Parcel& data,
-                                    Parcel* reply,
-                                    uint32_t flags = 0);
-};
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
-
-#endif /*IAUDIORECORD_H_*/

diff --git a/media/libaudioclient/include/media/IAudioTrack.h b/media/libaudioclient/include/media/IAudioTrack.h
index 27a62d6..94afe3c 100644
--- a/media/libaudioclient/include/media/IAudioTrack.h
+++ b/media/libaudioclient/include/media/IAudioTrack.h

@@ -77,12 +77,12 @@
     virtual void        signal() = 0;
 
     /* Sets the volume shaper */
-    virtual VolumeShaper::Status applyVolumeShaper(
-            const sp<VolumeShaper::Configuration>& configuration,
-            const sp<VolumeShaper::Operation>& operation) = 0;
+    virtual media::VolumeShaper::Status applyVolumeShaper(
+            const sp<media::VolumeShaper::Configuration>& configuration,
+            const sp<media::VolumeShaper::Operation>& operation) = 0;
 
     /* gets the volume shaper state */
-    virtual sp<VolumeShaper::State> getVolumeShaperState(int id) = 0;
+    virtual sp<media::VolumeShaper::State> getVolumeShaperState(int id) = 0;
 };
 
 // ----------------------------------------------------------------------------

diff --git a/media/libaudioclient/include/media/PlayerBase.h b/media/libaudioclient/include/media/PlayerBase.h
index e63090b..e7a8abc 100644
--- a/media/libaudioclient/include/media/PlayerBase.h
+++ b/media/libaudioclient/include/media/PlayerBase.h

@@ -17,35 +17,31 @@
 #ifndef __ANDROID_PLAYER_BASE_H__
 #define __ANDROID_PLAYER_BASE_H__
 
-#include <audiomanager/IPlayer.h>
 #include <audiomanager/AudioManager.h>
 #include <audiomanager/IAudioManager.h>
 
+#include "android/media/BnPlayer.h"
 
 namespace android {
 
-class PlayerBase : public BnPlayer
+class PlayerBase : public ::android::media::BnPlayer
 {
 public:
     explicit PlayerBase();
-    virtual ~PlayerBase();
+    virtual ~PlayerBase() override;
 
     virtual void destroy() = 0;
 
     //IPlayer implementation
-    virtual void start();
-    virtual void pause();
-    virtual void stop();
-    virtual void setVolume(float vol);
-    virtual void setPan(float pan);
-    virtual void setStartDelayMs(int32_t delayMs);
-    virtual void applyVolumeShaper(
-            const sp<VolumeShaper::Configuration>& configuration,
-            const sp<VolumeShaper::Operation>& operation) override;
-
-    virtual status_t onTransact(
-                uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
-
+    virtual binder::Status start() override;
+    virtual binder::Status pause() override;
+    virtual binder::Status stop() override;
+    virtual binder::Status setVolume(float vol) override;
+    virtual binder::Status setPan(float pan) override;
+    virtual binder::Status setStartDelayMs(int32_t delayMs) override;
+    virtual binder::Status applyVolumeShaper(
+            const media::VolumeShaper::Configuration& configuration,
+            const media::VolumeShaper::Operation& operation) override;
 
             status_t startWithStatus();
             status_t pauseWithStatus();

diff --git a/media/libaudioclient/include/media/ToneGenerator.h b/media/libaudioclient/include/media/ToneGenerator.h
index fc3d3ee..7ae6aa6 100644
--- a/media/libaudioclient/include/media/ToneGenerator.h
+++ b/media/libaudioclient/include/media/ToneGenerator.h

@@ -177,7 +177,7 @@
 
 
     // Region specific tones.
-    // These supervisory tones are different depending on the region (USA/CANADA, JAPAN, rest of the world).
+    // These supervisory tones are different depending on the region (USA/CANADA, JAPAN, Singapore, Hong Kong, rest of the world).
     // When a tone in the range [FIRST_SUP_TONE, LAST_SUP_TONE] is requested, the region is determined
     // from system property gsm.operator.iso-country and the proper tone descriptor is selected with the
     // help of sToneMappingTable[]
@@ -194,12 +194,30 @@
         TONE_JAPAN_BUSY,            // Busy tone: 400Hz, 500ms ON, 500ms OFF...
         TONE_JAPAN_RADIO_ACK,       // Radio path acknowlegment: 400Hz, 1s ON, 2s OFF...
         // GB Supervisory tones
+        TONE_GB_BUSY,               // Busy tone: 400 Hz, 375ms ON, 375ms OFF...
+        TONE_GB_CONGESTION,         // Congestion Tone: 400 Hz, 400ms ON, 350ms OFF, 225ms ON, 525ms OFF...
         TONE_GB_RINGTONE,           // Ring Tone: A 400Hz + 450Hz tone repeated in a 0.4s on, 0.2s off, 0.4s on, 2.0s off pattern.
         // AUSTRALIA Supervisory tones
+        TONE_AUSTRALIA_DIAL,        // Dial tone: 425 Hz tone modulated with 25 Hz, continuous
         TONE_AUSTRALIA_RINGTONE,    // Ring tone: A 400Hz + 450Hz tone repeated in a 0.4s on, 0.2s off, 0.4s on, 2.0s off pattern.
         TONE_AUSTRALIA_BUSY,        // Busy tone: 425 Hz repeated in a 0.375s on, 0.375s off pattern.
         TONE_AUSTRALIA_CALL_WAITING,// Call waiting tone: 425Hz tone repeated in a 0.2s on, 0.2s off, 0.2s on, 4.4s off pattern.
         TONE_AUSTRALIA_CONGESTION,  // Congestion tone: 425Hz tone repeated in a 0.375s on, 0.375s off pattern
+        // SINGAPORE Supervisory tones
+        TONE_SG_BUSY,               // Busy tone: 425 Hz, 750ms ON, 750ms OFF...
+        TONE_SG_RINGTONE,           // Ring Tone: 425 Hz tone modulated with 24 Hz, 400ms ON, 200ms OFF, 400ms ON, 2s OFF...
+        // HONG KONG Supervisory tones
+        TONE_HK_BUSY,               // Busy tone: 480 Hz + 620 Hz, 500ms ON, 500ms OFF...
+        TONE_HK_RINGTONE,           // Ring Tone: 440 Hz + 480 Hz repeated with pattern 0,4s on, 0,2s off, 0,4s on and 3s off.
+        // IRELAND Supervisory tones
+        TONE_IE_RINGTONE,           // Ring Tone: A 400Hz + 450Hz tone repeated in a 0.4s on, 0.2s off, 0.4s on, 2.0s off pattern.
+        TONE_IE_CALL_WAITING,       // Call waiting tone: 425Hz tone repeated in a 0.18s on, 0.2s off, 0.2s on, 4.5s off pattern.
+        // INDIA supervisory tones
+        TONE_INDIA_DIAL,            // Dial tone: 400 Hz tone modulated with 25Hz, continuous
+        TONE_INDIA_BUSY,            // Busy tone: 400 Hz, 750ms ON, 750ms OFF...
+        TONE_INDIA_CONGESTION,      // Congestion tone: 400 Hz, 250ms ON, 250ms OFF...
+        TONE_INDIA_CALL_WAITING,    // Call waiting tone: 400 Hz, tone repeated in a 0.2s on, 0.1s off, 0.2s on, 7.5s off pattern.
+        TONE_INDIA_RINGTONE,        // Ring tone: 400 Hz tone modulated with 25Hz, 0.4 on 0.2 off 0.4 on 2..0 off
         NUM_ALTERNATE_TONES
     };
 
@@ -208,6 +226,10 @@
         JAPAN,
         GB,
         AUSTRALIA,
+        SINGAPORE,
+        HONGKONG,
+        IRELAND,
+        INDIA,
         CEPT,
         NUM_REGIONS
     };

diff --git a/media/libaudioclient/include/media/TrackPlayerBase.h b/media/libaudioclient/include/media/TrackPlayerBase.h
index 2d113c0..66e9b3b 100644
--- a/media/libaudioclient/include/media/TrackPlayerBase.h
+++ b/media/libaudioclient/include/media/TrackPlayerBase.h

@@ -32,9 +32,9 @@
     virtual void destroy();
 
     //IPlayer implementation
-    virtual void applyVolumeShaper(
-            const sp<VolumeShaper::Configuration>& configuration,
-            const sp<VolumeShaper::Operation>& operation);
+    virtual binder::Status applyVolumeShaper(
+            const media::VolumeShaper::Configuration& configuration,
+            const media::VolumeShaper::Operation& operation);
 
     //FIXME move to protected field, so far made public to minimize changes to AudioTrack logic
     sp<AudioTrack> mAudioTrack;

diff --git a/media/libaudioclient/tests/Android.bp b/media/libaudioclient/tests/Android.bp
new file mode 100644
index 0000000..52bb2fb
--- /dev/null
+++ b/media/libaudioclient/tests/Android.bp

@@ -0,0 +1,35 @@
+cc_defaults {
+    name: "libaudioclient_tests_defaults",
+    cflags: [
+        "-Wall",
+        "-Werror",
+    ],
+}
+
+cc_test {
+    name: "test_create_audiotrack",
+    defaults: ["libaudioclient_tests_defaults"],
+    srcs: ["test_create_audiotrack.cpp",
+           "test_create_utils.cpp"],
+    shared_libs: [
+        "libaudioclient",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+    data: ["track_test_input_*.txt"],
+}
+
+cc_test {
+    name: "test_create_audiorecord",
+    defaults: ["libaudioclient_tests_defaults"],
+    srcs: ["test_create_audiorecord.cpp",
+           "test_create_utils.cpp"],
+    shared_libs: [
+        "libaudioclient",
+        "libbinder",
+        "libcutils",
+        "libutils",
+    ],
+    data: ["record_test_input_*.txt"],
+}

diff --git a/media/libaudioclient/tests/record_test_input_v1.0_ref.txt b/media/libaudioclient/tests/record_test_input_v1.0_ref.txt
new file mode 100644
index 0000000..e01598e
--- /dev/null
+++ b/media/libaudioclient/tests/record_test_input_v1.0_ref.txt

@@ -0,0 +1,33 @@
+version 1.0
+# Input file for test_create_audiorecord
+# Add one line for each tested AudioRecord constructor with the following arguments:
+# sampleRate format   channelMask frameCount notificationFrames flags sessionId inputSource
+# sample rate tests
+  48000      0x1      0x10        4800       2400               0x0   0         0
+  24000      0x1      0x10        4800       2400               0x0   0         0
+  16000      0x1      0x10        4800       2400               0x0   0         0
+   8000      0x1      0x10        4800       2400               0x0   0         0
+  44100      0x1      0x10        4410       2205               0x0   0         0
+  22050      0x1      0x10        4410       2205               0x0   0         0
+  11025      0x1      0x10        4410       2205               0x0   0         0
+# format tests
+  48000      0x2      0x10        4800       2400               0x0   0         0
+  48000      0x3      0x10        4800       2400               0x0   0         0
+  48000      0x5      0x10        4800       2400               0x0   0         0
+# channel mask tests
+  48000      0x1      0x0C        4800       2400               0x0   0         0
+# frame count tests
+  48000      0x1      0x10        0          0                  0x0   0         0
+  48000      0x1      0x10        48000      0                  0x0   0         0
+  48000      0x1      0x10        12000      6000               0x0   0         0
+# flags test
+  48000      0x1      0x0C        0          0                  0x1   0         0
+  44100      0x1      0x0C        0          0                  0x5   0         0
+# session tests
+  48000      0x1      0x10        0          0                  0     1001      0
+# input source tests
+  48000      0x1      0x10        0          0                  0     0         1
+  48000      0x1      0x10        0          0                  0     0         5
+  48000      0x1      0x10        0          0                  0     0         6
+  48000      0x1      0x10        0          0                  0     0         7
+  48000      0x1      0x10        0          0                  0     0         9

diff --git a/media/libaudioclient/tests/record_test_output_v1.0_ref_walleye.txt b/media/libaudioclient/tests/record_test_output_v1.0_ref_walleye.txt
new file mode 100644
index 0000000..76608eb
--- /dev/null
+++ b/media/libaudioclient/tests/record_test_output_v1.0_ref_walleye.txt

@@ -0,0 +1,198 @@
+
+#### Test 1 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(65)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(960), req. notif. frame count(2400)
+  input(150), latency(100), selected device Id(0), routed device Id(11)
+
+#### Test 2 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(73)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(24000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(480), req. notif. frame count(2400)
+  input(158), latency(200), selected device Id(0), routed device Id(11)
+
+#### Test 3 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(81)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(16000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(320), req. notif. frame count(2400)
+  input(166), latency(300), selected device Id(0), routed device Id(11)
+
+#### Test 4 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(89)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(8000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(160), req. notif. frame count(2400)
+  input(174), latency(600), selected device Id(0), routed device Id(11)
+
+#### Test 5 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(97)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(44100)
+  frame count(4410), req. frame count(4410)
+  notif. frame count(896), req. notif. frame count(2205)
+  input(182), latency(100), selected device Id(0), routed device Id(11)
+
+#### Test 6 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(105)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(22050)
+  frame count(4410), req. frame count(4410)
+  notif. frame count(448), req. notif. frame count(2205)
+  input(190), latency(200), selected device Id(0), routed device Id(11)
+
+#### Test 7 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(113)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(11025)
+  frame count(4410), req. frame count(4410)
+  notif. frame count(224), req. notif. frame count(2205)
+  input(198), latency(400), selected device Id(0), routed device Id(11)
+
+#### Test 8 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(121)
+  flags(0), req. flags(0), audio source(0)
+  format(0x2), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(960), req. notif. frame count(2400)
+  input(206), latency(100), selected device Id(0), routed device Id(11)
+
+#### Test 9 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(129)
+  flags(0), req. flags(0), audio source(0)
+  format(0x3), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(960), req. notif. frame count(2400)
+  input(214), latency(100), selected device Id(0), routed device Id(11)
+
+#### Test 10 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(137)
+  flags(0), req. flags(0), audio source(0)
+  format(0x5), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(960), req. notif. frame count(2400)
+  input(222), latency(100), selected device Id(0), routed device Id(11)
+
+#### Test 11 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(145)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0xc), channel count(2), sample rate(48000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(960), req. notif. frame count(2400)
+  input(230), latency(100), selected device Id(0), routed device Id(11)
+
+#### Test 12 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(153)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(2880), req. frame count(2880)
+  notif. frame count(960), req. notif. frame count(0)
+  input(238), latency(60), selected device Id(0), routed device Id(11)
+
+#### Test 13 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(161)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(48000), req. frame count(48000)
+  notif. frame count(960), req. notif. frame count(0)
+  input(246), latency(1000), selected device Id(0), routed device Id(11)
+
+#### Test 14 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(169)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(12000), req. frame count(12000)
+  notif. frame count(960), req. notif. frame count(6000)
+  input(254), latency(250), selected device Id(0), routed device Id(11)
+
+#### Test 15 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(177)
+  flags(0x1), req. flags(0x1), audio source(0)
+  format(0x1), channel mask(0xc), channel count(2), sample rate(48000)
+  frame count(4096), req. frame count(4096)
+  notif. frame count(96), req. notif. frame count(0)
+  input(262), latency(85), selected device Id(0), routed device Id(11)
+
+#### Test 16 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(185)
+  flags(0), req. flags(0x5), audio source(0)
+  format(0x1), channel mask(0xc), channel count(2), sample rate(44100)
+  frame count(2664), req. frame count(2664)
+  notif. frame count(888), req. notif. frame count(0)
+  input(278), latency(60), selected device Id(0), routed device Id(11)
+
+#### Test 17 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(1001)
+  flags(0), req. flags(0), audio source(0)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(2880), req. frame count(2880)
+  notif. frame count(960), req. notif. frame count(0)
+  input(286), latency(60), selected device Id(0), routed device Id(11)
+
+#### Test 18 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(193)
+  flags(0), req. flags(0), audio source(1)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(2880), req. frame count(2880)
+  notif. frame count(960), req. notif. frame count(0)
+  input(294), latency(60), selected device Id(0), routed device Id(11)
+
+#### Test 19 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(201)
+  flags(0), req. flags(0), audio source(5)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(2880), req. frame count(2880)
+  notif. frame count(960), req. notif. frame count(0)
+  input(302), latency(60), selected device Id(0), routed device Id(12)
+
+#### Test 20 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(209)
+  flags(0), req. flags(0), audio source(6)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(2880), req. frame count(2880)
+  notif. frame count(960), req. notif. frame count(0)
+  input(310), latency(60), selected device Id(0), routed device Id(11)
+
+#### Test 21 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(217)
+  flags(0), req. flags(0), audio source(7)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(2880), req. frame count(2880)
+  notif. frame count(960), req. notif. frame count(0)
+  input(318), latency(60), selected device Id(0), routed device Id(11)
+
+#### Test 22 status 0
+ AudioRecord::dump
+  status(0), active(0), session Id(225)
+  flags(0), req. flags(0), audio source(9)
+  format(0x1), channel mask(0x10), channel count(1), sample rate(48000)
+  frame count(2880), req. frame count(2880)
+  notif. frame count(960), req. notif. frame count(0)
+  input(326), latency(60), selected device Id(0), routed device Id(11)

diff --git a/media/libaudioclient/tests/test_create_audiorecord.cpp b/media/libaudioclient/tests/test_create_audiorecord.cpp
new file mode 100644
index 0000000..cf6a734
--- /dev/null
+++ b/media/libaudioclient/tests/test_create_audiorecord.cpp

@@ -0,0 +1,129 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <fcntl.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
+#include <binder/MemoryBase.h>
+#include <binder/MemoryDealer.h>
+#include <binder/MemoryHeapBase.h>
+#include <media/AudioRecord.h>
+
+#include "test_create_utils.h"
+
+#define NUM_ARGUMENTS 8
+#define VERSION_VALUE "1.0"
+#define PACKAGE_NAME  "AudioRecord test"
+
+namespace android {
+
+int testRecord(FILE *inputFile, int outputFileFd)
+{
+    char line[MAX_INPUT_FILE_LINE_LENGTH];
+    uint32_t testCount = 0;
+    Vector<String16> args;
+    int ret = 0;
+
+    if (inputFile == nullptr) {
+        sp<AudioRecord> record = new AudioRecord(AUDIO_SOURCE_DEFAULT,
+                                              0 /* sampleRate */,
+                                              AUDIO_FORMAT_DEFAULT,
+                                              AUDIO_CHANNEL_IN_MONO,
+                                              String16(PACKAGE_NAME));
+        if (record == 0 || record->initCheck() != NO_ERROR) {
+            write(outputFileFd, "Error creating AudioRecord\n",
+                  sizeof("Error creating AudioRecord\n"));
+        } else {
+            record->dump(outputFileFd, args);
+        }
+        return 0;
+    }
+
+    // check version
+    if (!checkVersion(inputFile, VERSION_VALUE)) {
+        return 1;
+    }
+
+    while (readLine(inputFile, line, MAX_INPUT_FILE_LINE_LENGTH) == 0) {
+        uint32_t sampleRate;
+        audio_format_t format;
+        audio_channel_mask_t channelMask;
+        size_t frameCount;
+        int32_t notificationFrames;
+        audio_input_flags_t flags;
+        audio_session_t sessionId;
+        audio_source_t inputSource;
+        audio_attributes_t attributes;
+        status_t status;
+        char statusStr[MAX_OUTPUT_FILE_LINE_LENGTH];
+        bool fast = false;
+
+        if (sscanf(line, " %u %x %x %zu %d %x %u %u",
+                   &sampleRate, &format, &channelMask,
+                   &frameCount, &notificationFrames,
+                   &flags, &sessionId, &inputSource) != NUM_ARGUMENTS) {
+            fprintf(stderr, "Malformed line for test #%u in input file\n", testCount+1);
+            ret = 1;
+            continue;
+        }
+        testCount++;
+
+        if ((flags & AUDIO_INPUT_FLAG_FAST) != 0) {
+            fast = true;
+        }
+
+        memset(&attributes, 0, sizeof(attributes));
+        attributes.source = inputSource;
+
+        sp<AudioRecord> record = new AudioRecord(String16(PACKAGE_NAME));
+
+        record->set(AUDIO_SOURCE_DEFAULT,
+                   sampleRate,
+                   format,
+                   channelMask,
+                   frameCount,
+                   fast ? callback : nullptr,
+                   nullptr,
+                   notificationFrames,
+                   false,
+                   sessionId,
+                   fast ? AudioRecord::TRANSFER_CALLBACK : AudioRecord::TRANSFER_DEFAULT,
+                   flags,
+                   getuid(),
+                   getpid(),
+                   &attributes,
+                   AUDIO_PORT_HANDLE_NONE);
+        status = record->initCheck();
+        sprintf(statusStr, "\n#### Test %u status %d\n", testCount, status);
+        write(outputFileFd, statusStr, strlen(statusStr));
+        if (status != NO_ERROR) {
+            continue;
+        }
+        record->dump(outputFileFd, args);
+    }
+    return ret;
+}
+
+}; // namespace android
+
+
+int main(int argc, char **argv)
+{
+    return android::main(argc, argv, android::testRecord);
+}
+

diff --git a/media/libaudioclient/tests/test_create_audiotrack.cpp b/media/libaudioclient/tests/test_create_audiotrack.cpp
new file mode 100644
index 0000000..cf9b925
--- /dev/null
+++ b/media/libaudioclient/tests/test_create_audiotrack.cpp

@@ -0,0 +1,153 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <fcntl.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
+#include <binder/MemoryBase.h>
+#include <binder/MemoryDealer.h>
+#include <binder/MemoryHeapBase.h>
+#include <media/AudioTrack.h>
+
+#include "test_create_utils.h"
+
+#define NUM_ARGUMENTS 10
+#define VERSION_VALUE "1.0"
+
+namespace android {
+
+int testTrack(FILE *inputFile, int outputFileFd)
+{
+    char line[MAX_INPUT_FILE_LINE_LENGTH];
+    uint32_t testCount = 0;
+    Vector<String16> args;
+    int ret = 0;
+
+    if (inputFile == nullptr) {
+        sp<AudioTrack> track = new AudioTrack(AUDIO_STREAM_DEFAULT,
+                                              0 /* sampleRate */,
+                                              AUDIO_FORMAT_DEFAULT,
+                                              AUDIO_CHANNEL_OUT_STEREO);
+        if (track == 0 || track->initCheck() != NO_ERROR) {
+            write(outputFileFd, "Error creating AudioTrack\n",
+                  sizeof("Error creating AudioTrack\n"));
+        } else {
+            track->dump(outputFileFd, args);
+        }
+        return 0;
+    }
+
+    // check version
+    if (!checkVersion(inputFile, VERSION_VALUE)) {
+        return 1;
+    }
+
+    while (readLine(inputFile, line, MAX_INPUT_FILE_LINE_LENGTH) == 0) {
+        uint32_t sampleRate;
+        audio_format_t format;
+        audio_channel_mask_t channelMask;
+        size_t frameCount;
+        int32_t notificationFrames;
+        uint32_t useSharedBuffer;
+        audio_output_flags_t flags;
+        audio_session_t sessionId;
+        audio_usage_t usage;
+        audio_content_type_t contentType;
+        audio_attributes_t attributes;
+        sp<IMemory> sharedBuffer;
+        sp<MemoryDealer> heap;
+        audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
+        status_t status;
+        char statusStr[MAX_OUTPUT_FILE_LINE_LENGTH];
+        bool offload = false;
+        bool fast = false;
+
+        if (sscanf(line, " %u %x %x %zu %d %u %x %u %u %u",
+                   &sampleRate, &format, &channelMask,
+                   &frameCount, &notificationFrames, &useSharedBuffer,
+                   &flags, &sessionId, &usage, &contentType) != NUM_ARGUMENTS) {
+            fprintf(stderr, "Malformed line for test #%u in input file\n", testCount+1);
+            ret = 1;
+            continue;
+        }
+        testCount++;
+
+        if (useSharedBuffer != 0) {
+            size_t heapSize = audio_channel_count_from_out_mask(channelMask) *
+                    audio_bytes_per_sample(format) * frameCount;
+            heap = new MemoryDealer(heapSize, "AudioTrack Heap Base");
+            sharedBuffer = heap->allocate(heapSize);
+            frameCount = 0;
+            notificationFrames = 0;
+        }
+        if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+            offloadInfo.sample_rate = sampleRate;
+            offloadInfo.channel_mask = channelMask;
+            offloadInfo.format = format;
+            offload = true;
+        }
+        if ((flags & AUDIO_OUTPUT_FLAG_FAST) != 0) {
+            fast = true;
+        }
+
+        memset(&attributes, 0, sizeof(attributes));
+        attributes.content_type = contentType;
+        attributes.usage = usage;
+
+        sp<AudioTrack> track = new AudioTrack();
+
+        track->set(AUDIO_STREAM_DEFAULT,
+                   sampleRate,
+                   format,
+                   channelMask,
+                   frameCount,
+                   flags,
+                   (fast || offload) ? callback : nullptr,
+                   nullptr,
+                   notificationFrames,
+                   sharedBuffer,
+                   false,
+                   sessionId,
+                   ((fast && sharedBuffer == 0) || offload) ?
+                           AudioTrack::TRANSFER_CALLBACK : AudioTrack::TRANSFER_DEFAULT,
+                   offload ? &offloadInfo : nullptr,
+                   getuid(),
+                   getpid(),
+                   &attributes,
+                   false,
+                   1.0f,
+                   AUDIO_PORT_HANDLE_NONE);
+        status = track->initCheck();
+        sprintf(statusStr, "\n#### Test %u status %d\n", testCount, status);
+        write(outputFileFd, statusStr, strlen(statusStr));
+        if (status != NO_ERROR) {
+            continue;
+        }
+        track->dump(outputFileFd, args);
+    }
+    return ret;
+}
+
+}; // namespace android
+
+
+int main(int argc, char **argv)
+{
+    return android::main(argc, argv, android::testTrack);
+}
+

diff --git a/media/libaudioclient/tests/test_create_utils.cpp b/media/libaudioclient/tests/test_create_utils.cpp
new file mode 100644
index 0000000..8aa1f13
--- /dev/null
+++ b/media/libaudioclient/tests/test_create_utils.cpp

@@ -0,0 +1,134 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <fcntl.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
+#include "test_create_utils.h"
+
+namespace android {
+
+int readLine(FILE *inputFile, char *line, int size) {
+    int ret = 0;
+    while (true) {
+        char *str = fgets(line, size, inputFile);
+        if (str == nullptr) {
+            ret = -1;
+            break;
+        }
+        if (feof(inputFile) != 0 || ferror(inputFile) != 0) {
+            ret = -1;
+            break;
+        }
+        if (strlen(str) != 0 && str[0] != COMMENT_CHAR) {
+            break;
+        }
+    }
+    return ret;
+}
+
+bool checkVersion(FILE *inputFile, const char *version)
+{
+    char line[MAX_INPUT_FILE_LINE_LENGTH];
+    char versionKey[MAX_INPUT_FILE_LINE_LENGTH];
+    char versionValue[MAX_INPUT_FILE_LINE_LENGTH];
+
+    if (readLine(inputFile, line, MAX_INPUT_FILE_LINE_LENGTH) != 0) {
+        fprintf(stderr, "Missing version in input file\n");
+        return false;
+    }
+
+    if (sscanf(line, " %s %s", versionKey, versionValue) != 2) {
+        fprintf(stderr, "Malformed version in input file\n");
+        return false;
+    }
+    if (strcmp(versionKey, VERSION_KEY) != 0) {
+        fprintf(stderr, "Malformed version in input file\n");
+        return false;
+    }
+    if (strcmp(versionValue, version) != 0) {
+        fprintf(stderr, "Wrong input file version %s expecting %s\n", versionValue, version);
+        return false;
+    }
+    return true;
+}
+
+void callback(int event __unused, void* user __unused, void *info __unused)
+{
+}
+
+int main(int argc, char **argv, test_func_t testFunc)
+{
+    FILE *inputFile = nullptr;
+    int outputFileFd = STDOUT_FILENO;
+    mode_t mode = S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH;
+    int ret = 0;
+
+    if (argc > 5) {
+        fprintf(stderr, "Usage: %s [-i input_params.txt] [-o output_params.txt]\n", argv[0]);
+        return 1;
+    }
+
+    argv++;
+    while (*argv) {
+        if (strcmp(*argv, "-i") == 0) {
+            argv++;
+            if (*argv) {
+                inputFile = fopen(*argv, "r");
+                if (inputFile == nullptr) {
+                    ret = 1;
+                }
+            } else {
+                ret = 1;
+            }
+        }
+        if (strcmp(*argv, "-o") == 0) {
+            argv++;
+            if (*argv) {
+                outputFileFd = open(*argv, O_WRONLY|O_CREAT, mode);
+                if (outputFileFd < 0) {
+                    ret = 1;
+                }
+            } else {
+                ret = 1;
+            }
+            argv++;
+        }
+        if (*argv) {
+            argv++;
+        }
+    }
+
+    if (ret != 0) {
+        return ret;
+    }
+
+    ret = testFunc(inputFile, outputFileFd);
+
+    if (inputFile) {
+        fclose(inputFile);
+    }
+    if (outputFileFd >= 0 && outputFileFd != STDOUT_FILENO) {
+        close(outputFileFd);
+    }
+
+    return ret;
+}
+
+}; // namespace android
+

diff --git a/media/libaudioclient/tests/test_create_utils.h b/media/libaudioclient/tests/test_create_utils.h
new file mode 100644
index 0000000..2ad646e
--- /dev/null
+++ b/media/libaudioclient/tests/test_create_utils.h

@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <fcntl.h>
+#include <stdio.h>
+#include <string.h>
+#include <unistd.h>
+
+#define MAX_INPUT_FILE_LINE_LENGTH 512
+#define MAX_OUTPUT_FILE_LINE_LENGTH 512
+
+#define COMMENT_CHAR '#'
+#define VERSION_KEY "version"
+
+namespace android {
+
+int readLine(FILE *inputFile, char *line, int size);
+
+bool checkVersion(FILE *inputFile, const char *version);
+
+void callback(int event, void* user, void *info);
+
+typedef int (*test_func_t)(FILE *inputFile, int outputFileFd);
+
+int main(int argc, char **argv, test_func_t testFunc);
+
+}; // namespace android

diff --git a/media/libaudioclient/tests/track_test_input_v1.0_ref.txt b/media/libaudioclient/tests/track_test_input_v1.0_ref.txt
new file mode 100644
index 0000000..b923ff3
--- /dev/null
+++ b/media/libaudioclient/tests/track_test_input_v1.0_ref.txt

@@ -0,0 +1,40 @@
+version 1.0
+# Input file for test_create_audiotrack
+# Add one line for each tested AudioTrack constructor with the following arguments:
+# sampleRate format 	channelMask frameCount notificationFrames sharedBuffer flags sessionId usage contentType
+# sample rate tests
+  48000      0x1    	0x3         4800       2400               0            0x0   0         1     2
+  24000      0x1    	0x3         4800       2400               0            0x0   0         1     2
+  16000      0x1    	0x3         4800       2400               0            0x0   0         1     2
+   8000      0x1    	0x3         4800       2400               0            0x0   0         1     2
+  44100      0x1    	0x3         4410       2205               0            0x0   0         1     2
+  22050      0x1    	0x3         4410       2205               0            0x0   0         1     2
+  11025      0x1    	0x3         4410       2205               0            0x0   0         1     2
+# format tests
+  48000      0x2    	0x3         4800       2400               0            0x0   0         1     2
+  48000      0x3    	0x3         4800       2400               0            0x0   0         1     2
+  48000      0x5    	0x3         4800       2400               0            0x0   0         1     2
+# channel mask tests
+  48000      0x1    	0x1         4800       2400               0            0x0   0         1     2
+  48000      0x1    	0x3F        4800       2400               0            0x0   0         1     2
+  48000      0x1    	0x63F       4800       2400               0            0x0   0         1     2
+# framecount tests
+  48000      0x1    	0x3         0          0                  0            0x0   0         1     2
+  48000      0x1    	0x3         48000      0                  0            0x0   0         1     2
+  48000      0x1    	0x3         0          -2                 0            0x4   0         1     2
+# shared memory tests
+  48000      0x1    	0x3         4800       2400               1            0x0   0         1     2
+  48000      0x1    	0x3         4800       2400               1            0x4   0         1     2
+# flags test
+  48000      0x1    	0x3         4800       2400               0            0x4   0         1     2
+  48000      0x1    	0x3         4800       2400               0            0x8   0         1     2
+  44100      0x1000000  0x3         4800       2400               0            0x11  0         1     2
+# session tests
+  48000      0x1    	0x3         4800       2400               0            0x0   1001      1     2
+# attributes tests
+  48000      0x1    	0x3         4800       2400               0            0x0   0         0     0
+  48000      0x1    	0x3         4800       2400               0            0x0   0         2     1
+  48000      0x1    	0x3         4800       2400               0            0x0   0         4     2
+  48000      0x1    	0x3         4800       2400               0            0x0   0         5     2
+  48000      0x1    	0x3         4800       2400               0            0x0   0         11    1
+  48000      0x1    	0x3         4800       2400               0            0x0   0         12    1

diff --git a/media/libaudioclient/tests/track_test_output_v1.0_ref_walleye.txt b/media/libaudioclient/tests/track_test_output_v1.0_ref_walleye.txt
new file mode 100644
index 0000000..5fe433c
--- /dev/null
+++ b/media/libaudioclient/tests/track_test_output_v1.0_ref_walleye.txt

@@ -0,0 +1,308 @@
+
+#### Test 1 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(49), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 2 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(57), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(24000), original sample rate(24000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(1600), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (250), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 3 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(65), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(16000), original sample rate(16000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(1600), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (350), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 4 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(73), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(8000), original sample rate(8000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(1600), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (650), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 5 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(81), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(44100), original sample rate(44100), speed(1.000000)
+  frame count(4410), req. frame count(4410)
+  notif. frame count(1470), req. notif. frame count(2205), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 6 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(89), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(22050), original sample rate(22050), speed(1.000000)
+  frame count(4410), req. frame count(4410)
+  notif. frame count(1470), req. notif. frame count(2205), req. notif. per buff(0)
+  latency (250), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 7 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(97), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(11025), original sample rate(11025), speed(1.000000)
+  frame count(4410), req. frame count(4410)
+  notif. frame count(1470), req. notif. frame count(2205), req. notif. per buff(0)
+  latency (450), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 8 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(105), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(2), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 9 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(113), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(3), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (180), selected device Id(0), routed device Id(2)
+  output(29) AF latency (80) AF frame count(1920) AF SampleRate(48000)
+
+#### Test 10 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(121), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(5), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (180), selected device Id(0), routed device Id(2)
+  output(29) AF latency (80) AF frame count(1920) AF SampleRate(48000)
+
+#### Test 11 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(129), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(1), channel count(1)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 12 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(137), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3f), channel count(6)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 13 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(145), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(63f), channel count(8)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 14 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(153), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(1924), req. frame count(1924)
+  notif. frame count(962), req. notif. frame count(0), req. notif. per buff(0)
+  latency (90), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 15 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(161), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(48000), req. frame count(48000)
+  notif. frame count(24000), req. notif. frame count(0), req. notif. per buff(0)
+  latency (1050), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 16 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(169), flags(4)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(480), req. frame count(480)
+  notif. frame count(240), req. notif. frame count(0), req. notif. per buff(2)
+  latency (60), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 17 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(177), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(0), req. notif. frame count(0), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 18 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(185), flags(4)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(0), req. notif. frame count(0), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 19 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(193), flags(4)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(240), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 20 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(201), flags(8)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (180), selected device Id(0), routed device Id(2)
+  output(29) AF latency (80) AF frame count(1920) AF SampleRate(48000)
+
+#### Test 21 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(209), flags(11)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1000000), channel mask(3), channel count(2)
+  sample rate(44100), original sample rate(44100), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(4800), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (204), selected device Id(0), routed device Id(2)
+  output(53) AF latency (96) AF frame count(262144) AF SampleRate(44100)
+
+#### Test 22 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(1001), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 23 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(217), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 24 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(225), flags(0)
+  stream type(0), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (140), selected device Id(0), routed device Id(1)
+  output(45) AF latency (40) AF frame count(960) AF SampleRate(48000)
+
+#### Test 25 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(233), flags(0)
+  stream type(4), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(3)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 26 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(241), flags(0)
+  stream type(5), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(3)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 27 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(249), flags(0)
+  stream type(10), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)
+
+#### Test 28 status 0
+ AudioTrack::dump
+  status(0), state(1), session Id(257), flags(0)
+  stream type(3), left - right volume(1.000000, 1.000000)
+  format(1), channel mask(3), channel count(2)
+  sample rate(48000), original sample rate(48000), speed(1.000000)
+  frame count(4800), req. frame count(4800)
+  notif. frame count(2400), req. notif. frame count(2400), req. notif. per buff(0)
+  latency (150), selected device Id(0), routed device Id(2)
+  output(13) AF latency (50) AF frame count(960) AF SampleRate(48000)

diff --git a/media/libaudiohal/2.0/Android.bp b/media/libaudiohal/2.0/Android.bp
new file mode 100644
index 0000000..574b435
--- /dev/null
+++ b/media/libaudiohal/2.0/Android.bp

@@ -0,0 +1,54 @@
+cc_library_shared {
+    name: "libaudiohal@2.0",
+
+    srcs: [
+        "DeviceHalLocal.cpp",
+        "DevicesFactoryHalHybrid.cpp",
+        "DevicesFactoryHalLocal.cpp",
+        "StreamHalLocal.cpp",
+
+        "ConversionHelperHidl.cpp",
+        "DeviceHalHidl.cpp",
+        "DevicesFactoryHalHidl.cpp",
+        "EffectBufferHalHidl.cpp",
+        "EffectHalHidl.cpp",
+        "EffectsFactoryHalHidl.cpp",
+        "StreamHalHidl.cpp",
+    ],
+
+    export_include_dirs: ["."],
+
+    cflags: [
+        "-Wall",
+        "-Werror",
+    ],
+    shared_libs: [
+        "libaudiohal_deathhandler",
+        "libaudioutils",
+        "libcutils",
+        "liblog",
+        "libutils",
+        "libhardware",
+        "libbase",
+        "libfmq",
+        "libhwbinder",
+        "libhidlbase",
+        "libhidlmemory",
+        "libhidltransport",
+        "android.hardware.audio@2.0",
+        "android.hardware.audio.common@2.0",
+        "android.hardware.audio.common@2.0-util",
+        "android.hardware.audio.effect@2.0",
+        "android.hidl.allocator@1.0",
+        "android.hidl.memory@1.0",
+        "libmedia_helper",
+        "libmediautils",
+    ],
+    header_libs: [
+        "libaudiohal_headers"
+    ],
+
+    export_shared_lib_headers: [
+        "libfmq",
+    ],
+}

diff --git a/media/libaudiohal/ConversionHelperHidl.cpp b/media/libaudiohal/2.0/ConversionHelperHidl.cpp
similarity index 100%
rename from media/libaudiohal/ConversionHelperHidl.cpp
rename to media/libaudiohal/2.0/ConversionHelperHidl.cpp


diff --git a/media/libaudiohal/ConversionHelperHidl.h b/media/libaudiohal/2.0/ConversionHelperHidl.h
similarity index 100%
rename from media/libaudiohal/ConversionHelperHidl.h
rename to media/libaudiohal/2.0/ConversionHelperHidl.h


diff --git a/media/libaudiohal/2.0/DeviceHalHidl.cpp b/media/libaudiohal/2.0/DeviceHalHidl.cpp
new file mode 100644
index 0000000..5b99d70
--- /dev/null
+++ b/media/libaudiohal/2.0/DeviceHalHidl.cpp

@@ -0,0 +1,364 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <stdio.h>
+
+#define LOG_TAG "DeviceHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <android/hardware/audio/2.0/IPrimaryDevice.h>
+#include <cutils/native_handle.h>
+#include <hwbinder/IPCThreadState.h>
+#include <utils/Log.h>
+
+#include "DeviceHalHidl.h"
+#include "HidlUtils.h"
+#include "StreamHalHidl.h"
+
+using ::android::hardware::audio::common::V2_0::AudioConfig;
+using ::android::hardware::audio::common::V2_0::AudioDevice;
+using ::android::hardware::audio::common::V2_0::AudioInputFlag;
+using ::android::hardware::audio::common::V2_0::AudioOutputFlag;
+using ::android::hardware::audio::common::V2_0::AudioPatchHandle;
+using ::android::hardware::audio::common::V2_0::AudioPort;
+using ::android::hardware::audio::common::V2_0::AudioPortConfig;
+using ::android::hardware::audio::common::V2_0::AudioMode;
+using ::android::hardware::audio::common::V2_0::AudioSource;
+using ::android::hardware::audio::common::V2_0::HidlUtils;
+using ::android::hardware::audio::V2_0::DeviceAddress;
+using ::android::hardware::audio::V2_0::IPrimaryDevice;
+using ::android::hardware::audio::V2_0::ParameterValue;
+using ::android::hardware::audio::V2_0::Result;
+using ::android::hardware::hidl_string;
+using ::android::hardware::hidl_vec;
+
+namespace android {
+
+namespace {
+
+status_t deviceAddressFromHal(
+        audio_devices_t device, const char* halAddress, DeviceAddress* address) {
+    address->device = AudioDevice(device);
+
+    if (halAddress == nullptr || strnlen(halAddress, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) {
+        return OK;
+    }
+    const bool isInput = (device & AUDIO_DEVICE_BIT_IN) != 0;
+    if (isInput) device &= ~AUDIO_DEVICE_BIT_IN;
+    if ((!isInput && (device & AUDIO_DEVICE_OUT_ALL_A2DP) != 0)
+            || (isInput && (device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) != 0)) {
+        int status = sscanf(halAddress,
+                "%hhX:%hhX:%hhX:%hhX:%hhX:%hhX",
+                &address->address.mac[0], &address->address.mac[1], &address->address.mac[2],
+                &address->address.mac[3], &address->address.mac[4], &address->address.mac[5]);
+        return status == 6 ? OK : BAD_VALUE;
+    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_IP) != 0)
+            || (isInput && (device & AUDIO_DEVICE_IN_IP) != 0)) {
+        int status = sscanf(halAddress,
+                "%hhu.%hhu.%hhu.%hhu",
+                &address->address.ipv4[0], &address->address.ipv4[1],
+                &address->address.ipv4[2], &address->address.ipv4[3]);
+        return status == 4 ? OK : BAD_VALUE;
+    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_ALL_USB)) != 0
+            || (isInput && (device & AUDIO_DEVICE_IN_ALL_USB)) != 0) {
+        int status = sscanf(halAddress,
+                "card=%d;device=%d",
+                &address->address.alsa.card, &address->address.alsa.device);
+        return status == 2 ? OK : BAD_VALUE;
+    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_BUS) != 0)
+            || (isInput && (device & AUDIO_DEVICE_IN_BUS) != 0)) {
+        if (halAddress != NULL) {
+            address->busAddress = halAddress;
+            return OK;
+        }
+        return BAD_VALUE;
+    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) != 0
+            || (isInput && (device & AUDIO_DEVICE_IN_REMOTE_SUBMIX) != 0)) {
+        if (halAddress != NULL) {
+            address->rSubmixAddress = halAddress;
+            return OK;
+        }
+        return BAD_VALUE;
+    }
+    return OK;
+}
+
+}  // namespace
+
+DeviceHalHidl::DeviceHalHidl(const sp<IDevice>& device)
+        : ConversionHelperHidl("Device"), mDevice(device),
+          mPrimaryDevice(IPrimaryDevice::castFrom(device)) {
+}
+
+DeviceHalHidl::~DeviceHalHidl() {
+    if (mDevice != 0) {
+        mDevice.clear();
+        hardware::IPCThreadState::self()->flushCommands();
+    }
+}
+
+status_t DeviceHalHidl::getSupportedDevices(uint32_t*) {
+    // Obsolete.
+    return INVALID_OPERATION;
+}
+
+status_t DeviceHalHidl::initCheck() {
+    if (mDevice == 0) return NO_INIT;
+    return processReturn("initCheck", mDevice->initCheck());
+}
+
+status_t DeviceHalHidl::setVoiceVolume(float volume) {
+    if (mDevice == 0) return NO_INIT;
+    if (mPrimaryDevice == 0) return INVALID_OPERATION;
+    return processReturn("setVoiceVolume", mPrimaryDevice->setVoiceVolume(volume));
+}
+
+status_t DeviceHalHidl::setMasterVolume(float volume) {
+    if (mDevice == 0) return NO_INIT;
+    if (mPrimaryDevice == 0) return INVALID_OPERATION;
+    return processReturn("setMasterVolume", mPrimaryDevice->setMasterVolume(volume));
+}
+
+status_t DeviceHalHidl::getMasterVolume(float *volume) {
+    if (mDevice == 0) return NO_INIT;
+    if (mPrimaryDevice == 0) return INVALID_OPERATION;
+    Result retval;
+    Return<void> ret = mPrimaryDevice->getMasterVolume(
+            [&](Result r, float v) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *volume = v;
+                }
+            });
+    return processReturn("getMasterVolume", ret, retval);
+}
+
+status_t DeviceHalHidl::setMode(audio_mode_t mode) {
+    if (mDevice == 0) return NO_INIT;
+    if (mPrimaryDevice == 0) return INVALID_OPERATION;
+    return processReturn("setMode", mPrimaryDevice->setMode(AudioMode(mode)));
+}
+
+status_t DeviceHalHidl::setMicMute(bool state) {
+    if (mDevice == 0) return NO_INIT;
+    return processReturn("setMicMute", mDevice->setMicMute(state));
+}
+
+status_t DeviceHalHidl::getMicMute(bool *state) {
+    if (mDevice == 0) return NO_INIT;
+    Result retval;
+    Return<void> ret = mDevice->getMicMute(
+            [&](Result r, bool mute) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *state = mute;
+                }
+            });
+    return processReturn("getMicMute", ret, retval);
+}
+
+status_t DeviceHalHidl::setMasterMute(bool state) {
+    if (mDevice == 0) return NO_INIT;
+    return processReturn("setMasterMute", mDevice->setMasterMute(state));
+}
+
+status_t DeviceHalHidl::getMasterMute(bool *state) {
+    if (mDevice == 0) return NO_INIT;
+    Result retval;
+    Return<void> ret = mDevice->getMasterMute(
+            [&](Result r, bool mute) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *state = mute;
+                }
+            });
+    return processReturn("getMasterMute", ret, retval);
+}
+
+status_t DeviceHalHidl::setParameters(const String8& kvPairs) {
+    if (mDevice == 0) return NO_INIT;
+    hidl_vec<ParameterValue> hidlParams;
+    status_t status = parametersFromHal(kvPairs, &hidlParams);
+    if (status != OK) return status;
+    return processReturn("setParameters", mDevice->setParameters(hidlParams));
+}
+
+status_t DeviceHalHidl::getParameters(const String8& keys, String8 *values) {
+    values->clear();
+    if (mDevice == 0) return NO_INIT;
+    hidl_vec<hidl_string> hidlKeys;
+    status_t status = keysFromHal(keys, &hidlKeys);
+    if (status != OK) return status;
+    Result retval;
+    Return<void> ret = mDevice->getParameters(
+            hidlKeys,
+            [&](Result r, const hidl_vec<ParameterValue>& parameters) {
+                retval = r;
+                if (retval == Result::OK) {
+                    parametersToHal(parameters, values);
+                }
+            });
+    return processReturn("getParameters", ret, retval);
+}
+
+status_t DeviceHalHidl::getInputBufferSize(
+        const struct audio_config *config, size_t *size) {
+    if (mDevice == 0) return NO_INIT;
+    AudioConfig hidlConfig;
+    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
+    Result retval;
+    Return<void> ret = mDevice->getInputBufferSize(
+            hidlConfig,
+            [&](Result r, uint64_t bufferSize) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *size = static_cast<size_t>(bufferSize);
+                }
+            });
+    return processReturn("getInputBufferSize", ret, retval);
+}
+
+status_t DeviceHalHidl::openOutputStream(
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        audio_output_flags_t flags,
+        struct audio_config *config,
+        const char *address,
+        sp<StreamOutHalInterface> *outStream) {
+    if (mDevice == 0) return NO_INIT;
+    DeviceAddress hidlDevice;
+    status_t status = deviceAddressFromHal(devices, address, &hidlDevice);
+    if (status != OK) return status;
+    AudioConfig hidlConfig;
+    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mDevice->openOutputStream(
+            handle,
+            hidlDevice,
+            hidlConfig,
+            AudioOutputFlag(flags),
+            [&](Result r, const sp<IStreamOut>& result, const AudioConfig& suggestedConfig) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *outStream = new StreamOutHalHidl(result);
+                }
+                HidlUtils::audioConfigToHal(suggestedConfig, config);
+            });
+    return processReturn("openOutputStream", ret, retval);
+}
+
+status_t DeviceHalHidl::openInputStream(
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        struct audio_config *config,
+        audio_input_flags_t flags,
+        const char *address,
+        audio_source_t source,
+        sp<StreamInHalInterface> *inStream) {
+    if (mDevice == 0) return NO_INIT;
+    DeviceAddress hidlDevice;
+    status_t status = deviceAddressFromHal(devices, address, &hidlDevice);
+    if (status != OK) return status;
+    AudioConfig hidlConfig;
+    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mDevice->openInputStream(
+            handle,
+            hidlDevice,
+            hidlConfig,
+            AudioInputFlag(flags),
+            AudioSource(source),
+            [&](Result r, const sp<IStreamIn>& result, const AudioConfig& suggestedConfig) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *inStream = new StreamInHalHidl(result);
+                }
+                HidlUtils::audioConfigToHal(suggestedConfig, config);
+            });
+    return processReturn("openInputStream", ret, retval);
+}
+
+status_t DeviceHalHidl::supportsAudioPatches(bool *supportsPatches) {
+    if (mDevice == 0) return NO_INIT;
+    return processReturn("supportsAudioPatches", mDevice->supportsAudioPatches(), supportsPatches);
+}
+
+status_t DeviceHalHidl::createAudioPatch(
+        unsigned int num_sources,
+        const struct audio_port_config *sources,
+        unsigned int num_sinks,
+        const struct audio_port_config *sinks,
+        audio_patch_handle_t *patch) {
+    if (mDevice == 0) return NO_INIT;
+    hidl_vec<AudioPortConfig> hidlSources, hidlSinks;
+    HidlUtils::audioPortConfigsFromHal(num_sources, sources, &hidlSources);
+    HidlUtils::audioPortConfigsFromHal(num_sinks, sinks, &hidlSinks);
+    Result retval;
+    Return<void> ret = mDevice->createAudioPatch(
+            hidlSources, hidlSinks,
+            [&](Result r, AudioPatchHandle hidlPatch) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *patch = static_cast<audio_patch_handle_t>(hidlPatch);
+                }
+            });
+    return processReturn("createAudioPatch", ret, retval);
+}
+
+status_t DeviceHalHidl::releaseAudioPatch(audio_patch_handle_t patch) {
+    if (mDevice == 0) return NO_INIT;
+    return processReturn("releaseAudioPatch", mDevice->releaseAudioPatch(patch));
+}
+
+status_t DeviceHalHidl::getAudioPort(struct audio_port *port) {
+    if (mDevice == 0) return NO_INIT;
+    AudioPort hidlPort;
+    HidlUtils::audioPortFromHal(*port, &hidlPort);
+    Result retval;
+    Return<void> ret = mDevice->getAudioPort(
+            hidlPort,
+            [&](Result r, const AudioPort& p) {
+                retval = r;
+                if (retval == Result::OK) {
+                    HidlUtils::audioPortToHal(p, port);
+                }
+            });
+    return processReturn("getAudioPort", ret, retval);
+}
+
+status_t DeviceHalHidl::setAudioPortConfig(const struct audio_port_config *config) {
+    if (mDevice == 0) return NO_INIT;
+    AudioPortConfig hidlConfig;
+    HidlUtils::audioPortConfigFromHal(*config, &hidlConfig);
+    return processReturn("setAudioPortConfig", mDevice->setAudioPortConfig(hidlConfig));
+}
+
+status_t DeviceHalHidl::getMicrophones(
+        std::vector<media::MicrophoneInfo> *microphonesInfo __unused) {
+    if (mDevice == 0) return NO_INIT;
+    return INVALID_OPERATION;
+}
+
+status_t DeviceHalHidl::dump(int fd) {
+    if (mDevice == 0) return NO_INIT;
+    native_handle_t* hidlHandle = native_handle_create(1, 0);
+    hidlHandle->data[0] = fd;
+    Return<void> ret = mDevice->debugDump(hidlHandle);
+    native_handle_delete(hidlHandle);
+    return processReturn("dump", ret);
+}
+
+} // namespace android

diff --git a/media/libaudiohal/2.0/DeviceHalHidl.h b/media/libaudiohal/2.0/DeviceHalHidl.h
new file mode 100644
index 0000000..3c1cb59
--- /dev/null
+++ b/media/libaudiohal/2.0/DeviceHalHidl.h

@@ -0,0 +1,129 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_DEVICE_HAL_HIDL_H
+#define ANDROID_HARDWARE_DEVICE_HAL_HIDL_H
+
+#include <android/hardware/audio/2.0/IDevice.h>
+#include <android/hardware/audio/2.0/IPrimaryDevice.h>
+#include <media/audiohal/DeviceHalInterface.h>
+
+#include "ConversionHelperHidl.h"
+
+using ::android::hardware::audio::V2_0::IDevice;
+using ::android::hardware::audio::V2_0::IPrimaryDevice;
+using ::android::hardware::Return;
+
+namespace android {
+
+class DeviceHalHidl : public DeviceHalInterface, public ConversionHelperHidl
+{
+  public:
+    // Sets the value of 'devices' to a bitmask of 1 or more values of audio_devices_t.
+    virtual status_t getSupportedDevices(uint32_t *devices);
+
+    // Check to see if the audio hardware interface has been initialized.
+    virtual status_t initCheck();
+
+    // Set the audio volume of a voice call. Range is between 0.0 and 1.0.
+    virtual status_t setVoiceVolume(float volume);
+
+    // Set the audio volume for all audio activities other than voice call.
+    virtual status_t setMasterVolume(float volume);
+
+    // Get the current master volume value for the HAL.
+    virtual status_t getMasterVolume(float *volume);
+
+    // Called when the audio mode changes.
+    virtual status_t setMode(audio_mode_t mode);
+
+    // Muting control.
+    virtual status_t setMicMute(bool state);
+    virtual status_t getMicMute(bool *state);
+    virtual status_t setMasterMute(bool state);
+    virtual status_t getMasterMute(bool *state);
+
+    // Set global audio parameters.
+    virtual status_t setParameters(const String8& kvPairs);
+
+    // Get global audio parameters.
+    virtual status_t getParameters(const String8& keys, String8 *values);
+
+    // Returns audio input buffer size according to parameters passed.
+    virtual status_t getInputBufferSize(const struct audio_config *config,
+            size_t *size);
+
+    // Creates and opens the audio hardware output stream. The stream is closed
+    // by releasing all references to the returned object.
+    virtual status_t openOutputStream(
+            audio_io_handle_t handle,
+            audio_devices_t devices,
+            audio_output_flags_t flags,
+            struct audio_config *config,
+            const char *address,
+            sp<StreamOutHalInterface> *outStream);
+
+    // Creates and opens the audio hardware input stream. The stream is closed
+    // by releasing all references to the returned object.
+    virtual status_t openInputStream(
+            audio_io_handle_t handle,
+            audio_devices_t devices,
+            struct audio_config *config,
+            audio_input_flags_t flags,
+            const char *address,
+            audio_source_t source,
+            sp<StreamInHalInterface> *inStream);
+
+    // Returns whether createAudioPatch and releaseAudioPatch operations are supported.
+    virtual status_t supportsAudioPatches(bool *supportsPatches);
+
+    // Creates an audio patch between several source and sink ports.
+    virtual status_t createAudioPatch(
+            unsigned int num_sources,
+            const struct audio_port_config *sources,
+            unsigned int num_sinks,
+            const struct audio_port_config *sinks,
+            audio_patch_handle_t *patch);
+
+    // Releases an audio patch.
+    virtual status_t releaseAudioPatch(audio_patch_handle_t patch);
+
+    // Fills the list of supported attributes for a given audio port.
+    virtual status_t getAudioPort(struct audio_port *port);
+
+    // Set audio port configuration.
+    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
+
+    // List microphones
+    virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+
+    virtual status_t dump(int fd);
+
+  private:
+    friend class DevicesFactoryHalHidl;
+    sp<IDevice> mDevice;
+    sp<IPrimaryDevice> mPrimaryDevice;  // Null if it's not a primary device.
+
+    // Can not be constructed directly by clients.
+    explicit DeviceHalHidl(const sp<IDevice>& device);
+
+    // The destructor automatically closes the device.
+    virtual ~DeviceHalHidl();
+};
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_DEVICE_HAL_HIDL_H

diff --git a/media/libaudiohal/2.0/DeviceHalLocal.cpp b/media/libaudiohal/2.0/DeviceHalLocal.cpp
new file mode 100644
index 0000000..ec3bf78
--- /dev/null
+++ b/media/libaudiohal/2.0/DeviceHalLocal.cpp

@@ -0,0 +1,204 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "DeviceHalLocal"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+
+#include "DeviceHalLocal.h"
+#include "StreamHalLocal.h"
+
+namespace android {
+
+DeviceHalLocal::DeviceHalLocal(audio_hw_device_t *dev)
+        : mDev(dev) {
+}
+
+DeviceHalLocal::~DeviceHalLocal() {
+    int status = audio_hw_device_close(mDev);
+    ALOGW_IF(status, "Error closing audio hw device %p: %s", mDev, strerror(-status));
+    mDev = 0;
+}
+
+status_t DeviceHalLocal::getSupportedDevices(uint32_t *devices) {
+    if (mDev->get_supported_devices == NULL) return INVALID_OPERATION;
+    *devices = mDev->get_supported_devices(mDev);
+    return OK;
+}
+
+status_t DeviceHalLocal::initCheck() {
+    return mDev->init_check(mDev);
+}
+
+status_t DeviceHalLocal::setVoiceVolume(float volume) {
+    return mDev->set_voice_volume(mDev, volume);
+}
+
+status_t DeviceHalLocal::setMasterVolume(float volume) {
+    if (mDev->set_master_volume == NULL) return INVALID_OPERATION;
+    return mDev->set_master_volume(mDev, volume);
+}
+
+status_t DeviceHalLocal::getMasterVolume(float *volume) {
+    if (mDev->get_master_volume == NULL) return INVALID_OPERATION;
+    return mDev->get_master_volume(mDev, volume);
+}
+
+status_t DeviceHalLocal::setMode(audio_mode_t mode) {
+    return mDev->set_mode(mDev, mode);
+}
+
+status_t DeviceHalLocal::setMicMute(bool state) {
+    return mDev->set_mic_mute(mDev, state);
+}
+
+status_t DeviceHalLocal::getMicMute(bool *state) {
+    return mDev->get_mic_mute(mDev, state);
+}
+
+status_t DeviceHalLocal::setMasterMute(bool state) {
+    if (mDev->set_master_mute == NULL) return INVALID_OPERATION;
+    return mDev->set_master_mute(mDev, state);
+}
+
+status_t DeviceHalLocal::getMasterMute(bool *state) {
+    if (mDev->get_master_mute == NULL) return INVALID_OPERATION;
+    return mDev->get_master_mute(mDev, state);
+}
+
+status_t DeviceHalLocal::setParameters(const String8& kvPairs) {
+    return mDev->set_parameters(mDev, kvPairs.string());
+}
+
+status_t DeviceHalLocal::getParameters(const String8& keys, String8 *values) {
+    char *halValues = mDev->get_parameters(mDev, keys.string());
+    if (halValues != NULL) {
+        values->setTo(halValues);
+        free(halValues);
+    } else {
+        values->clear();
+    }
+    return OK;
+}
+
+status_t DeviceHalLocal::getInputBufferSize(
+        const struct audio_config *config, size_t *size) {
+    *size = mDev->get_input_buffer_size(mDev, config);
+    return OK;
+}
+
+status_t DeviceHalLocal::openOutputStream(
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        audio_output_flags_t flags,
+        struct audio_config *config,
+        const char *address,
+        sp<StreamOutHalInterface> *outStream) {
+    audio_stream_out_t *halStream;
+    ALOGV("open_output_stream handle: %d devices: %x flags: %#x"
+            "srate: %d format %#x channels %x address %s",
+            handle, devices, flags,
+            config->sample_rate, config->format, config->channel_mask,
+            address);
+    int openResut = mDev->open_output_stream(
+            mDev, handle, devices, flags, config, &halStream, address);
+    if (openResut == OK) {
+        *outStream = new StreamOutHalLocal(halStream, this);
+    }
+    ALOGV("open_output_stream status %d stream %p", openResut, halStream);
+    return openResut;
+}
+
+status_t DeviceHalLocal::openInputStream(
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        struct audio_config *config,
+        audio_input_flags_t flags,
+        const char *address,
+        audio_source_t source,
+        sp<StreamInHalInterface> *inStream) {
+    audio_stream_in_t *halStream;
+    ALOGV("open_input_stream handle: %d devices: %x flags: %#x "
+            "srate: %d format %#x channels %x address %s source %d",
+            handle, devices, flags,
+            config->sample_rate, config->format, config->channel_mask,
+            address, source);
+    int openResult = mDev->open_input_stream(
+            mDev, handle, devices, config, &halStream, flags, address, source);
+    if (openResult == OK) {
+        *inStream = new StreamInHalLocal(halStream, this);
+    }
+    ALOGV("open_input_stream status %d stream %p", openResult, inStream);
+    return openResult;
+}
+
+status_t DeviceHalLocal::supportsAudioPatches(bool *supportsPatches) {
+    *supportsPatches = version() >= AUDIO_DEVICE_API_VERSION_3_0;
+    return OK;
+}
+
+status_t DeviceHalLocal::createAudioPatch(
+        unsigned int num_sources,
+        const struct audio_port_config *sources,
+        unsigned int num_sinks,
+        const struct audio_port_config *sinks,
+        audio_patch_handle_t *patch) {
+    if (version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+        return mDev->create_audio_patch(
+                mDev, num_sources, sources, num_sinks, sinks, patch);
+    } else {
+        return INVALID_OPERATION;
+    }
+}
+
+status_t DeviceHalLocal::releaseAudioPatch(audio_patch_handle_t patch) {
+    if (version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+        return mDev->release_audio_patch(mDev, patch);
+    } else {
+        return INVALID_OPERATION;
+    }
+}
+
+status_t DeviceHalLocal::getAudioPort(struct audio_port *port) {
+    return mDev->get_audio_port(mDev, port);
+}
+
+status_t DeviceHalLocal::setAudioPortConfig(const struct audio_port_config *config) {
+    if (version() >= AUDIO_DEVICE_API_VERSION_3_0)
+        return mDev->set_audio_port_config(mDev, config);
+    else
+        return INVALID_OPERATION;
+}
+
+status_t DeviceHalLocal::getMicrophones(
+        std::vector<media::MicrophoneInfo> *microphones __unused) {
+    return INVALID_OPERATION;
+}
+
+status_t DeviceHalLocal::dump(int fd) {
+    return mDev->dump(mDev, fd);
+}
+
+void DeviceHalLocal::closeOutputStream(struct audio_stream_out *stream_out) {
+    mDev->close_output_stream(mDev, stream_out);
+}
+
+void DeviceHalLocal::closeInputStream(struct audio_stream_in *stream_in) {
+    mDev->close_input_stream(mDev, stream_in);
+}
+
+} // namespace android

diff --git a/media/libaudiohal/2.0/DeviceHalLocal.h b/media/libaudiohal/2.0/DeviceHalLocal.h
new file mode 100644
index 0000000..aec201a
--- /dev/null
+++ b/media/libaudiohal/2.0/DeviceHalLocal.h

@@ -0,0 +1,127 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_DEVICE_HAL_LOCAL_H
+#define ANDROID_HARDWARE_DEVICE_HAL_LOCAL_H
+
+#include <hardware/audio.h>
+#include <media/audiohal/DeviceHalInterface.h>
+
+namespace android {
+
+class DeviceHalLocal : public DeviceHalInterface
+{
+  public:
+    // Sets the value of 'devices' to a bitmask of 1 or more values of audio_devices_t.
+    virtual status_t getSupportedDevices(uint32_t *devices);
+
+    // Check to see if the audio hardware interface has been initialized.
+    virtual status_t initCheck();
+
+    // Set the audio volume of a voice call. Range is between 0.0 and 1.0.
+    virtual status_t setVoiceVolume(float volume);
+
+    // Set the audio volume for all audio activities other than voice call.
+    virtual status_t setMasterVolume(float volume);
+
+    // Get the current master volume value for the HAL.
+    virtual status_t getMasterVolume(float *volume);
+
+    // Called when the audio mode changes.
+    virtual status_t setMode(audio_mode_t mode);
+
+    // Muting control.
+    virtual status_t setMicMute(bool state);
+    virtual status_t getMicMute(bool *state);
+    virtual status_t setMasterMute(bool state);
+    virtual status_t getMasterMute(bool *state);
+
+    // Set global audio parameters.
+    virtual status_t setParameters(const String8& kvPairs);
+
+    // Get global audio parameters.
+    virtual status_t getParameters(const String8& keys, String8 *values);
+
+    // Returns audio input buffer size according to parameters passed.
+    virtual status_t getInputBufferSize(const struct audio_config *config,
+            size_t *size);
+
+    // Creates and opens the audio hardware output stream. The stream is closed
+    // by releasing all references to the returned object.
+    virtual status_t openOutputStream(
+            audio_io_handle_t handle,
+            audio_devices_t devices,
+            audio_output_flags_t flags,
+            struct audio_config *config,
+            const char *address,
+            sp<StreamOutHalInterface> *outStream);
+
+    // Creates and opens the audio hardware input stream. The stream is closed
+    // by releasing all references to the returned object.
+    virtual status_t openInputStream(
+            audio_io_handle_t handle,
+            audio_devices_t devices,
+            struct audio_config *config,
+            audio_input_flags_t flags,
+            const char *address,
+            audio_source_t source,
+            sp<StreamInHalInterface> *inStream);
+
+    // Returns whether createAudioPatch and releaseAudioPatch operations are supported.
+    virtual status_t supportsAudioPatches(bool *supportsPatches);
+
+    // Creates an audio patch between several source and sink ports.
+    virtual status_t createAudioPatch(
+            unsigned int num_sources,
+            const struct audio_port_config *sources,
+            unsigned int num_sinks,
+            const struct audio_port_config *sinks,
+            audio_patch_handle_t *patch);
+
+    // Releases an audio patch.
+    virtual status_t releaseAudioPatch(audio_patch_handle_t patch);
+
+    // Fills the list of supported attributes for a given audio port.
+    virtual status_t getAudioPort(struct audio_port *port);
+
+    // Set audio port configuration.
+    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
+
+    // List microphones
+    virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+
+    virtual status_t dump(int fd);
+
+    void closeOutputStream(struct audio_stream_out *stream_out);
+    void closeInputStream(struct audio_stream_in *stream_in);
+
+  private:
+    audio_hw_device_t *mDev;
+
+    friend class DevicesFactoryHalLocal;
+
+    // Can not be constructed directly by clients.
+    explicit DeviceHalLocal(audio_hw_device_t *dev);
+
+    // The destructor automatically closes the device.
+    virtual ~DeviceHalLocal();
+
+    uint32_t version() const { return mDev->common.version; }
+};
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_DEVICE_HAL_LOCAL_H

diff --git a/media/libaudiohal/2.0/DevicesFactoryHalHidl.cpp b/media/libaudiohal/2.0/DevicesFactoryHalHidl.cpp
new file mode 100644
index 0000000..5b33592
--- /dev/null
+++ b/media/libaudiohal/2.0/DevicesFactoryHalHidl.cpp

@@ -0,0 +1,98 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <string.h>
+
+#define LOG_TAG "DevicesFactoryHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <android/hardware/audio/2.0/IDevice.h>
+#include <media/audiohal/hidl/HalDeathHandler.h>
+#include <utils/Log.h>
+
+#include "ConversionHelperHidl.h"
+#include "DeviceHalHidl.h"
+#include "DevicesFactoryHalHidl.h"
+
+using ::android::hardware::audio::V2_0::IDevice;
+using ::android::hardware::audio::V2_0::Result;
+using ::android::hardware::Return;
+
+namespace android {
+
+DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
+    mDevicesFactory = IDevicesFactory::getService();
+    if (mDevicesFactory != 0) {
+        // It is assumed that DevicesFactory is owned by AudioFlinger
+        // and thus have the same lifespan.
+        mDevicesFactory->linkToDeath(HalDeathHandler::getInstance(), 0 /*cookie*/);
+    } else {
+        ALOGE("Failed to obtain IDevicesFactory service, terminating process.");
+        exit(1);
+    }
+    // The MSD factory is optional
+    mDevicesFactoryMsd = IDevicesFactory::getService(AUDIO_HAL_SERVICE_NAME_MSD);
+    // TODO: Register death handler, and add 'restart' directive to audioserver.rc
+}
+
+DevicesFactoryHalHidl::~DevicesFactoryHalHidl() {
+}
+
+// static
+status_t DevicesFactoryHalHidl::nameFromHal(const char *name, IDevicesFactory::Device *device) {
+    if (strcmp(name, AUDIO_HARDWARE_MODULE_ID_PRIMARY) == 0) {
+        *device = IDevicesFactory::Device::PRIMARY;
+        return OK;
+    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) {
+        *device = IDevicesFactory::Device::A2DP;
+        return OK;
+    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) {
+        *device = IDevicesFactory::Device::USB;
+        return OK;
+    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) {
+        *device = IDevicesFactory::Device::R_SUBMIX;
+        return OK;
+    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_STUB) == 0) {
+        *device = IDevicesFactory::Device::STUB;
+        return OK;
+    }
+    ALOGE("Invalid device name %s", name);
+    return BAD_VALUE;
+}
+
+status_t DevicesFactoryHalHidl::openDevice(const char *name, sp<DeviceHalInterface> *device) {
+    if (mDevicesFactory == 0) return NO_INIT;
+    IDevicesFactory::Device hidlDevice;
+    status_t status = nameFromHal(name, &hidlDevice);
+    if (status != OK) return status;
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mDevicesFactory->openDevice(
+            hidlDevice,
+            [&](Result r, const sp<IDevice>& result) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *device = new DeviceHalHidl(result);
+                }
+            });
+    if (ret.isOk()) {
+        if (retval == Result::OK) return OK;
+        else if (retval == Result::INVALID_ARGUMENTS) return BAD_VALUE;
+        else return NO_INIT;
+    }
+    return FAILED_TRANSACTION;
+}
+
+} // namespace android

diff --git a/media/libaudiohal/2.0/DevicesFactoryHalHidl.h b/media/libaudiohal/2.0/DevicesFactoryHalHidl.h
new file mode 100644
index 0000000..0748849
--- /dev/null
+++ b/media/libaudiohal/2.0/DevicesFactoryHalHidl.h

@@ -0,0 +1,54 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_H
+#define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_H
+
+#include <android/hardware/audio/2.0/IDevicesFactory.h>
+#include <media/audiohal/DevicesFactoryHalInterface.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+#include "DeviceHalHidl.h"
+
+using ::android::hardware::audio::V2_0::IDevicesFactory;
+
+namespace android {
+
+class DevicesFactoryHalHidl : public DevicesFactoryHalInterface
+{
+  public:
+    // Opens a device with the specified name. To close the device, it is
+    // necessary to release references to the returned object.
+    virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
+
+  private:
+    friend class DevicesFactoryHalHybrid;
+
+    sp<IDevicesFactory> mDevicesFactory;
+    sp<IDevicesFactory> mDevicesFactoryMsd;
+
+    static status_t nameFromHal(const char *name, IDevicesFactory::Device *device);
+
+    // Can not be constructed directly by clients.
+    DevicesFactoryHalHidl();
+
+    virtual ~DevicesFactoryHalHidl();
+};
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_H

diff --git a/media/libaudiohal/2.0/DevicesFactoryHalHybrid.cpp b/media/libaudiohal/2.0/DevicesFactoryHalHybrid.cpp
new file mode 100644
index 0000000..1c4be74
--- /dev/null
+++ b/media/libaudiohal/2.0/DevicesFactoryHalHybrid.cpp

@@ -0,0 +1,42 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "DevicesFactoryHalHybrid"
+//#define LOG_NDEBUG 0
+
+#include "DevicesFactoryHalHybrid.h"
+#include "DevicesFactoryHalLocal.h"
+#include "DevicesFactoryHalHidl.h"
+
+namespace android {
+
+DevicesFactoryHalHybrid::DevicesFactoryHalHybrid()
+        : mLocalFactory(new DevicesFactoryHalLocal()),
+          mHidlFactory(new DevicesFactoryHalHidl()) {
+}
+
+DevicesFactoryHalHybrid::~DevicesFactoryHalHybrid() {
+}
+
+status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
+    if (mHidlFactory != 0 && strcmp(AUDIO_HARDWARE_MODULE_ID_A2DP, name) != 0 &&
+        strcmp(AUDIO_HARDWARE_MODULE_ID_HEARING_AID, name) != 0) {
+        return mHidlFactory->openDevice(name, device);
+    }
+    return mLocalFactory->openDevice(name, device);
+}
+
+} // namespace android

diff --git a/media/libaudiohal/DevicesFactoryHalHybrid.h b/media/libaudiohal/2.0/DevicesFactoryHalHybrid.h
similarity index 100%
rename from media/libaudiohal/DevicesFactoryHalHybrid.h
rename to media/libaudiohal/2.0/DevicesFactoryHalHybrid.h


diff --git a/media/libaudiohal/DevicesFactoryHalLocal.cpp b/media/libaudiohal/2.0/DevicesFactoryHalLocal.cpp
similarity index 100%
rename from media/libaudiohal/DevicesFactoryHalLocal.cpp
rename to media/libaudiohal/2.0/DevicesFactoryHalLocal.cpp


diff --git a/media/libaudiohal/DevicesFactoryHalLocal.h b/media/libaudiohal/2.0/DevicesFactoryHalLocal.h
similarity index 100%
rename from media/libaudiohal/DevicesFactoryHalLocal.h
rename to media/libaudiohal/2.0/DevicesFactoryHalLocal.h


diff --git a/media/libaudiohal/2.0/EffectBufferHalHidl.cpp b/media/libaudiohal/2.0/EffectBufferHalHidl.cpp
new file mode 100644
index 0000000..226a500
--- /dev/null
+++ b/media/libaudiohal/2.0/EffectBufferHalHidl.cpp

@@ -0,0 +1,144 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <atomic>
+
+#define LOG_TAG "EffectBufferHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <android/hidl/allocator/1.0/IAllocator.h>
+#include <hidlmemory/mapping.h>
+#include <utils/Log.h>
+
+#include "ConversionHelperHidl.h"
+#include "EffectBufferHalHidl.h"
+
+using ::android::hardware::Return;
+using ::android::hidl::allocator::V1_0::IAllocator;
+
+namespace android {
+
+// static
+uint64_t EffectBufferHalHidl::makeUniqueId() {
+    static std::atomic<uint64_t> counter{1};
+    return counter++;
+}
+
+status_t EffectBufferHalHidl::allocate(
+        size_t size, sp<EffectBufferHalInterface>* buffer) {
+    return mirror(nullptr, size, buffer);
+}
+
+status_t EffectBufferHalHidl::mirror(
+        void* external, size_t size, sp<EffectBufferHalInterface>* buffer) {
+    sp<EffectBufferHalInterface> tempBuffer = new EffectBufferHalHidl(size);
+    status_t result = static_cast<EffectBufferHalHidl*>(tempBuffer.get())->init();
+    if (result == OK) {
+        tempBuffer->setExternalData(external);
+        *buffer = tempBuffer;
+    }
+    return result;
+}
+
+EffectBufferHalHidl::EffectBufferHalHidl(size_t size)
+        : mBufferSize(size), mFrameCountChanged(false),
+          mExternalData(nullptr), mAudioBuffer{0, {nullptr}} {
+    mHidlBuffer.id = makeUniqueId();
+    mHidlBuffer.frameCount = 0;
+}
+
+EffectBufferHalHidl::~EffectBufferHalHidl() {
+}
+
+status_t EffectBufferHalHidl::init() {
+    sp<IAllocator> ashmem = IAllocator::getService("ashmem");
+    if (ashmem == 0) {
+        ALOGE("Failed to retrieve ashmem allocator service");
+        return NO_INIT;
+    }
+    status_t retval = NO_MEMORY;
+    Return<void> result = ashmem->allocate(
+            mBufferSize,
+            [&](bool success, const hidl_memory& memory) {
+                if (success) {
+                    mHidlBuffer.data = memory;
+                    retval = OK;
+                }
+            });
+    if (result.isOk() && retval == OK) {
+        mMemory = hardware::mapMemory(mHidlBuffer.data);
+        if (mMemory != 0) {
+            mMemory->update();
+            mAudioBuffer.raw = static_cast<void*>(mMemory->getPointer());
+            memset(mAudioBuffer.raw, 0, mMemory->getSize());
+            mMemory->commit();
+        } else {
+            ALOGE("Failed to map allocated ashmem");
+            retval = NO_MEMORY;
+        }
+    } else {
+        ALOGE("Failed to allocate %d bytes from ashmem", (int)mBufferSize);
+    }
+    return result.isOk() ? retval : FAILED_TRANSACTION;
+}
+
+audio_buffer_t* EffectBufferHalHidl::audioBuffer() {
+    return &mAudioBuffer;
+}
+
+void* EffectBufferHalHidl::externalData() const {
+    return mExternalData;
+}
+
+void EffectBufferHalHidl::setFrameCount(size_t frameCount) {
+    mHidlBuffer.frameCount = frameCount;
+    mAudioBuffer.frameCount = frameCount;
+    mFrameCountChanged = true;
+}
+
+bool EffectBufferHalHidl::checkFrameCountChange() {
+    bool result = mFrameCountChanged;
+    mFrameCountChanged = false;
+    return result;
+}
+
+void EffectBufferHalHidl::setExternalData(void* external) {
+    mExternalData = external;
+}
+
+void EffectBufferHalHidl::update() {
+    update(mBufferSize);
+}
+
+void EffectBufferHalHidl::commit() {
+    commit(mBufferSize);
+}
+
+void EffectBufferHalHidl::update(size_t size) {
+    if (mExternalData == nullptr) return;
+    mMemory->update();
+    if (size > mBufferSize) size = mBufferSize;
+    memcpy(mAudioBuffer.raw, mExternalData, size);
+    mMemory->commit();
+}
+
+void EffectBufferHalHidl::commit(size_t size) {
+    if (mExternalData == nullptr) return;
+    if (size > mBufferSize) size = mBufferSize;
+    memcpy(mExternalData, mAudioBuffer.raw, size);
+}
+
+} // namespace android

diff --git a/media/libaudiohal/2.0/EffectBufferHalHidl.h b/media/libaudiohal/2.0/EffectBufferHalHidl.h
new file mode 100644
index 0000000..31e0087
--- /dev/null
+++ b/media/libaudiohal/2.0/EffectBufferHalHidl.h

@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_H
+#define ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_H
+
+#include <android/hardware/audio/effect/2.0/types.h>
+#include <android/hidl/memory/1.0/IMemory.h>
+#include <hidl/HidlSupport.h>
+#include <media/audiohal/EffectBufferHalInterface.h>
+#include <system/audio_effect.h>
+
+using android::hardware::audio::effect::V2_0::AudioBuffer;
+using android::hardware::hidl_memory;
+using android::hidl::memory::V1_0::IMemory;
+
+namespace android {
+
+class EffectBufferHalHidl : public EffectBufferHalInterface
+{
+  public:
+    static status_t allocate(size_t size, sp<EffectBufferHalInterface>* buffer);
+    static status_t mirror(void* external, size_t size, sp<EffectBufferHalInterface>* buffer);
+
+    virtual audio_buffer_t* audioBuffer();
+    virtual void* externalData() const;
+
+    virtual size_t getSize() const override { return mBufferSize; }
+
+    virtual void setExternalData(void* external);
+    virtual void setFrameCount(size_t frameCount);
+    virtual bool checkFrameCountChange();
+
+    virtual void update();
+    virtual void commit();
+    virtual void update(size_t size);
+    virtual void commit(size_t size);
+
+    const AudioBuffer& hidlBuffer() const { return mHidlBuffer; }
+
+  private:
+    friend class EffectBufferHalInterface;
+
+    static uint64_t makeUniqueId();
+
+    const size_t mBufferSize;
+    bool mFrameCountChanged;
+    void* mExternalData;
+    AudioBuffer mHidlBuffer;
+    sp<IMemory> mMemory;
+    audio_buffer_t mAudioBuffer;
+
+    // Can not be constructed directly by clients.
+    explicit EffectBufferHalHidl(size_t size);
+
+    virtual ~EffectBufferHalHidl();
+
+    status_t init();
+};
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_H

diff --git a/media/libaudiohal/2.0/EffectHalHidl.cpp b/media/libaudiohal/2.0/EffectHalHidl.cpp
new file mode 100644
index 0000000..4fb032c
--- /dev/null
+++ b/media/libaudiohal/2.0/EffectHalHidl.cpp

@@ -0,0 +1,338 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "EffectHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <hwbinder/IPCThreadState.h>
+#include <media/EffectsFactoryApi.h>
+#include <utils/Log.h>
+
+#include "ConversionHelperHidl.h"
+#include "EffectBufferHalHidl.h"
+#include "EffectHalHidl.h"
+#include "HidlUtils.h"
+
+using ::android::hardware::audio::effect::V2_0::AudioBuffer;
+using ::android::hardware::audio::effect::V2_0::EffectBufferAccess;
+using ::android::hardware::audio::effect::V2_0::EffectConfigParameters;
+using ::android::hardware::audio::effect::V2_0::MessageQueueFlagBits;
+using ::android::hardware::audio::effect::V2_0::Result;
+using ::android::hardware::audio::common::V2_0::HidlUtils;
+using ::android::hardware::audio::common::V2_0::AudioChannelMask;
+using ::android::hardware::audio::common::V2_0::AudioFormat;
+using ::android::hardware::hidl_vec;
+using ::android::hardware::MQDescriptorSync;
+using ::android::hardware::Return;
+
+namespace android {
+
+EffectHalHidl::EffectHalHidl(const sp<IEffect>& effect, uint64_t effectId)
+        : mEffect(effect), mEffectId(effectId), mBuffersChanged(true), mEfGroup(nullptr) {
+}
+
+EffectHalHidl::~EffectHalHidl() {
+    if (mEffect != 0) {
+        close();
+        mEffect.clear();
+        hardware::IPCThreadState::self()->flushCommands();
+    }
+    if (mEfGroup) {
+        EventFlag::deleteEventFlag(&mEfGroup);
+    }
+}
+
+// static
+void EffectHalHidl::effectDescriptorToHal(
+        const EffectDescriptor& descriptor, effect_descriptor_t* halDescriptor) {
+    HidlUtils::uuidToHal(descriptor.type, &halDescriptor->type);
+    HidlUtils::uuidToHal(descriptor.uuid, &halDescriptor->uuid);
+    halDescriptor->flags = static_cast<uint32_t>(descriptor.flags);
+    halDescriptor->cpuLoad = descriptor.cpuLoad;
+    halDescriptor->memoryUsage = descriptor.memoryUsage;
+    memcpy(halDescriptor->name, descriptor.name.data(), descriptor.name.size());
+    memcpy(halDescriptor->implementor,
+            descriptor.implementor.data(), descriptor.implementor.size());
+}
+
+// TODO(mnaganov): These buffer conversion functions should be shared with Effect wrapper
+// via HidlUtils. Move them there when hardware/interfaces will get un-frozen again.
+
+// static
+void EffectHalHidl::effectBufferConfigFromHal(
+        const buffer_config_t& halConfig, EffectBufferConfig* config) {
+    config->samplingRateHz = halConfig.samplingRate;
+    config->channels = AudioChannelMask(halConfig.channels);
+    config->format = AudioFormat(halConfig.format);
+    config->accessMode = EffectBufferAccess(halConfig.accessMode);
+    config->mask = EffectConfigParameters(halConfig.mask);
+}
+
+// static
+void EffectHalHidl::effectBufferConfigToHal(
+        const EffectBufferConfig& config, buffer_config_t* halConfig) {
+    halConfig->buffer.frameCount = 0;
+    halConfig->buffer.raw = NULL;
+    halConfig->samplingRate = config.samplingRateHz;
+    halConfig->channels = static_cast<uint32_t>(config.channels);
+    halConfig->bufferProvider.cookie = NULL;
+    halConfig->bufferProvider.getBuffer = NULL;
+    halConfig->bufferProvider.releaseBuffer = NULL;
+    halConfig->format = static_cast<uint8_t>(config.format);
+    halConfig->accessMode = static_cast<uint8_t>(config.accessMode);
+    halConfig->mask = static_cast<uint8_t>(config.mask);
+}
+
+// static
+void EffectHalHidl::effectConfigFromHal(const effect_config_t& halConfig, EffectConfig* config) {
+    effectBufferConfigFromHal(halConfig.inputCfg, &config->inputCfg);
+    effectBufferConfigFromHal(halConfig.outputCfg, &config->outputCfg);
+}
+
+// static
+void EffectHalHidl::effectConfigToHal(const EffectConfig& config, effect_config_t* halConfig) {
+    effectBufferConfigToHal(config.inputCfg, &halConfig->inputCfg);
+    effectBufferConfigToHal(config.outputCfg, &halConfig->outputCfg);
+}
+
+// static
+status_t EffectHalHidl::analyzeResult(const Result& result) {
+    switch (result) {
+        case Result::OK: return OK;
+        case Result::INVALID_ARGUMENTS: return BAD_VALUE;
+        case Result::INVALID_STATE: return NOT_ENOUGH_DATA;
+        case Result::NOT_INITIALIZED: return NO_INIT;
+        case Result::NOT_SUPPORTED: return INVALID_OPERATION;
+        case Result::RESULT_TOO_BIG: return NO_MEMORY;
+        default: return NO_INIT;
+    }
+}
+
+status_t EffectHalHidl::setInBuffer(const sp<EffectBufferHalInterface>& buffer) {
+    if (!mBuffersChanged) {
+        if (buffer.get() == nullptr || mInBuffer.get() == nullptr) {
+            mBuffersChanged = buffer.get() != mInBuffer.get();
+        } else {
+            mBuffersChanged = buffer->audioBuffer() != mInBuffer->audioBuffer();
+        }
+    }
+    mInBuffer = buffer;
+    return OK;
+}
+
+status_t EffectHalHidl::setOutBuffer(const sp<EffectBufferHalInterface>& buffer) {
+    if (!mBuffersChanged) {
+        if (buffer.get() == nullptr || mOutBuffer.get() == nullptr) {
+            mBuffersChanged = buffer.get() != mOutBuffer.get();
+        } else {
+            mBuffersChanged = buffer->audioBuffer() != mOutBuffer->audioBuffer();
+        }
+    }
+    mOutBuffer = buffer;
+    return OK;
+}
+
+status_t EffectHalHidl::process() {
+    return processImpl(static_cast<uint32_t>(MessageQueueFlagBits::REQUEST_PROCESS));
+}
+
+status_t EffectHalHidl::processReverse() {
+    return processImpl(static_cast<uint32_t>(MessageQueueFlagBits::REQUEST_PROCESS_REVERSE));
+}
+
+status_t EffectHalHidl::prepareForProcessing() {
+    std::unique_ptr<StatusMQ> tempStatusMQ;
+    Result retval;
+    Return<void> ret = mEffect->prepareForProcessing(
+            [&](Result r, const MQDescriptorSync<Result>& statusMQ) {
+                retval = r;
+                if (retval == Result::OK) {
+                    tempStatusMQ.reset(new StatusMQ(statusMQ));
+                    if (tempStatusMQ->isValid() && tempStatusMQ->getEventFlagWord()) {
+                        EventFlag::createEventFlag(tempStatusMQ->getEventFlagWord(), &mEfGroup);
+                    }
+                }
+            });
+    if (!ret.isOk() || retval != Result::OK) {
+        return ret.isOk() ? analyzeResult(retval) : FAILED_TRANSACTION;
+    }
+    if (!tempStatusMQ || !tempStatusMQ->isValid() || !mEfGroup) {
+        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for effects");
+        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
+                "Status message queue for effects is invalid");
+        ALOGE_IF(!mEfGroup, "Event flag creation for effects failed");
+        return NO_INIT;
+    }
+    mStatusMQ = std::move(tempStatusMQ);
+    return OK;
+}
+
+bool EffectHalHidl::needToResetBuffers() {
+    if (mBuffersChanged) return true;
+    bool inBufferFrameCountUpdated = mInBuffer->checkFrameCountChange();
+    bool outBufferFrameCountUpdated = mOutBuffer->checkFrameCountChange();
+    return inBufferFrameCountUpdated || outBufferFrameCountUpdated;
+}
+
+status_t EffectHalHidl::processImpl(uint32_t mqFlag) {
+    if (mEffect == 0 || mInBuffer == 0 || mOutBuffer == 0) return NO_INIT;
+    status_t status;
+    if (!mStatusMQ && (status = prepareForProcessing()) != OK) {
+        return status;
+    }
+    if (needToResetBuffers() && (status = setProcessBuffers()) != OK) {
+        return status;
+    }
+    // The data is already in the buffers, just need to flush it and wake up the server side.
+    std::atomic_thread_fence(std::memory_order_release);
+    mEfGroup->wake(mqFlag);
+    uint32_t efState = 0;
+retry:
+    status_t ret = mEfGroup->wait(
+            static_cast<uint32_t>(MessageQueueFlagBits::DONE_PROCESSING), &efState);
+    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::DONE_PROCESSING)) {
+        Result retval = Result::NOT_INITIALIZED;
+        mStatusMQ->read(&retval);
+        if (retval == Result::OK || retval == Result::INVALID_STATE) {
+            // Sync back the changed contents of the buffer.
+            std::atomic_thread_fence(std::memory_order_acquire);
+        }
+        return analyzeResult(retval);
+    }
+    if (ret == -EAGAIN || ret == -EINTR) {
+        // Spurious wakeup. This normally retries no more than once.
+        goto retry;
+    }
+    return ret;
+}
+
+status_t EffectHalHidl::setProcessBuffers() {
+    Return<Result> ret = mEffect->setProcessBuffers(
+            static_cast<EffectBufferHalHidl*>(mInBuffer.get())->hidlBuffer(),
+            static_cast<EffectBufferHalHidl*>(mOutBuffer.get())->hidlBuffer());
+    if (ret.isOk() && ret == Result::OK) {
+        mBuffersChanged = false;
+        return OK;
+    }
+    return ret.isOk() ? analyzeResult(ret) : FAILED_TRANSACTION;
+}
+
+status_t EffectHalHidl::command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData,
+        uint32_t *replySize, void *pReplyData) {
+    if (mEffect == 0) return NO_INIT;
+
+    // Special cases.
+    if (cmdCode == EFFECT_CMD_SET_CONFIG || cmdCode == EFFECT_CMD_SET_CONFIG_REVERSE) {
+        return setConfigImpl(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
+    } else if (cmdCode == EFFECT_CMD_GET_CONFIG || cmdCode == EFFECT_CMD_GET_CONFIG_REVERSE) {
+        return getConfigImpl(cmdCode, replySize, pReplyData);
+    }
+
+    // Common case.
+    hidl_vec<uint8_t> hidlData;
+    if (pCmdData != nullptr && cmdSize > 0) {
+        hidlData.setToExternal(reinterpret_cast<uint8_t*>(pCmdData), cmdSize);
+    }
+    status_t status;
+    uint32_t replySizeStub = 0;
+    if (replySize == nullptr || pReplyData == nullptr) replySize = &replySizeStub;
+    Return<void> ret = mEffect->command(cmdCode, hidlData, *replySize,
+            [&](int32_t s, const hidl_vec<uint8_t>& result) {
+                status = s;
+                if (status == 0) {
+                    if (*replySize > result.size()) *replySize = result.size();
+                    if (pReplyData != nullptr && *replySize > 0) {
+                        memcpy(pReplyData, &result[0], *replySize);
+                    }
+                }
+            });
+    return ret.isOk() ? status : FAILED_TRANSACTION;
+}
+
+status_t EffectHalHidl::getDescriptor(effect_descriptor_t *pDescriptor) {
+    if (mEffect == 0) return NO_INIT;
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mEffect->getDescriptor(
+            [&](Result r, const EffectDescriptor& result) {
+                retval = r;
+                if (retval == Result::OK) {
+                    effectDescriptorToHal(result, pDescriptor);
+                }
+            });
+    return ret.isOk() ? analyzeResult(retval) : FAILED_TRANSACTION;
+}
+
+status_t EffectHalHidl::close() {
+    if (mEffect == 0) return NO_INIT;
+    Return<Result> ret = mEffect->close();
+    return ret.isOk() ? analyzeResult(ret) : FAILED_TRANSACTION;
+}
+
+status_t EffectHalHidl::getConfigImpl(
+        uint32_t cmdCode, uint32_t *replySize, void *pReplyData) {
+    if (replySize == NULL || *replySize != sizeof(effect_config_t) || pReplyData == NULL) {
+        return BAD_VALUE;
+    }
+    status_t result = FAILED_TRANSACTION;
+    Return<void> ret;
+    if (cmdCode == EFFECT_CMD_GET_CONFIG) {
+        ret = mEffect->getConfig([&] (Result r, const EffectConfig &hidlConfig) {
+            result = analyzeResult(r);
+            if (r == Result::OK) {
+                effectConfigToHal(hidlConfig, static_cast<effect_config_t*>(pReplyData));
+            }
+        });
+    } else {
+        ret = mEffect->getConfigReverse([&] (Result r, const EffectConfig &hidlConfig) {
+            result = analyzeResult(r);
+            if (r == Result::OK) {
+                effectConfigToHal(hidlConfig, static_cast<effect_config_t*>(pReplyData));
+            }
+        });
+    }
+    if (!ret.isOk()) {
+        result = FAILED_TRANSACTION;
+    }
+    return result;
+}
+
+status_t EffectHalHidl::setConfigImpl(
+        uint32_t cmdCode, uint32_t cmdSize, void *pCmdData, uint32_t *replySize, void *pReplyData) {
+    if (pCmdData == NULL || cmdSize != sizeof(effect_config_t) ||
+            replySize == NULL || *replySize != sizeof(int32_t) || pReplyData == NULL) {
+        return BAD_VALUE;
+    }
+    const effect_config_t *halConfig = static_cast<effect_config_t*>(pCmdData);
+    if (halConfig->inputCfg.bufferProvider.getBuffer != NULL ||
+            halConfig->inputCfg.bufferProvider.releaseBuffer != NULL ||
+            halConfig->outputCfg.bufferProvider.getBuffer != NULL ||
+            halConfig->outputCfg.bufferProvider.releaseBuffer != NULL) {
+        ALOGE("Buffer provider callbacks are not supported");
+    }
+    EffectConfig hidlConfig;
+    effectConfigFromHal(*halConfig, &hidlConfig);
+    Return<Result> ret = cmdCode == EFFECT_CMD_SET_CONFIG ?
+            mEffect->setConfig(hidlConfig, nullptr, nullptr) :
+            mEffect->setConfigReverse(hidlConfig, nullptr, nullptr);
+    status_t result = FAILED_TRANSACTION;
+    if (ret.isOk()) {
+        result = analyzeResult(ret);
+        *static_cast<int32_t*>(pReplyData) = result;
+    }
+    return result;
+}
+
+} // namespace android

diff --git a/media/libaudiohal/EffectHalHidl.h b/media/libaudiohal/2.0/EffectHalHidl.h
similarity index 100%
rename from media/libaudiohal/EffectHalHidl.h
rename to media/libaudiohal/2.0/EffectHalHidl.h


diff --git a/media/libaudiohal/2.0/EffectsFactoryHalHidl.cpp b/media/libaudiohal/2.0/EffectsFactoryHalHidl.cpp
new file mode 100644
index 0000000..0d40e6d
--- /dev/null
+++ b/media/libaudiohal/2.0/EffectsFactoryHalHidl.cpp

@@ -0,0 +1,150 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "EffectsFactoryHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <cutils/native_handle.h>
+
+#include "ConversionHelperHidl.h"
+#include "EffectBufferHalHidl.h"
+#include "EffectHalHidl.h"
+#include "EffectsFactoryHalHidl.h"
+#include "HidlUtils.h"
+
+using ::android::hardware::audio::common::V2_0::HidlUtils;
+using ::android::hardware::audio::common::V2_0::Uuid;
+using ::android::hardware::audio::effect::V2_0::IEffect;
+using ::android::hardware::audio::effect::V2_0::Result;
+using ::android::hardware::Return;
+
+namespace android {
+
+EffectsFactoryHalHidl::EffectsFactoryHalHidl() : ConversionHelperHidl("EffectsFactory") {
+    mEffectsFactory = IEffectsFactory::getService();
+    if (mEffectsFactory == 0) {
+        ALOGE("Failed to obtain IEffectsFactory service, terminating process.");
+        exit(1);
+    }
+}
+
+EffectsFactoryHalHidl::~EffectsFactoryHalHidl() {
+}
+
+status_t EffectsFactoryHalHidl::queryAllDescriptors() {
+    if (mEffectsFactory == 0) return NO_INIT;
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mEffectsFactory->getAllDescriptors(
+            [&](Result r, const hidl_vec<EffectDescriptor>& result) {
+                retval = r;
+                if (retval == Result::OK) {
+                    mLastDescriptors = result;
+                }
+            });
+    if (ret.isOk()) {
+        return retval == Result::OK ? OK : NO_INIT;
+    }
+    mLastDescriptors.resize(0);
+    return processReturn(__FUNCTION__, ret);
+}
+
+status_t EffectsFactoryHalHidl::queryNumberEffects(uint32_t *pNumEffects) {
+    status_t queryResult = queryAllDescriptors();
+    if (queryResult == OK) {
+        *pNumEffects = mLastDescriptors.size();
+    }
+    return queryResult;
+}
+
+status_t EffectsFactoryHalHidl::getDescriptor(
+        uint32_t index, effect_descriptor_t *pDescriptor) {
+    // TODO: We need somehow to track the changes on the server side
+    // or figure out how to convert everybody to query all the descriptors at once.
+    // TODO: check for nullptr
+    if (mLastDescriptors.size() == 0) {
+        status_t queryResult = queryAllDescriptors();
+        if (queryResult != OK) return queryResult;
+    }
+    if (index >= mLastDescriptors.size()) return NAME_NOT_FOUND;
+    EffectHalHidl::effectDescriptorToHal(mLastDescriptors[index], pDescriptor);
+    return OK;
+}
+
+status_t EffectsFactoryHalHidl::getDescriptor(
+        const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor) {
+    // TODO: check for nullptr
+    if (mEffectsFactory == 0) return NO_INIT;
+    Uuid hidlUuid;
+    HidlUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mEffectsFactory->getDescriptor(hidlUuid,
+            [&](Result r, const EffectDescriptor& result) {
+                retval = r;
+                if (retval == Result::OK) {
+                    EffectHalHidl::effectDescriptorToHal(result, pDescriptor);
+                }
+            });
+    if (ret.isOk()) {
+        if (retval == Result::OK) return OK;
+        else if (retval == Result::INVALID_ARGUMENTS) return NAME_NOT_FOUND;
+        else return NO_INIT;
+    }
+    return processReturn(__FUNCTION__, ret);
+}
+
+status_t EffectsFactoryHalHidl::createEffect(
+        const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId,
+        sp<EffectHalInterface> *effect) {
+    if (mEffectsFactory == 0) return NO_INIT;
+    Uuid hidlUuid;
+    HidlUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mEffectsFactory->createEffect(
+            hidlUuid, sessionId, ioId,
+            [&](Result r, const sp<IEffect>& result, uint64_t effectId) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *effect = new EffectHalHidl(result, effectId);
+                }
+            });
+    if (ret.isOk()) {
+        if (retval == Result::OK) return OK;
+        else if (retval == Result::INVALID_ARGUMENTS) return NAME_NOT_FOUND;
+        else return NO_INIT;
+    }
+    return processReturn(__FUNCTION__, ret);
+}
+
+status_t EffectsFactoryHalHidl::dumpEffects(int fd) {
+    if (mEffectsFactory == 0) return NO_INIT;
+    native_handle_t* hidlHandle = native_handle_create(1, 0);
+    hidlHandle->data[0] = fd;
+    Return<void> ret = mEffectsFactory->debugDump(hidlHandle);
+    native_handle_delete(hidlHandle);
+    return processReturn(__FUNCTION__, ret);
+}
+
+status_t EffectsFactoryHalHidl::allocateBuffer(size_t size, sp<EffectBufferHalInterface>* buffer) {
+    return EffectBufferHalHidl::allocate(size, buffer);
+}
+
+status_t EffectsFactoryHalHidl::mirrorBuffer(void* external, size_t size,
+                          sp<EffectBufferHalInterface>* buffer) {
+    return EffectBufferHalHidl::mirror(external, size, buffer);
+}
+
+
+} // namespace android

diff --git a/media/libaudiohal/2.0/EffectsFactoryHalHidl.h b/media/libaudiohal/2.0/EffectsFactoryHalHidl.h
new file mode 100644
index 0000000..82b5481
--- /dev/null
+++ b/media/libaudiohal/2.0/EffectsFactoryHalHidl.h

@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H
+#define ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H
+
+#include <android/hardware/audio/effect/2.0/IEffectsFactory.h>
+#include <android/hardware/audio/effect/2.0/types.h>
+#include <media/audiohal/EffectsFactoryHalInterface.h>
+
+#include "ConversionHelperHidl.h"
+
+namespace android {
+
+using ::android::hardware::audio::effect::V2_0::EffectDescriptor;
+using ::android::hardware::audio::effect::V2_0::IEffectsFactory;
+using ::android::hardware::hidl_vec;
+
+class EffectsFactoryHalHidl : public EffectsFactoryHalInterface, public ConversionHelperHidl
+{
+  public:
+    // Returns the number of different effects in all loaded libraries.
+    virtual status_t queryNumberEffects(uint32_t *pNumEffects);
+
+    // Returns a descriptor of the next available effect.
+    virtual status_t getDescriptor(uint32_t index,
+            effect_descriptor_t *pDescriptor);
+
+    virtual status_t getDescriptor(const effect_uuid_t *pEffectUuid,
+            effect_descriptor_t *pDescriptor);
+
+    // Creates an effect engine of the specified type.
+    // To release the effect engine, it is necessary to release references
+    // to the returned effect object.
+    virtual status_t createEffect(const effect_uuid_t *pEffectUuid,
+            int32_t sessionId, int32_t ioId,
+            sp<EffectHalInterface> *effect);
+
+    virtual status_t dumpEffects(int fd);
+
+    status_t allocateBuffer(size_t size, sp<EffectBufferHalInterface>* buffer) override;
+    status_t mirrorBuffer(void* external, size_t size,
+                          sp<EffectBufferHalInterface>* buffer) override;
+
+  private:
+    friend class EffectsFactoryHalInterface;
+
+    sp<IEffectsFactory> mEffectsFactory;
+    hidl_vec<EffectDescriptor> mLastDescriptors;
+
+    // Can not be constructed directly by clients.
+    EffectsFactoryHalHidl();
+    virtual ~EffectsFactoryHalHidl();
+
+    status_t queryAllDescriptors();
+};
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H

diff --git a/media/libaudiohal/2.0/StreamHalHidl.cpp b/media/libaudiohal/2.0/StreamHalHidl.cpp
new file mode 100644
index 0000000..9869cd2
--- /dev/null
+++ b/media/libaudiohal/2.0/StreamHalHidl.cpp

@@ -0,0 +1,768 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "StreamHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <android/hardware/audio/2.0/IStreamOutCallback.h>
+#include <hwbinder/IPCThreadState.h>
+#include <mediautils/SchedulingPolicyService.h>
+#include <utils/Log.h>
+
+#include "DeviceHalHidl.h"
+#include "EffectHalHidl.h"
+#include "StreamHalHidl.h"
+
+using ::android::hardware::audio::common::V2_0::AudioChannelMask;
+using ::android::hardware::audio::common::V2_0::AudioFormat;
+using ::android::hardware::audio::common::V2_0::ThreadInfo;
+using ::android::hardware::audio::V2_0::AudioDrain;
+using ::android::hardware::audio::V2_0::IStreamOutCallback;
+using ::android::hardware::audio::V2_0::MessageQueueFlagBits;
+using ::android::hardware::audio::V2_0::MmapBufferInfo;
+using ::android::hardware::audio::V2_0::MmapPosition;
+using ::android::hardware::audio::V2_0::ParameterValue;
+using ::android::hardware::audio::V2_0::Result;
+using ::android::hardware::audio::V2_0::TimeSpec;
+using ::android::hardware::MQDescriptorSync;
+using ::android::hardware::Return;
+using ::android::hardware::Void;
+using ReadCommand = ::android::hardware::audio::V2_0::IStreamIn::ReadCommand;
+
+namespace android {
+
+StreamHalHidl::StreamHalHidl(IStream *stream)
+        : ConversionHelperHidl("Stream"),
+          mStream(stream),
+          mHalThreadPriority(HAL_THREAD_PRIORITY_DEFAULT),
+          mCachedBufferSize(0){
+
+    // Instrument audio signal power logging.
+    // Note: This assumes channel mask, format, and sample rate do not change after creation.
+    if (mStream != nullptr && mStreamPowerLog.isUserDebugOrEngBuild()) {
+        // Obtain audio properties (see StreamHalHidl::getAudioProperties() below).
+        Return<void> ret = mStream->getAudioProperties(
+                [&](uint32_t sr, AudioChannelMask m, AudioFormat f) {
+                mStreamPowerLog.init(sr,
+                        static_cast<audio_channel_mask_t>(m),
+                        static_cast<audio_format_t>(f));
+            });
+    }
+}
+
+StreamHalHidl::~StreamHalHidl() {
+    mStream = nullptr;
+}
+
+status_t StreamHalHidl::getSampleRate(uint32_t *rate) {
+    if (!mStream) return NO_INIT;
+    return processReturn("getSampleRate", mStream->getSampleRate(), rate);
+}
+
+status_t StreamHalHidl::getBufferSize(size_t *size) {
+    if (!mStream) return NO_INIT;
+    status_t status = processReturn("getBufferSize", mStream->getBufferSize(), size);
+    if (status == OK) {
+        mCachedBufferSize = *size;
+    }
+    return status;
+}
+
+status_t StreamHalHidl::getChannelMask(audio_channel_mask_t *mask) {
+    if (!mStream) return NO_INIT;
+    return processReturn("getChannelMask", mStream->getChannelMask(), mask);
+}
+
+status_t StreamHalHidl::getFormat(audio_format_t *format) {
+    if (!mStream) return NO_INIT;
+    return processReturn("getFormat", mStream->getFormat(), format);
+}
+
+status_t StreamHalHidl::getAudioProperties(
+        uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format) {
+    if (!mStream) return NO_INIT;
+    Return<void> ret = mStream->getAudioProperties(
+            [&](uint32_t sr, AudioChannelMask m, AudioFormat f) {
+                *sampleRate = sr;
+                *mask = static_cast<audio_channel_mask_t>(m);
+                *format = static_cast<audio_format_t>(f);
+            });
+    return processReturn("getAudioProperties", ret);
+}
+
+status_t StreamHalHidl::setParameters(const String8& kvPairs) {
+    if (!mStream) return NO_INIT;
+    hidl_vec<ParameterValue> hidlParams;
+    status_t status = parametersFromHal(kvPairs, &hidlParams);
+    if (status != OK) return status;
+    return processReturn("setParameters", mStream->setParameters(hidlParams));
+}
+
+status_t StreamHalHidl::getParameters(const String8& keys, String8 *values) {
+    values->clear();
+    if (!mStream) return NO_INIT;
+    hidl_vec<hidl_string> hidlKeys;
+    status_t status = keysFromHal(keys, &hidlKeys);
+    if (status != OK) return status;
+    Result retval;
+    Return<void> ret = mStream->getParameters(
+            hidlKeys,
+            [&](Result r, const hidl_vec<ParameterValue>& parameters) {
+                retval = r;
+                if (retval == Result::OK) {
+                    parametersToHal(parameters, values);
+                }
+            });
+    return processReturn("getParameters", ret, retval);
+}
+
+status_t StreamHalHidl::addEffect(sp<EffectHalInterface> effect) {
+    if (!mStream) return NO_INIT;
+    return processReturn("addEffect", mStream->addEffect(
+                    static_cast<EffectHalHidl*>(effect.get())->effectId()));
+}
+
+status_t StreamHalHidl::removeEffect(sp<EffectHalInterface> effect) {
+    if (!mStream) return NO_INIT;
+    return processReturn("removeEffect", mStream->removeEffect(
+                    static_cast<EffectHalHidl*>(effect.get())->effectId()));
+}
+
+status_t StreamHalHidl::standby() {
+    if (!mStream) return NO_INIT;
+    return processReturn("standby", mStream->standby());
+}
+
+status_t StreamHalHidl::dump(int fd) {
+    if (!mStream) return NO_INIT;
+    native_handle_t* hidlHandle = native_handle_create(1, 0);
+    hidlHandle->data[0] = fd;
+    Return<void> ret = mStream->debugDump(hidlHandle);
+    native_handle_delete(hidlHandle);
+    mStreamPowerLog.dump(fd);
+    return processReturn("dump", ret);
+}
+
+status_t StreamHalHidl::start() {
+    if (!mStream) return NO_INIT;
+    return processReturn("start", mStream->start());
+}
+
+status_t StreamHalHidl::stop() {
+    if (!mStream) return NO_INIT;
+    return processReturn("stop", mStream->stop());
+}
+
+status_t StreamHalHidl::createMmapBuffer(int32_t minSizeFrames,
+                                  struct audio_mmap_buffer_info *info) {
+    Result retval;
+    Return<void> ret = mStream->createMmapBuffer(
+            minSizeFrames,
+            [&](Result r, const MmapBufferInfo& hidlInfo) {
+                retval = r;
+                if (retval == Result::OK) {
+                    const native_handle *handle = hidlInfo.sharedMemory.handle();
+                    if (handle->numFds > 0) {
+                        info->shared_memory_fd = handle->data[0];
+                        info->buffer_size_frames = hidlInfo.bufferSizeFrames;
+                        info->burst_size_frames = hidlInfo.burstSizeFrames;
+                        // info->shared_memory_address is not needed in HIDL context
+                        info->shared_memory_address = NULL;
+                    } else {
+                        retval = Result::NOT_INITIALIZED;
+                    }
+                }
+            });
+    return processReturn("createMmapBuffer", ret, retval);
+}
+
+status_t StreamHalHidl::getMmapPosition(struct audio_mmap_position *position) {
+    Result retval;
+    Return<void> ret = mStream->getMmapPosition(
+            [&](Result r, const MmapPosition& hidlPosition) {
+                retval = r;
+                if (retval == Result::OK) {
+                    position->time_nanoseconds = hidlPosition.timeNanoseconds;
+                    position->position_frames = hidlPosition.positionFrames;
+                }
+            });
+    return processReturn("getMmapPosition", ret, retval);
+}
+
+status_t StreamHalHidl::setHalThreadPriority(int priority) {
+    mHalThreadPriority = priority;
+    return OK;
+}
+
+status_t StreamHalHidl::getCachedBufferSize(size_t *size) {
+    if (mCachedBufferSize != 0) {
+        *size = mCachedBufferSize;
+        return OK;
+    }
+    return getBufferSize(size);
+}
+
+bool StreamHalHidl::requestHalThreadPriority(pid_t threadPid, pid_t threadId) {
+    if (mHalThreadPriority == HAL_THREAD_PRIORITY_DEFAULT) {
+        return true;
+    }
+    int err = requestPriority(
+            threadPid, threadId,
+            mHalThreadPriority, false /*isForApp*/, true /*asynchronous*/);
+    ALOGE_IF(err, "failed to set priority %d for pid %d tid %d; error %d",
+            mHalThreadPriority, threadPid, threadId, err);
+    // Audio will still work, but latency will be higher and sometimes unacceptable.
+    return err == 0;
+}
+
+namespace {
+
+/* Notes on callback ownership.
+
+This is how (Hw)Binder ownership model looks like. The server implementation
+is owned by Binder framework (via sp<>). Proxies are owned by clients.
+When the last proxy disappears, Binder framework releases the server impl.
+
+Thus, it is not needed to keep any references to StreamOutCallback (this is
+the server impl) -- it will live as long as HAL server holds a strong ref to
+IStreamOutCallback proxy. We clear that reference by calling 'clearCallback'
+from the destructor of StreamOutHalHidl.
+
+The callback only keeps a weak reference to the stream. The stream is owned
+by AudioFlinger.
+
+*/
+
+struct StreamOutCallback : public IStreamOutCallback {
+    StreamOutCallback(const wp<StreamOutHalHidl>& stream) : mStream(stream) {}
+
+    // IStreamOutCallback implementation
+    Return<void> onWriteReady()  override {
+        sp<StreamOutHalHidl> stream = mStream.promote();
+        if (stream != 0) {
+            stream->onWriteReady();
+        }
+        return Void();
+    }
+
+    Return<void> onDrainReady()  override {
+        sp<StreamOutHalHidl> stream = mStream.promote();
+        if (stream != 0) {
+            stream->onDrainReady();
+        }
+        return Void();
+    }
+
+    Return<void> onError()  override {
+        sp<StreamOutHalHidl> stream = mStream.promote();
+        if (stream != 0) {
+            stream->onError();
+        }
+        return Void();
+    }
+
+  private:
+    wp<StreamOutHalHidl> mStream;
+};
+
+}  // namespace
+
+StreamOutHalHidl::StreamOutHalHidl(const sp<IStreamOut>& stream)
+        : StreamHalHidl(stream.get()), mStream(stream), mWriterClient(0), mEfGroup(nullptr) {
+}
+
+StreamOutHalHidl::~StreamOutHalHidl() {
+    if (mStream != 0) {
+        if (mCallback.unsafe_get()) {
+            processReturn("clearCallback", mStream->clearCallback());
+        }
+        processReturn("close", mStream->close());
+        mStream.clear();
+    }
+    mCallback.clear();
+    hardware::IPCThreadState::self()->flushCommands();
+    if (mEfGroup) {
+        EventFlag::deleteEventFlag(&mEfGroup);
+    }
+}
+
+status_t StreamOutHalHidl::getFrameSize(size_t *size) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("getFrameSize", mStream->getFrameSize(), size);
+}
+
+status_t StreamOutHalHidl::getLatency(uint32_t *latency) {
+    if (mStream == 0) return NO_INIT;
+    if (mWriterClient == gettid() && mCommandMQ) {
+        return callWriterThread(
+                WriteCommand::GET_LATENCY, "getLatency", nullptr, 0,
+                [&](const WriteStatus& writeStatus) {
+                    *latency = writeStatus.reply.latencyMs;
+                });
+    } else {
+        return processReturn("getLatency", mStream->getLatency(), latency);
+    }
+}
+
+status_t StreamOutHalHidl::setVolume(float left, float right) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("setVolume", mStream->setVolume(left, right));
+}
+
+status_t StreamOutHalHidl::write(const void *buffer, size_t bytes, size_t *written) {
+    if (mStream == 0) return NO_INIT;
+    *written = 0;
+
+    if (bytes == 0 && !mDataMQ) {
+        // Can't determine the size for the MQ buffer. Wait for a non-empty write request.
+        ALOGW_IF(mCallback.unsafe_get(), "First call to async write with 0 bytes");
+        return OK;
+    }
+
+    status_t status;
+    if (!mDataMQ) {
+        // In case if playback starts close to the end of a compressed track, the bytes
+        // that need to be written is less than the actual buffer size. Need to use
+        // full buffer size for the MQ since otherwise after seeking back to the middle
+        // data will be truncated.
+        size_t bufferSize;
+        if ((status = getCachedBufferSize(&bufferSize)) != OK) {
+            return status;
+        }
+        if (bytes > bufferSize) bufferSize = bytes;
+        if ((status = prepareForWriting(bufferSize)) != OK) {
+            return status;
+        }
+    }
+
+    status = callWriterThread(
+            WriteCommand::WRITE, "write", static_cast<const uint8_t*>(buffer), bytes,
+            [&] (const WriteStatus& writeStatus) {
+                *written = writeStatus.reply.written;
+                // Diagnostics of the cause of b/35813113.
+                ALOGE_IF(*written > bytes,
+                        "hal reports more bytes written than asked for: %lld > %lld",
+                        (long long)*written, (long long)bytes);
+            });
+    mStreamPowerLog.log(buffer, *written);
+    return status;
+}
+
+status_t StreamOutHalHidl::callWriterThread(
+        WriteCommand cmd, const char* cmdName,
+        const uint8_t* data, size_t dataSize, StreamOutHalHidl::WriterCallback callback) {
+    if (!mCommandMQ->write(&cmd)) {
+        ALOGE("command message queue write failed for \"%s\"", cmdName);
+        return -EAGAIN;
+    }
+    if (data != nullptr) {
+        size_t availableToWrite = mDataMQ->availableToWrite();
+        if (dataSize > availableToWrite) {
+            ALOGW("truncating write data from %lld to %lld due to insufficient data queue space",
+                    (long long)dataSize, (long long)availableToWrite);
+            dataSize = availableToWrite;
+        }
+        if (!mDataMQ->write(data, dataSize)) {
+            ALOGE("data message queue write failed for \"%s\"", cmdName);
+        }
+    }
+    mEfGroup->wake(static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY));
+
+    // TODO: Remove manual event flag handling once blocking MQ is implemented. b/33815422
+    uint32_t efState = 0;
+retry:
+    status_t ret = mEfGroup->wait(static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL), &efState);
+    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL)) {
+        WriteStatus writeStatus;
+        writeStatus.retval = Result::NOT_INITIALIZED;
+        if (!mStatusMQ->read(&writeStatus)) {
+            ALOGE("status message read failed for \"%s\"", cmdName);
+        }
+        if (writeStatus.retval == Result::OK) {
+            ret = OK;
+            callback(writeStatus);
+        } else {
+            ret = processReturn(cmdName, writeStatus.retval);
+        }
+        return ret;
+    }
+    if (ret == -EAGAIN || ret == -EINTR) {
+        // Spurious wakeup. This normally retries no more than once.
+        goto retry;
+    }
+    return ret;
+}
+
+status_t StreamOutHalHidl::prepareForWriting(size_t bufferSize) {
+    std::unique_ptr<CommandMQ> tempCommandMQ;
+    std::unique_ptr<DataMQ> tempDataMQ;
+    std::unique_ptr<StatusMQ> tempStatusMQ;
+    Result retval;
+    pid_t halThreadPid, halThreadTid;
+    Return<void> ret = mStream->prepareForWriting(
+            1, bufferSize,
+            [&](Result r,
+                    const CommandMQ::Descriptor& commandMQ,
+                    const DataMQ::Descriptor& dataMQ,
+                    const StatusMQ::Descriptor& statusMQ,
+                    const ThreadInfo& halThreadInfo) {
+                retval = r;
+                if (retval == Result::OK) {
+                    tempCommandMQ.reset(new CommandMQ(commandMQ));
+                    tempDataMQ.reset(new DataMQ(dataMQ));
+                    tempStatusMQ.reset(new StatusMQ(statusMQ));
+                    if (tempDataMQ->isValid() && tempDataMQ->getEventFlagWord()) {
+                        EventFlag::createEventFlag(tempDataMQ->getEventFlagWord(), &mEfGroup);
+                    }
+                    halThreadPid = halThreadInfo.pid;
+                    halThreadTid = halThreadInfo.tid;
+                }
+            });
+    if (!ret.isOk() || retval != Result::OK) {
+        return processReturn("prepareForWriting", ret, retval);
+    }
+    if (!tempCommandMQ || !tempCommandMQ->isValid() ||
+            !tempDataMQ || !tempDataMQ->isValid() ||
+            !tempStatusMQ || !tempStatusMQ->isValid() ||
+            !mEfGroup) {
+        ALOGE_IF(!tempCommandMQ, "Failed to obtain command message queue for writing");
+        ALOGE_IF(tempCommandMQ && !tempCommandMQ->isValid(),
+                "Command message queue for writing is invalid");
+        ALOGE_IF(!tempDataMQ, "Failed to obtain data message queue for writing");
+        ALOGE_IF(tempDataMQ && !tempDataMQ->isValid(), "Data message queue for writing is invalid");
+        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for writing");
+        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
+                "Status message queue for writing is invalid");
+        ALOGE_IF(!mEfGroup, "Event flag creation for writing failed");
+        return NO_INIT;
+    }
+    requestHalThreadPriority(halThreadPid, halThreadTid);
+
+    mCommandMQ = std::move(tempCommandMQ);
+    mDataMQ = std::move(tempDataMQ);
+    mStatusMQ = std::move(tempStatusMQ);
+    mWriterClient = gettid();
+    return OK;
+}
+
+status_t StreamOutHalHidl::getRenderPosition(uint32_t *dspFrames) {
+    if (mStream == 0) return NO_INIT;
+    Result retval;
+    Return<void> ret = mStream->getRenderPosition(
+            [&](Result r, uint32_t d) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *dspFrames = d;
+                }
+            });
+    return processReturn("getRenderPosition", ret, retval);
+}
+
+status_t StreamOutHalHidl::getNextWriteTimestamp(int64_t *timestamp) {
+    if (mStream == 0) return NO_INIT;
+    Result retval;
+    Return<void> ret = mStream->getNextWriteTimestamp(
+            [&](Result r, int64_t t) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *timestamp = t;
+                }
+            });
+    return processReturn("getRenderPosition", ret, retval);
+}
+
+status_t StreamOutHalHidl::setCallback(wp<StreamOutHalInterfaceCallback> callback) {
+    if (mStream == 0) return NO_INIT;
+    status_t status = processReturn(
+            "setCallback", mStream->setCallback(new StreamOutCallback(this)));
+    if (status == OK) {
+        mCallback = callback;
+    }
+    return status;
+}
+
+status_t StreamOutHalHidl::supportsPauseAndResume(bool *supportsPause, bool *supportsResume) {
+    if (mStream == 0) return NO_INIT;
+    Return<void> ret = mStream->supportsPauseAndResume(
+            [&](bool p, bool r) {
+                *supportsPause = p;
+                *supportsResume = r;
+            });
+    return processReturn("supportsPauseAndResume", ret);
+}
+
+status_t StreamOutHalHidl::pause() {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("pause", mStream->pause());
+}
+
+status_t StreamOutHalHidl::resume() {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("pause", mStream->resume());
+}
+
+status_t StreamOutHalHidl::supportsDrain(bool *supportsDrain) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("supportsDrain", mStream->supportsDrain(), supportsDrain);
+}
+
+status_t StreamOutHalHidl::drain(bool earlyNotify) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn(
+            "drain", mStream->drain(earlyNotify ? AudioDrain::EARLY_NOTIFY : AudioDrain::ALL));
+}
+
+status_t StreamOutHalHidl::flush() {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("pause", mStream->flush());
+}
+
+status_t StreamOutHalHidl::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) {
+    if (mStream == 0) return NO_INIT;
+    if (mWriterClient == gettid() && mCommandMQ) {
+        return callWriterThread(
+                WriteCommand::GET_PRESENTATION_POSITION, "getPresentationPosition", nullptr, 0,
+                [&](const WriteStatus& writeStatus) {
+                    *frames = writeStatus.reply.presentationPosition.frames;
+                    timestamp->tv_sec = writeStatus.reply.presentationPosition.timeStamp.tvSec;
+                    timestamp->tv_nsec = writeStatus.reply.presentationPosition.timeStamp.tvNSec;
+                });
+    } else {
+        Result retval;
+        Return<void> ret = mStream->getPresentationPosition(
+                [&](Result r, uint64_t hidlFrames, const TimeSpec& hidlTimeStamp) {
+                    retval = r;
+                    if (retval == Result::OK) {
+                        *frames = hidlFrames;
+                        timestamp->tv_sec = hidlTimeStamp.tvSec;
+                        timestamp->tv_nsec = hidlTimeStamp.tvNSec;
+                    }
+                });
+        return processReturn("getPresentationPosition", ret, retval);
+    }
+}
+
+status_t StreamOutHalHidl::updateSourceMetadata(const SourceMetadata& /* sourceMetadata */) {
+    // Audio HAL V2.0 does not support propagating source metadata
+    return INVALID_OPERATION;
+}
+
+void StreamOutHalHidl::onWriteReady() {
+    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
+    if (callback == 0) return;
+    ALOGV("asyncCallback onWriteReady");
+    callback->onWriteReady();
+}
+
+void StreamOutHalHidl::onDrainReady() {
+    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
+    if (callback == 0) return;
+    ALOGV("asyncCallback onDrainReady");
+    callback->onDrainReady();
+}
+
+void StreamOutHalHidl::onError() {
+    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
+    if (callback == 0) return;
+    ALOGV("asyncCallback onError");
+    callback->onError();
+}
+
+
+StreamInHalHidl::StreamInHalHidl(const sp<IStreamIn>& stream)
+        : StreamHalHidl(stream.get()), mStream(stream), mReaderClient(0), mEfGroup(nullptr) {
+}
+
+StreamInHalHidl::~StreamInHalHidl() {
+    if (mStream != 0) {
+        processReturn("close", mStream->close());
+        mStream.clear();
+        hardware::IPCThreadState::self()->flushCommands();
+    }
+    if (mEfGroup) {
+        EventFlag::deleteEventFlag(&mEfGroup);
+    }
+}
+
+status_t StreamInHalHidl::getFrameSize(size_t *size) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("getFrameSize", mStream->getFrameSize(), size);
+}
+
+status_t StreamInHalHidl::setGain(float gain) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("setGain", mStream->setGain(gain));
+}
+
+status_t StreamInHalHidl::read(void *buffer, size_t bytes, size_t *read) {
+    if (mStream == 0) return NO_INIT;
+    *read = 0;
+
+    if (bytes == 0 && !mDataMQ) {
+        // Can't determine the size for the MQ buffer. Wait for a non-empty read request.
+        return OK;
+    }
+
+    status_t status;
+    if (!mDataMQ && (status = prepareForReading(bytes)) != OK) {
+        return status;
+    }
+
+    ReadParameters params;
+    params.command = ReadCommand::READ;
+    params.params.read = bytes;
+    status = callReaderThread(params, "read",
+            [&](const ReadStatus& readStatus) {
+                const size_t availToRead = mDataMQ->availableToRead();
+                if (!mDataMQ->read(static_cast<uint8_t*>(buffer), std::min(bytes, availToRead))) {
+                    ALOGE("data message queue read failed for \"read\"");
+                }
+                ALOGW_IF(availToRead != readStatus.reply.read,
+                        "HAL read report inconsistent: mq = %d, status = %d",
+                        (int32_t)availToRead, (int32_t)readStatus.reply.read);
+                *read = readStatus.reply.read;
+            });
+    mStreamPowerLog.log(buffer, *read);
+    return status;
+}
+
+status_t StreamInHalHidl::callReaderThread(
+        const ReadParameters& params, const char* cmdName,
+        StreamInHalHidl::ReaderCallback callback) {
+    if (!mCommandMQ->write(&params)) {
+        ALOGW("command message queue write failed");
+        return -EAGAIN;
+    }
+    mEfGroup->wake(static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL));
+
+    // TODO: Remove manual event flag handling once blocking MQ is implemented. b/33815422
+    uint32_t efState = 0;
+retry:
+    status_t ret = mEfGroup->wait(static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY), &efState);
+    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY)) {
+        ReadStatus readStatus;
+        readStatus.retval = Result::NOT_INITIALIZED;
+        if (!mStatusMQ->read(&readStatus)) {
+            ALOGE("status message read failed for \"%s\"", cmdName);
+        }
+         if (readStatus.retval == Result::OK) {
+            ret = OK;
+            callback(readStatus);
+        } else {
+            ret = processReturn(cmdName, readStatus.retval);
+        }
+        return ret;
+    }
+    if (ret == -EAGAIN || ret == -EINTR) {
+        // Spurious wakeup. This normally retries no more than once.
+        goto retry;
+    }
+    return ret;
+}
+
+status_t StreamInHalHidl::prepareForReading(size_t bufferSize) {
+    std::unique_ptr<CommandMQ> tempCommandMQ;
+    std::unique_ptr<DataMQ> tempDataMQ;
+    std::unique_ptr<StatusMQ> tempStatusMQ;
+    Result retval;
+    pid_t halThreadPid, halThreadTid;
+    Return<void> ret = mStream->prepareForReading(
+            1, bufferSize,
+            [&](Result r,
+                    const CommandMQ::Descriptor& commandMQ,
+                    const DataMQ::Descriptor& dataMQ,
+                    const StatusMQ::Descriptor& statusMQ,
+                    const ThreadInfo& halThreadInfo) {
+                retval = r;
+                if (retval == Result::OK) {
+                    tempCommandMQ.reset(new CommandMQ(commandMQ));
+                    tempDataMQ.reset(new DataMQ(dataMQ));
+                    tempStatusMQ.reset(new StatusMQ(statusMQ));
+                    if (tempDataMQ->isValid() && tempDataMQ->getEventFlagWord()) {
+                        EventFlag::createEventFlag(tempDataMQ->getEventFlagWord(), &mEfGroup);
+                    }
+                    halThreadPid = halThreadInfo.pid;
+                    halThreadTid = halThreadInfo.tid;
+                }
+            });
+    if (!ret.isOk() || retval != Result::OK) {
+        return processReturn("prepareForReading", ret, retval);
+    }
+    if (!tempCommandMQ || !tempCommandMQ->isValid() ||
+            !tempDataMQ || !tempDataMQ->isValid() ||
+            !tempStatusMQ || !tempStatusMQ->isValid() ||
+            !mEfGroup) {
+        ALOGE_IF(!tempCommandMQ, "Failed to obtain command message queue for writing");
+        ALOGE_IF(tempCommandMQ && !tempCommandMQ->isValid(),
+                "Command message queue for writing is invalid");
+        ALOGE_IF(!tempDataMQ, "Failed to obtain data message queue for reading");
+        ALOGE_IF(tempDataMQ && !tempDataMQ->isValid(), "Data message queue for reading is invalid");
+        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for reading");
+        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
+                "Status message queue for reading is invalid");
+        ALOGE_IF(!mEfGroup, "Event flag creation for reading failed");
+        return NO_INIT;
+    }
+    requestHalThreadPriority(halThreadPid, halThreadTid);
+
+    mCommandMQ = std::move(tempCommandMQ);
+    mDataMQ = std::move(tempDataMQ);
+    mStatusMQ = std::move(tempStatusMQ);
+    mReaderClient = gettid();
+    return OK;
+}
+
+status_t StreamInHalHidl::getInputFramesLost(uint32_t *framesLost) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("getInputFramesLost", mStream->getInputFramesLost(), framesLost);
+}
+
+status_t StreamInHalHidl::getCapturePosition(int64_t *frames, int64_t *time) {
+    if (mStream == 0) return NO_INIT;
+    if (mReaderClient == gettid() && mCommandMQ) {
+        ReadParameters params;
+        params.command = ReadCommand::GET_CAPTURE_POSITION;
+        return callReaderThread(params, "getCapturePosition",
+                [&](const ReadStatus& readStatus) {
+                    *frames = readStatus.reply.capturePosition.frames;
+                    *time = readStatus.reply.capturePosition.time;
+                });
+    } else {
+        Result retval;
+        Return<void> ret = mStream->getCapturePosition(
+                [&](Result r, uint64_t hidlFrames, uint64_t hidlTime) {
+                    retval = r;
+                    if (retval == Result::OK) {
+                        *frames = hidlFrames;
+                        *time = hidlTime;
+                    }
+                });
+        return processReturn("getCapturePosition", ret, retval);
+    }
+}
+
+status_t StreamInHalHidl::getActiveMicrophones(
+        std::vector<media::MicrophoneInfo> *microphones __unused) {
+    if (mStream == 0) return NO_INIT;
+    return INVALID_OPERATION;
+}
+
+status_t StreamInHalHidl::updateSinkMetadata(const SinkMetadata& /* sinkMetadata */) {
+    // Audio HAL V2.0 does not support propagating sink metadata
+    return INVALID_OPERATION;
+}
+
+} // namespace android

diff --git a/media/libaudiohal/2.0/StreamHalHidl.h b/media/libaudiohal/2.0/StreamHalHidl.h
new file mode 100644
index 0000000..ebad8ae
--- /dev/null
+++ b/media/libaudiohal/2.0/StreamHalHidl.h

@@ -0,0 +1,248 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_STREAM_HAL_HIDL_H
+#define ANDROID_HARDWARE_STREAM_HAL_HIDL_H
+
+#include <atomic>
+
+#include <android/hardware/audio/2.0/IStream.h>
+#include <android/hardware/audio/2.0/IStreamIn.h>
+#include <android/hardware/audio/2.0/IStreamOut.h>
+#include <fmq/EventFlag.h>
+#include <fmq/MessageQueue.h>
+#include <media/audiohal/StreamHalInterface.h>
+
+#include "ConversionHelperHidl.h"
+#include "StreamPowerLog.h"
+
+using ::android::hardware::audio::V2_0::IStream;
+using ::android::hardware::audio::V2_0::IStreamIn;
+using ::android::hardware::audio::V2_0::IStreamOut;
+using ::android::hardware::EventFlag;
+using ::android::hardware::MessageQueue;
+using ::android::hardware::Return;
+using ReadParameters = ::android::hardware::audio::V2_0::IStreamIn::ReadParameters;
+using ReadStatus = ::android::hardware::audio::V2_0::IStreamIn::ReadStatus;
+using WriteCommand = ::android::hardware::audio::V2_0::IStreamOut::WriteCommand;
+using WriteStatus = ::android::hardware::audio::V2_0::IStreamOut::WriteStatus;
+
+namespace android {
+
+class DeviceHalHidl;
+
+class StreamHalHidl : public virtual StreamHalInterface, public ConversionHelperHidl
+{
+  public:
+    // Return the sampling rate in Hz - eg. 44100.
+    virtual status_t getSampleRate(uint32_t *rate);
+
+    // Return size of input/output buffer in bytes for this stream - eg. 4800.
+    virtual status_t getBufferSize(size_t *size);
+
+    // Return the channel mask.
+    virtual status_t getChannelMask(audio_channel_mask_t *mask);
+
+    // Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT.
+    virtual status_t getFormat(audio_format_t *format);
+
+    // Convenience method.
+    virtual status_t getAudioProperties(
+            uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format);
+
+    // Set audio stream parameters.
+    virtual status_t setParameters(const String8& kvPairs);
+
+    // Get audio stream parameters.
+    virtual status_t getParameters(const String8& keys, String8 *values);
+
+    // Add or remove the effect on the stream.
+    virtual status_t addEffect(sp<EffectHalInterface> effect);
+    virtual status_t removeEffect(sp<EffectHalInterface> effect);
+
+    // Put the audio hardware input/output into standby mode.
+    virtual status_t standby();
+
+    virtual status_t dump(int fd);
+
+    // Start a stream operating in mmap mode.
+    virtual status_t start();
+
+    // Stop a stream operating in mmap mode.
+    virtual status_t stop();
+
+    // Retrieve information on the data buffer in mmap mode.
+    virtual status_t createMmapBuffer(int32_t minSizeFrames,
+                                      struct audio_mmap_buffer_info *info);
+
+    // Get current read/write position in the mmap buffer
+    virtual status_t getMmapPosition(struct audio_mmap_position *position);
+
+    // Set the priority of the thread that interacts with the HAL
+    // (must match the priority of the audioflinger's thread that calls 'read' / 'write')
+    virtual status_t setHalThreadPriority(int priority);
+
+  protected:
+    // Subclasses can not be constructed directly by clients.
+    explicit StreamHalHidl(IStream *stream);
+
+    // The destructor automatically closes the stream.
+    virtual ~StreamHalHidl();
+
+    status_t getCachedBufferSize(size_t *size);
+
+    bool requestHalThreadPriority(pid_t threadPid, pid_t threadId);
+
+    // mStreamPowerLog is used for audio signal power logging.
+    StreamPowerLog mStreamPowerLog;
+
+  private:
+    const int HAL_THREAD_PRIORITY_DEFAULT = -1;
+    IStream *mStream;
+    int mHalThreadPriority;
+    size_t mCachedBufferSize;
+};
+
+class StreamOutHalHidl : public StreamOutHalInterface, public StreamHalHidl {
+  public:
+    // Return the frame size (number of bytes per sample) of a stream.
+    virtual status_t getFrameSize(size_t *size);
+
+    // Return the audio hardware driver estimated latency in milliseconds.
+    virtual status_t getLatency(uint32_t *latency);
+
+    // Use this method in situations where audio mixing is done in the hardware.
+    virtual status_t setVolume(float left, float right);
+
+    // Write audio buffer to driver.
+    virtual status_t write(const void *buffer, size_t bytes, size_t *written);
+
+    // Return the number of audio frames written by the audio dsp to DAC since
+    // the output has exited standby.
+    virtual status_t getRenderPosition(uint32_t *dspFrames);
+
+    // Get the local time at which the next write to the audio driver will be presented.
+    virtual status_t getNextWriteTimestamp(int64_t *timestamp);
+
+    // Set the callback for notifying completion of non-blocking write and drain.
+    virtual status_t setCallback(wp<StreamOutHalInterfaceCallback> callback);
+
+    // Returns whether pause and resume operations are supported.
+    virtual status_t supportsPauseAndResume(bool *supportsPause, bool *supportsResume);
+
+    // Notifies to the audio driver to resume playback following a pause.
+    virtual status_t pause();
+
+    // Notifies to the audio driver to resume playback following a pause.
+    virtual status_t resume();
+
+    // Returns whether drain operation is supported.
+    virtual status_t supportsDrain(bool *supportsDrain);
+
+    // Requests notification when data buffered by the driver/hardware has been played.
+    virtual status_t drain(bool earlyNotify);
+
+    // Notifies to the audio driver to flush the queued data.
+    virtual status_t flush();
+
+    // Return a recent count of the number of audio frames presented to an external observer.
+    virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
+
+    // Called when the metadata of the stream's source has been changed.
+    status_t updateSourceMetadata(const SourceMetadata& sourceMetadata) override;
+
+    // Methods used by StreamOutCallback (HIDL).
+    void onWriteReady();
+    void onDrainReady();
+    void onError();
+
+  private:
+    friend class DeviceHalHidl;
+    typedef MessageQueue<WriteCommand, hardware::kSynchronizedReadWrite> CommandMQ;
+    typedef MessageQueue<uint8_t, hardware::kSynchronizedReadWrite> DataMQ;
+    typedef MessageQueue<WriteStatus, hardware::kSynchronizedReadWrite> StatusMQ;
+
+    wp<StreamOutHalInterfaceCallback> mCallback;
+    sp<IStreamOut> mStream;
+    std::unique_ptr<CommandMQ> mCommandMQ;
+    std::unique_ptr<DataMQ> mDataMQ;
+    std::unique_ptr<StatusMQ> mStatusMQ;
+    std::atomic<pid_t> mWriterClient;
+    EventFlag* mEfGroup;
+
+    // Can not be constructed directly by clients.
+    StreamOutHalHidl(const sp<IStreamOut>& stream);
+
+    virtual ~StreamOutHalHidl();
+
+    using WriterCallback = std::function<void(const WriteStatus& writeStatus)>;
+    status_t callWriterThread(
+            WriteCommand cmd, const char* cmdName,
+            const uint8_t* data, size_t dataSize, WriterCallback callback);
+    status_t prepareForWriting(size_t bufferSize);
+};
+
+class StreamInHalHidl : public StreamInHalInterface, public StreamHalHidl {
+  public:
+    // Return the frame size (number of bytes per sample) of a stream.
+    virtual status_t getFrameSize(size_t *size);
+
+    // Set the input gain for the audio driver.
+    virtual status_t setGain(float gain);
+
+    // Read audio buffer in from driver.
+    virtual status_t read(void *buffer, size_t bytes, size_t *read);
+
+    // Return the amount of input frames lost in the audio driver.
+    virtual status_t getInputFramesLost(uint32_t *framesLost);
+
+    // Return a recent count of the number of audio frames received and
+    // the clock time associated with that frame count.
+    virtual status_t getCapturePosition(int64_t *frames, int64_t *time);
+
+    // Get active microphones
+    virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+
+    // Called when the metadata of the stream's sink has been changed.
+    status_t updateSinkMetadata(const SinkMetadata& sinkMetadata) override;
+
+  private:
+    friend class DeviceHalHidl;
+    typedef MessageQueue<ReadParameters, hardware::kSynchronizedReadWrite> CommandMQ;
+    typedef MessageQueue<uint8_t, hardware::kSynchronizedReadWrite> DataMQ;
+    typedef MessageQueue<ReadStatus, hardware::kSynchronizedReadWrite> StatusMQ;
+
+    sp<IStreamIn> mStream;
+    std::unique_ptr<CommandMQ> mCommandMQ;
+    std::unique_ptr<DataMQ> mDataMQ;
+    std::unique_ptr<StatusMQ> mStatusMQ;
+    std::atomic<pid_t> mReaderClient;
+    EventFlag* mEfGroup;
+
+    // Can not be constructed directly by clients.
+    StreamInHalHidl(const sp<IStreamIn>& stream);
+
+    virtual ~StreamInHalHidl();
+
+    using ReaderCallback = std::function<void(const ReadStatus& readStatus)>;
+    status_t callReaderThread(
+            const ReadParameters& params, const char* cmdName, ReaderCallback callback);
+    status_t prepareForReading(size_t bufferSize);
+};
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_STREAM_HAL_HIDL_H

diff --git a/media/libaudiohal/2.0/StreamHalLocal.cpp b/media/libaudiohal/2.0/StreamHalLocal.cpp
new file mode 100644
index 0000000..98107e5
--- /dev/null
+++ b/media/libaudiohal/2.0/StreamHalLocal.cpp

@@ -0,0 +1,347 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "StreamHalLocal"
+//#define LOG_NDEBUG 0
+
+#include <hardware/audio.h>
+#include <utils/Log.h>
+
+#include "DeviceHalLocal.h"
+#include "StreamHalLocal.h"
+
+namespace android {
+
+StreamHalLocal::StreamHalLocal(audio_stream_t *stream, sp<DeviceHalLocal> device)
+        : mDevice(device),
+          mStream(stream) {
+    // Instrument audio signal power logging.
+    // Note: This assumes channel mask, format, and sample rate do not change after creation.
+    if (mStream != nullptr && mStreamPowerLog.isUserDebugOrEngBuild()) {
+        mStreamPowerLog.init(mStream->get_sample_rate(mStream),
+                mStream->get_channels(mStream),
+                mStream->get_format(mStream));
+    }
+}
+
+StreamHalLocal::~StreamHalLocal() {
+    mStream = 0;
+    mDevice.clear();
+}
+
+status_t StreamHalLocal::getSampleRate(uint32_t *rate) {
+    *rate = mStream->get_sample_rate(mStream);
+    return OK;
+}
+
+status_t StreamHalLocal::getBufferSize(size_t *size) {
+    *size = mStream->get_buffer_size(mStream);
+    return OK;
+}
+
+status_t StreamHalLocal::getChannelMask(audio_channel_mask_t *mask) {
+    *mask = mStream->get_channels(mStream);
+    return OK;
+}
+
+status_t StreamHalLocal::getFormat(audio_format_t *format) {
+    *format = mStream->get_format(mStream);
+    return OK;
+}
+
+status_t StreamHalLocal::getAudioProperties(
+        uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format) {
+    *sampleRate = mStream->get_sample_rate(mStream);
+    *mask = mStream->get_channels(mStream);
+    *format = mStream->get_format(mStream);
+    return OK;
+}
+
+status_t StreamHalLocal::setParameters(const String8& kvPairs) {
+    return mStream->set_parameters(mStream, kvPairs.string());
+}
+
+status_t StreamHalLocal::getParameters(const String8& keys, String8 *values) {
+    char *halValues = mStream->get_parameters(mStream, keys.string());
+    if (halValues != NULL) {
+        values->setTo(halValues);
+        free(halValues);
+    } else {
+        values->clear();
+    }
+    return OK;
+}
+
+status_t StreamHalLocal::addEffect(sp<EffectHalInterface>) {
+    LOG_ALWAYS_FATAL("Local streams can not have effects");
+    return INVALID_OPERATION;
+}
+
+status_t StreamHalLocal::removeEffect(sp<EffectHalInterface>) {
+    LOG_ALWAYS_FATAL("Local streams can not have effects");
+    return INVALID_OPERATION;
+}
+
+status_t StreamHalLocal::standby() {
+    return mStream->standby(mStream);
+}
+
+status_t StreamHalLocal::dump(int fd) {
+    status_t status = mStream->dump(mStream, fd);
+    mStreamPowerLog.dump(fd);
+    return status;
+}
+
+status_t StreamHalLocal::setHalThreadPriority(int) {
+    // Don't need to do anything as local hal is executed by audioflinger directly
+    // on the same thread.
+    return OK;
+}
+
+StreamOutHalLocal::StreamOutHalLocal(audio_stream_out_t *stream, sp<DeviceHalLocal> device)
+        : StreamHalLocal(&stream->common, device), mStream(stream) {
+}
+
+StreamOutHalLocal::~StreamOutHalLocal() {
+    mCallback.clear();
+    mDevice->closeOutputStream(mStream);
+    mStream = 0;
+}
+
+status_t StreamOutHalLocal::getFrameSize(size_t *size) {
+    *size = audio_stream_out_frame_size(mStream);
+    return OK;
+}
+
+status_t StreamOutHalLocal::getLatency(uint32_t *latency) {
+    *latency = mStream->get_latency(mStream);
+    return OK;
+}
+
+status_t StreamOutHalLocal::setVolume(float left, float right) {
+    if (mStream->set_volume == NULL) return INVALID_OPERATION;
+    return mStream->set_volume(mStream, left, right);
+}
+
+status_t StreamOutHalLocal::write(const void *buffer, size_t bytes, size_t *written) {
+    ssize_t writeResult = mStream->write(mStream, buffer, bytes);
+    if (writeResult > 0) {
+        *written = writeResult;
+        mStreamPowerLog.log(buffer, *written);
+        return OK;
+    } else {
+        *written = 0;
+        return writeResult;
+    }
+}
+
+status_t StreamOutHalLocal::getRenderPosition(uint32_t *dspFrames) {
+    return mStream->get_render_position(mStream, dspFrames);
+}
+
+status_t StreamOutHalLocal::getNextWriteTimestamp(int64_t *timestamp) {
+    if (mStream->get_next_write_timestamp == NULL) return INVALID_OPERATION;
+    return mStream->get_next_write_timestamp(mStream, timestamp);
+}
+
+status_t StreamOutHalLocal::setCallback(wp<StreamOutHalInterfaceCallback> callback) {
+    if (mStream->set_callback == NULL) return INVALID_OPERATION;
+    status_t result = mStream->set_callback(mStream, StreamOutHalLocal::asyncCallback, this);
+    if (result == OK) {
+        mCallback = callback;
+    }
+    return result;
+}
+
+// static
+int StreamOutHalLocal::asyncCallback(stream_callback_event_t event, void*, void *cookie) {
+    // We act as if we gave a wp<StreamOutHalLocal> to HAL. This way we should handle
+    // correctly the case when the callback is invoked while StreamOutHalLocal's destructor is
+    // already running, because the destructor is invoked after the refcount has been atomically
+    // decremented.
+    wp<StreamOutHalLocal> weakSelf(static_cast<StreamOutHalLocal*>(cookie));
+    sp<StreamOutHalLocal> self = weakSelf.promote();
+    if (self == 0) return 0;
+    sp<StreamOutHalInterfaceCallback> callback = self->mCallback.promote();
+    if (callback == 0) return 0;
+    ALOGV("asyncCallback() event %d", event);
+    switch (event) {
+        case STREAM_CBK_EVENT_WRITE_READY:
+            callback->onWriteReady();
+            break;
+        case STREAM_CBK_EVENT_DRAIN_READY:
+            callback->onDrainReady();
+            break;
+        case STREAM_CBK_EVENT_ERROR:
+            callback->onError();
+            break;
+        default:
+            ALOGW("asyncCallback() unknown event %d", event);
+            break;
+    }
+    return 0;
+}
+
+status_t StreamOutHalLocal::supportsPauseAndResume(bool *supportsPause, bool *supportsResume) {
+    *supportsPause = mStream->pause != NULL;
+    *supportsResume = mStream->resume != NULL;
+    return OK;
+}
+
+status_t StreamOutHalLocal::pause() {
+    if (mStream->pause == NULL) return INVALID_OPERATION;
+    return mStream->pause(mStream);
+}
+
+status_t StreamOutHalLocal::resume() {
+    if (mStream->resume == NULL) return INVALID_OPERATION;
+    return mStream->resume(mStream);
+}
+
+status_t StreamOutHalLocal::supportsDrain(bool *supportsDrain) {
+    *supportsDrain = mStream->drain != NULL;
+    return OK;
+}
+
+status_t StreamOutHalLocal::drain(bool earlyNotify) {
+    if (mStream->drain == NULL) return INVALID_OPERATION;
+    return mStream->drain(mStream, earlyNotify ? AUDIO_DRAIN_EARLY_NOTIFY : AUDIO_DRAIN_ALL);
+}
+
+status_t StreamOutHalLocal::flush() {
+    if (mStream->flush == NULL) return INVALID_OPERATION;
+    return mStream->flush(mStream);
+}
+
+status_t StreamOutHalLocal::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) {
+    if (mStream->get_presentation_position == NULL) return INVALID_OPERATION;
+    return mStream->get_presentation_position(mStream, frames, timestamp);
+}
+
+status_t StreamOutHalLocal::updateSourceMetadata(const SourceMetadata& sourceMetadata) {
+    if (mStream->update_source_metadata == nullptr) {
+        return INVALID_OPERATION;
+    }
+    const source_metadata_t metadata {
+        .track_count = sourceMetadata.tracks.size(),
+        // const cast is fine as it is in a const structure
+        .tracks = const_cast<playback_track_metadata*>(sourceMetadata.tracks.data()),
+    };
+    mStream->update_source_metadata(mStream, &metadata);
+    return OK;
+}
+
+status_t StreamOutHalLocal::start() {
+    if (mStream->start == NULL) return INVALID_OPERATION;
+    return mStream->start(mStream);
+}
+
+status_t StreamOutHalLocal::stop() {
+    if (mStream->stop == NULL) return INVALID_OPERATION;
+    return mStream->stop(mStream);
+}
+
+status_t StreamOutHalLocal::createMmapBuffer(int32_t minSizeFrames,
+                                  struct audio_mmap_buffer_info *info) {
+    if (mStream->create_mmap_buffer == NULL) return INVALID_OPERATION;
+    return mStream->create_mmap_buffer(mStream, minSizeFrames, info);
+}
+
+status_t StreamOutHalLocal::getMmapPosition(struct audio_mmap_position *position) {
+    if (mStream->get_mmap_position == NULL) return INVALID_OPERATION;
+    return mStream->get_mmap_position(mStream, position);
+}
+
+StreamInHalLocal::StreamInHalLocal(audio_stream_in_t *stream, sp<DeviceHalLocal> device)
+        : StreamHalLocal(&stream->common, device), mStream(stream) {
+}
+
+StreamInHalLocal::~StreamInHalLocal() {
+    mDevice->closeInputStream(mStream);
+    mStream = 0;
+}
+
+status_t StreamInHalLocal::getFrameSize(size_t *size) {
+    *size = audio_stream_in_frame_size(mStream);
+    return OK;
+}
+
+status_t StreamInHalLocal::setGain(float gain) {
+    return mStream->set_gain(mStream, gain);
+}
+
+status_t StreamInHalLocal::read(void *buffer, size_t bytes, size_t *read) {
+    ssize_t readResult = mStream->read(mStream, buffer, bytes);
+    if (readResult > 0) {
+        *read = readResult;
+        mStreamPowerLog.log( buffer, *read);
+        return OK;
+    } else {
+        *read = 0;
+        return readResult;
+    }
+}
+
+status_t StreamInHalLocal::getInputFramesLost(uint32_t *framesLost) {
+    *framesLost = mStream->get_input_frames_lost(mStream);
+    return OK;
+}
+
+status_t StreamInHalLocal::getCapturePosition(int64_t *frames, int64_t *time) {
+    if (mStream->get_capture_position == NULL) return INVALID_OPERATION;
+    return mStream->get_capture_position(mStream, frames, time);
+}
+
+status_t StreamInHalLocal::updateSinkMetadata(const SinkMetadata& sinkMetadata) {
+    if (mStream->update_sink_metadata == nullptr) {
+        return INVALID_OPERATION;
+    }
+    const sink_metadata_t metadata {
+        .track_count = sinkMetadata.tracks.size(),
+        // const cast is fine as it is in a const structure
+        .tracks = const_cast<record_track_metadata*>(sinkMetadata.tracks.data()),
+    };
+    mStream->update_sink_metadata(mStream, &metadata);
+    return OK;
+}
+
+status_t StreamInHalLocal::start() {
+    if (mStream->start == NULL) return INVALID_OPERATION;
+    return mStream->start(mStream);
+}
+
+status_t StreamInHalLocal::stop() {
+    if (mStream->stop == NULL) return INVALID_OPERATION;
+    return mStream->stop(mStream);
+}
+
+status_t StreamInHalLocal::createMmapBuffer(int32_t minSizeFrames,
+                                  struct audio_mmap_buffer_info *info) {
+    if (mStream->create_mmap_buffer == NULL) return INVALID_OPERATION;
+    return mStream->create_mmap_buffer(mStream, minSizeFrames, info);
+}
+
+status_t StreamInHalLocal::getMmapPosition(struct audio_mmap_position *position) {
+    if (mStream->get_mmap_position == NULL) return INVALID_OPERATION;
+    return mStream->get_mmap_position(mStream, position);
+}
+
+status_t StreamInHalLocal::getActiveMicrophones(
+        std::vector<media::MicrophoneInfo> *microphones __unused) {
+    return INVALID_OPERATION;
+}
+
+} // namespace android

diff --git a/media/libaudiohal/2.0/StreamHalLocal.h b/media/libaudiohal/2.0/StreamHalLocal.h
new file mode 100644
index 0000000..cda8d0c
--- /dev/null
+++ b/media/libaudiohal/2.0/StreamHalLocal.h

@@ -0,0 +1,219 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_STREAM_HAL_LOCAL_H
+#define ANDROID_HARDWARE_STREAM_HAL_LOCAL_H
+
+#include <media/audiohal/StreamHalInterface.h>
+#include "StreamPowerLog.h"
+
+namespace android {
+
+class DeviceHalLocal;
+
+class StreamHalLocal : public virtual StreamHalInterface
+{
+  public:
+    // Return the sampling rate in Hz - eg. 44100.
+    virtual status_t getSampleRate(uint32_t *rate);
+
+    // Return size of input/output buffer in bytes for this stream - eg. 4800.
+    virtual status_t getBufferSize(size_t *size);
+
+    // Return the channel mask.
+    virtual status_t getChannelMask(audio_channel_mask_t *mask);
+
+    // Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT.
+    virtual status_t getFormat(audio_format_t *format);
+
+    // Convenience method.
+    virtual status_t getAudioProperties(
+            uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format);
+
+    // Set audio stream parameters.
+    virtual status_t setParameters(const String8& kvPairs);
+
+    // Get audio stream parameters.
+    virtual status_t getParameters(const String8& keys, String8 *values);
+
+    // Add or remove the effect on the stream.
+    virtual status_t addEffect(sp<EffectHalInterface> effect);
+    virtual status_t removeEffect(sp<EffectHalInterface> effect);
+
+    // Put the audio hardware input/output into standby mode.
+    virtual status_t standby();
+
+    virtual status_t dump(int fd);
+
+    // Start a stream operating in mmap mode.
+    virtual status_t start() = 0;
+
+    // Stop a stream operating in mmap mode.
+    virtual status_t stop() = 0;
+
+    // Retrieve information on the data buffer in mmap mode.
+    virtual status_t createMmapBuffer(int32_t minSizeFrames,
+                                      struct audio_mmap_buffer_info *info) = 0;
+
+    // Get current read/write position in the mmap buffer
+    virtual status_t getMmapPosition(struct audio_mmap_position *position) = 0;
+
+    // Set the priority of the thread that interacts with the HAL
+    // (must match the priority of the audioflinger's thread that calls 'read' / 'write')
+    virtual status_t setHalThreadPriority(int priority);
+
+  protected:
+    // Subclasses can not be constructed directly by clients.
+    StreamHalLocal(audio_stream_t *stream, sp<DeviceHalLocal> device);
+
+    // The destructor automatically closes the stream.
+    virtual ~StreamHalLocal();
+
+    sp<DeviceHalLocal> mDevice;
+
+    // mStreamPowerLog is used for audio signal power logging.
+    StreamPowerLog mStreamPowerLog;
+
+  private:
+    audio_stream_t *mStream;
+};
+
+class StreamOutHalLocal : public StreamOutHalInterface, public StreamHalLocal {
+  public:
+    // Return the frame size (number of bytes per sample) of a stream.
+    virtual status_t getFrameSize(size_t *size);
+
+    // Return the audio hardware driver estimated latency in milliseconds.
+    virtual status_t getLatency(uint32_t *latency);
+
+    // Use this method in situations where audio mixing is done in the hardware.
+    virtual status_t setVolume(float left, float right);
+
+    // Write audio buffer to driver.
+    virtual status_t write(const void *buffer, size_t bytes, size_t *written);
+
+    // Return the number of audio frames written by the audio dsp to DAC since
+    // the output has exited standby.
+    virtual status_t getRenderPosition(uint32_t *dspFrames);
+
+    // Get the local time at which the next write to the audio driver will be presented.
+    virtual status_t getNextWriteTimestamp(int64_t *timestamp);
+
+    // Set the callback for notifying completion of non-blocking write and drain.
+    virtual status_t setCallback(wp<StreamOutHalInterfaceCallback> callback);
+
+    // Returns whether pause and resume operations are supported.
+    virtual status_t supportsPauseAndResume(bool *supportsPause, bool *supportsResume);
+
+    // Notifies to the audio driver to resume playback following a pause.
+    virtual status_t pause();
+
+    // Notifies to the audio driver to resume playback following a pause.
+    virtual status_t resume();
+
+    // Returns whether drain operation is supported.
+    virtual status_t supportsDrain(bool *supportsDrain);
+
+    // Requests notification when data buffered by the driver/hardware has been played.
+    virtual status_t drain(bool earlyNotify);
+
+    // Notifies to the audio driver to flush the queued data.
+    virtual status_t flush();
+
+    // Return a recent count of the number of audio frames presented to an external observer.
+    virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
+
+    // Start a stream operating in mmap mode.
+    virtual status_t start();
+
+    // Stop a stream operating in mmap mode.
+    virtual status_t stop();
+
+    // Retrieve information on the data buffer in mmap mode.
+    virtual status_t createMmapBuffer(int32_t minSizeFrames,
+                                      struct audio_mmap_buffer_info *info);
+
+    // Get current read/write position in the mmap buffer
+    virtual status_t getMmapPosition(struct audio_mmap_position *position);
+
+    // Called when the metadata of the stream's source has been changed.
+    status_t updateSourceMetadata(const SourceMetadata& sourceMetadata) override;
+
+  private:
+    audio_stream_out_t *mStream;
+    wp<StreamOutHalInterfaceCallback> mCallback;
+
+    friend class DeviceHalLocal;
+
+    // Can not be constructed directly by clients.
+    StreamOutHalLocal(audio_stream_out_t *stream, sp<DeviceHalLocal> device);
+
+    virtual ~StreamOutHalLocal();
+
+    static int asyncCallback(stream_callback_event_t event, void *param, void *cookie);
+};
+
+class StreamInHalLocal : public StreamInHalInterface, public StreamHalLocal {
+  public:
+    // Return the frame size (number of bytes per sample) of a stream.
+    virtual status_t getFrameSize(size_t *size);
+
+    // Set the input gain for the audio driver.
+    virtual status_t setGain(float gain);
+
+    // Read audio buffer in from driver.
+    virtual status_t read(void *buffer, size_t bytes, size_t *read);
+
+    // Return the amount of input frames lost in the audio driver.
+    virtual status_t getInputFramesLost(uint32_t *framesLost);
+
+    // Return a recent count of the number of audio frames received and
+    // the clock time associated with that frame count.
+    virtual status_t getCapturePosition(int64_t *frames, int64_t *time);
+
+    // Start a stream operating in mmap mode.
+    virtual status_t start();
+
+    // Stop a stream operating in mmap mode.
+    virtual status_t stop();
+
+    // Retrieve information on the data buffer in mmap mode.
+    virtual status_t createMmapBuffer(int32_t minSizeFrames,
+                                      struct audio_mmap_buffer_info *info);
+
+    // Get current read/write position in the mmap buffer
+    virtual status_t getMmapPosition(struct audio_mmap_position *position);
+
+    // Get active microphones
+    virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+
+    // Called when the metadata of the stream's sink has been changed.
+    status_t updateSinkMetadata(const SinkMetadata& sinkMetadata) override;
+
+  private:
+    audio_stream_in_t *mStream;
+
+    friend class DeviceHalLocal;
+
+    // Can not be constructed directly by clients.
+    StreamInHalLocal(audio_stream_in_t *stream, sp<DeviceHalLocal> device);
+
+    virtual ~StreamInHalLocal();
+};
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_STREAM_HAL_LOCAL_H

diff --git a/media/libaudiohal/StreamPowerLog.h b/media/libaudiohal/2.0/StreamPowerLog.h
similarity index 100%
rename from media/libaudiohal/StreamPowerLog.h
rename to media/libaudiohal/2.0/StreamPowerLog.h


diff --git a/media/libaudiohal/4.0/Android.bp b/media/libaudiohal/4.0/Android.bp
new file mode 100644
index 0000000..833defa
--- /dev/null
+++ b/media/libaudiohal/4.0/Android.bp

@@ -0,0 +1,58 @@
+cc_library_shared {
+    name: "libaudiohal@4.0",
+
+    srcs: [
+        "DeviceHalLocal.cpp",
+        "DevicesFactoryHalHybrid.cpp",
+        "DevicesFactoryHalLocal.cpp",
+        "StreamHalLocal.cpp",
+
+        "ConversionHelperHidl.cpp",
+        "DeviceHalHidl.cpp",
+        "DevicesFactoryHalHidl.cpp",
+        "EffectBufferHalHidl.cpp",
+        "EffectHalHidl.cpp",
+        "EffectsFactoryHalHidl.cpp",
+        "StreamHalHidl.cpp",
+    ],
+
+    export_include_dirs: ["include"],
+
+    cflags: [
+        "-Wall",
+        "-Wextra",
+        "-Werror",
+    ],
+    shared_libs: [
+        "libaudiohal_deathhandler",
+        "libaudioutils",
+        "libbinder",
+        "libcutils",
+        "liblog",
+        "libutils",
+        "libhardware",
+        "libbase",
+        "libfmq",
+        "libhwbinder",
+        "libhidlbase",
+        "libhidlmemory",
+        "libhidltransport",
+        "android.hardware.audio@4.0",
+        "android.hardware.audio.common-util",
+        "android.hardware.audio.common@4.0",
+        "android.hardware.audio.common@4.0-util",
+        "android.hardware.audio.effect@4.0",
+        "android.hidl.allocator@1.0",
+        "android.hidl.memory@1.0",
+        "libmedia_helper",
+        "libmediautils",
+    ],
+    header_libs: [
+        "android.hardware.audio.common.util@all-versions",
+        "libaudiohal_headers"
+    ],
+
+    export_shared_lib_headers: [
+        "libfmq",
+    ],
+}

diff --git a/media/libaudiohal/4.0/ConversionHelperHidl.cpp b/media/libaudiohal/4.0/ConversionHelperHidl.cpp
new file mode 100644
index 0000000..fe27504
--- /dev/null
+++ b/media/libaudiohal/4.0/ConversionHelperHidl.cpp

@@ -0,0 +1,237 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <string.h>
+
+#define LOG_TAG "HalHidl"
+#include <media/AudioParameter.h>
+#include <utils/Log.h>
+
+#include "ConversionHelperHidl.h"
+
+using ::android::hardware::audio::V4_0::AudioMicrophoneChannelMapping;
+using ::android::hardware::audio::V4_0::AudioMicrophoneDirectionality;
+using ::android::hardware::audio::V4_0::AudioMicrophoneLocation;
+using ::android::hardware::audio::V4_0::DeviceAddress;
+using ::android::hardware::audio::V4_0::MicrophoneInfo;
+using ::android::hardware::audio::V4_0::Result;
+
+namespace android {
+namespace V4_0 {
+
+// static
+status_t ConversionHelperHidl::keysFromHal(const String8& keys, hidl_vec<hidl_string> *hidlKeys) {
+    AudioParameter halKeys(keys);
+    if (halKeys.size() == 0) return BAD_VALUE;
+    hidlKeys->resize(halKeys.size());
+    //FIXME:  keyStreamSupportedChannels and keyStreamSupportedSamplingRates come with a
+    // "keyFormat=<value>" pair. We need to transform it into a single key string so that it is
+    // carried over to the legacy HAL via HIDL.
+    String8 value;
+    bool keepFormatValue = halKeys.size() == 2 &&
+         (halKeys.get(String8(AudioParameter::keyStreamSupportedChannels), value) == NO_ERROR ||
+         halKeys.get(String8(AudioParameter::keyStreamSupportedSamplingRates), value) == NO_ERROR);
+
+    for (size_t i = 0; i < halKeys.size(); ++i) {
+        String8 key;
+        status_t status = halKeys.getAt(i, key);
+        if (status != OK) return status;
+        if (keepFormatValue && key == AudioParameter::keyFormat) {
+            AudioParameter formatParam;
+            halKeys.getAt(i, key, value);
+            formatParam.add(key, value);
+            key = formatParam.toString();
+        }
+        (*hidlKeys)[i] = key.string();
+    }
+    return OK;
+}
+
+// static
+status_t ConversionHelperHidl::parametersFromHal(
+        const String8& kvPairs, hidl_vec<ParameterValue> *hidlParams) {
+    AudioParameter params(kvPairs);
+    if (params.size() == 0) return BAD_VALUE;
+    hidlParams->resize(params.size());
+    for (size_t i = 0; i < params.size(); ++i) {
+        String8 key, value;
+        status_t status = params.getAt(i, key, value);
+        if (status != OK) return status;
+        (*hidlParams)[i].key = key.string();
+        (*hidlParams)[i].value = value.string();
+    }
+    return OK;
+}
+
+// static
+void ConversionHelperHidl::parametersToHal(
+        const hidl_vec<ParameterValue>& parameters, String8 *values) {
+    AudioParameter params;
+    for (size_t i = 0; i < parameters.size(); ++i) {
+        params.add(String8(parameters[i].key.c_str()), String8(parameters[i].value.c_str()));
+    }
+    values->setTo(params.toString());
+}
+
+ConversionHelperHidl::ConversionHelperHidl(const char* className)
+        : mClassName(className) {
+}
+
+// static
+status_t ConversionHelperHidl::analyzeResult(const Result& result) {
+    switch (result) {
+        case Result::OK: return OK;
+        case Result::INVALID_ARGUMENTS: return BAD_VALUE;
+        case Result::INVALID_STATE: return NOT_ENOUGH_DATA;
+        case Result::NOT_INITIALIZED: return NO_INIT;
+        case Result::NOT_SUPPORTED: return INVALID_OPERATION;
+        default: return NO_INIT;
+    }
+}
+
+void ConversionHelperHidl::emitError(const char* funcName, const char* description) {
+    ALOGE("%s %p %s: %s (from rpc)", mClassName, this, funcName, description);
+}
+
+// TODO: Use the same implementation in the hal when it moves to a util library.
+std::string deviceAddressToHal(const DeviceAddress& address) {
+    // HAL assumes that the address is NUL-terminated.
+    char halAddress[AUDIO_DEVICE_MAX_ADDRESS_LEN];
+    memset(halAddress, 0, sizeof(halAddress));
+    audio_devices_t halDevice = static_cast<audio_devices_t>(address.device);
+    const bool isInput = (halDevice & AUDIO_DEVICE_BIT_IN) != 0;
+    if (isInput) halDevice &= ~AUDIO_DEVICE_BIT_IN;
+    if ((!isInput && (halDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) ||
+        (isInput && (halDevice & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) != 0)) {
+        snprintf(halAddress, sizeof(halAddress), "%02X:%02X:%02X:%02X:%02X:%02X",
+                 address.address.mac[0], address.address.mac[1], address.address.mac[2],
+                 address.address.mac[3], address.address.mac[4], address.address.mac[5]);
+    } else if ((!isInput && (halDevice & AUDIO_DEVICE_OUT_IP) != 0) ||
+               (isInput && (halDevice & AUDIO_DEVICE_IN_IP) != 0)) {
+        snprintf(halAddress, sizeof(halAddress), "%d.%d.%d.%d", address.address.ipv4[0],
+                 address.address.ipv4[1], address.address.ipv4[2], address.address.ipv4[3]);
+    } else if ((!isInput && (halDevice & AUDIO_DEVICE_OUT_ALL_USB) != 0) ||
+               (isInput && (halDevice & AUDIO_DEVICE_IN_ALL_USB) != 0)) {
+        snprintf(halAddress, sizeof(halAddress), "card=%d;device=%d", address.address.alsa.card,
+                 address.address.alsa.device);
+    } else if ((!isInput && (halDevice & AUDIO_DEVICE_OUT_BUS) != 0) ||
+               (isInput && (halDevice & AUDIO_DEVICE_IN_BUS) != 0)) {
+        snprintf(halAddress, sizeof(halAddress), "%s", address.busAddress.c_str());
+    } else if ((!isInput && (halDevice & AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) != 0 ||
+               (isInput && (halDevice & AUDIO_DEVICE_IN_REMOTE_SUBMIX) != 0)) {
+        snprintf(halAddress, sizeof(halAddress), "%s", address.rSubmixAddress.c_str());
+    } else {
+        snprintf(halAddress, sizeof(halAddress), "%s", address.busAddress.c_str());
+    }
+    return halAddress;
+}
+
+//local conversion helpers
+
+audio_microphone_channel_mapping_t  channelMappingToHal(AudioMicrophoneChannelMapping mapping) {
+    switch (mapping) {
+        case AudioMicrophoneChannelMapping::UNUSED:
+            return AUDIO_MICROPHONE_CHANNEL_MAPPING_UNUSED;
+        case AudioMicrophoneChannelMapping::DIRECT:
+            return AUDIO_MICROPHONE_CHANNEL_MAPPING_DIRECT;
+        case AudioMicrophoneChannelMapping::PROCESSED:
+            return AUDIO_MICROPHONE_CHANNEL_MAPPING_PROCESSED;
+        default:
+            LOG_ALWAYS_FATAL("Unknown channelMappingToHal conversion %d", mapping);
+    }
+}
+
+audio_microphone_location_t locationToHal(AudioMicrophoneLocation location) {
+    switch (location) {
+        case AudioMicrophoneLocation::UNKNOWN:
+            return AUDIO_MICROPHONE_LOCATION_UNKNOWN;
+        case AudioMicrophoneLocation::MAINBODY:
+            return AUDIO_MICROPHONE_LOCATION_MAINBODY;
+        case AudioMicrophoneLocation::MAINBODY_MOVABLE:
+            return AUDIO_MICROPHONE_LOCATION_MAINBODY_MOVABLE;
+        case AudioMicrophoneLocation::PERIPHERAL:
+            return AUDIO_MICROPHONE_LOCATION_PERIPHERAL;
+        default:
+            LOG_ALWAYS_FATAL("Unknown locationToHal conversion %d", location);
+    }
+}
+audio_microphone_directionality_t directionalityToHal(AudioMicrophoneDirectionality dir) {
+    switch (dir) {
+        case AudioMicrophoneDirectionality::UNKNOWN:
+            return AUDIO_MICROPHONE_DIRECTIONALITY_UNKNOWN;
+        case AudioMicrophoneDirectionality::OMNI:
+            return AUDIO_MICROPHONE_DIRECTIONALITY_OMNI;
+        case AudioMicrophoneDirectionality::BI_DIRECTIONAL:
+            return AUDIO_MICROPHONE_DIRECTIONALITY_BI_DIRECTIONAL;
+        case AudioMicrophoneDirectionality::CARDIOID:
+            return AUDIO_MICROPHONE_DIRECTIONALITY_CARDIOID;
+        case AudioMicrophoneDirectionality::HYPER_CARDIOID:
+            return AUDIO_MICROPHONE_DIRECTIONALITY_HYPER_CARDIOID;
+        case AudioMicrophoneDirectionality::SUPER_CARDIOID:
+            return AUDIO_MICROPHONE_DIRECTIONALITY_SUPER_CARDIOID;
+        default:
+            LOG_ALWAYS_FATAL("Unknown directionalityToHal conversion %d", dir);
+    }
+}
+
+// static
+void ConversionHelperHidl::microphoneInfoToHal(const MicrophoneInfo& src,
+                                                     audio_microphone_characteristic_t *pDst) {
+    if (pDst != NULL) {
+        snprintf(pDst->device_id, sizeof(pDst->device_id),
+                 "%s", src.deviceId.c_str());
+        pDst->device = static_cast<audio_devices_t>(src.deviceAddress.device);
+        snprintf(pDst->address, sizeof(pDst->address),
+                 "%s", deviceAddressToHal(src.deviceAddress).c_str());
+        if (src.channelMapping.size() > AUDIO_CHANNEL_COUNT_MAX) {
+            ALOGW("microphoneInfoToStruct found %zu channelMapping elements. Max expected is %d",
+                  src.channelMapping.size(), AUDIO_CHANNEL_COUNT_MAX);
+        }
+        size_t ch;
+        for (ch = 0; ch < src.channelMapping.size() && ch < AUDIO_CHANNEL_COUNT_MAX; ch++) {
+            pDst->channel_mapping[ch] = channelMappingToHal(src.channelMapping[ch]);
+        }
+        for (; ch < AUDIO_CHANNEL_COUNT_MAX; ch++) {
+            pDst->channel_mapping[ch] = AUDIO_MICROPHONE_CHANNEL_MAPPING_UNUSED;
+        }
+        pDst->location = locationToHal(src.location);
+        pDst->group = (audio_microphone_group_t)src.group;
+        pDst->index_in_the_group = (unsigned int)src.indexInTheGroup;
+        pDst->sensitivity = src.sensitivity;
+        pDst->max_spl = src.maxSpl;
+        pDst->min_spl = src.minSpl;
+        pDst->directionality = directionalityToHal(src.directionality);
+        pDst->num_frequency_responses = (unsigned int)src.frequencyResponse.size();
+        if (pDst->num_frequency_responses > AUDIO_MICROPHONE_MAX_FREQUENCY_RESPONSES) {
+            ALOGW("microphoneInfoToStruct found %d frequency responses. Max expected is %d",
+                  pDst->num_frequency_responses, AUDIO_MICROPHONE_MAX_FREQUENCY_RESPONSES);
+            pDst->num_frequency_responses = AUDIO_MICROPHONE_MAX_FREQUENCY_RESPONSES;
+        }
+        for (size_t k = 0; k < pDst->num_frequency_responses; k++) {
+            pDst->frequency_responses[0][k] = src.frequencyResponse[k].frequency;
+            pDst->frequency_responses[1][k] = src.frequencyResponse[k].level;
+        }
+        pDst->geometric_location.x = src.position.x;
+        pDst->geometric_location.y = src.position.y;
+        pDst->geometric_location.z = src.position.z;
+        pDst->orientation.x = src.orientation.x;
+        pDst->orientation.y = src.orientation.y;
+        pDst->orientation.z = src.orientation.z;
+    }
+}
+
+}  // namespace V4_0
+}  // namespace android

diff --git a/media/libaudiohal/4.0/ConversionHelperHidl.h b/media/libaudiohal/4.0/ConversionHelperHidl.h
new file mode 100644
index 0000000..8823a8d
--- /dev/null
+++ b/media/libaudiohal/4.0/ConversionHelperHidl.h

@@ -0,0 +1,89 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_CONVERSION_HELPER_HIDL_4_0_H
+#define ANDROID_HARDWARE_CONVERSION_HELPER_HIDL_4_0_H
+
+#include <android/hardware/audio/4.0/types.h>
+#include <hidl/HidlSupport.h>
+#include <system/audio.h>
+#include <utils/String8.h>
+
+using ::android::hardware::audio::V4_0::ParameterValue;
+using ::android::hardware::audio::V4_0::MicrophoneInfo;
+using ::android::hardware::Return;
+using ::android::hardware::hidl_string;
+using ::android::hardware::hidl_vec;
+
+namespace android {
+namespace V4_0 {
+
+class ConversionHelperHidl {
+  protected:
+    static status_t keysFromHal(const String8& keys, hidl_vec<hidl_string> *hidlKeys);
+    static status_t parametersFromHal(const String8& kvPairs, hidl_vec<ParameterValue> *hidlParams);
+    static void parametersToHal(const hidl_vec<ParameterValue>& parameters, String8 *values);
+    static void microphoneInfoToHal(const MicrophoneInfo& src,
+                                    audio_microphone_characteristic_t *pDst);
+
+    ConversionHelperHidl(const char* className);
+
+    template<typename R, typename T>
+    status_t processReturn(const char* funcName, const Return<R>& ret, T *retval) {
+        if (ret.isOk()) {
+            // This way it also works for enum class to unscoped enum conversion.
+            *retval = static_cast<T>(static_cast<R>(ret));
+            return OK;
+        }
+        return processReturn(funcName, ret);
+    }
+
+    template<typename T>
+    status_t processReturn(const char* funcName, const Return<T>& ret) {
+        if (!ret.isOk()) {
+            emitError(funcName, ret.description().c_str());
+        }
+        return ret.isOk() ? OK : FAILED_TRANSACTION;
+    }
+
+    status_t processReturn(const char* funcName, const Return<hardware::audio::V4_0::Result>& ret) {
+        if (!ret.isOk()) {
+            emitError(funcName, ret.description().c_str());
+        }
+        return ret.isOk() ? analyzeResult(ret) : FAILED_TRANSACTION;
+    }
+
+    template<typename T>
+    status_t processReturn(
+            const char* funcName, const Return<T>& ret, hardware::audio::V4_0::Result retval) {
+        if (!ret.isOk()) {
+            emitError(funcName, ret.description().c_str());
+        }
+        return ret.isOk() ? analyzeResult(retval) : FAILED_TRANSACTION;
+    }
+
+  private:
+    const char* mClassName;
+
+    static status_t analyzeResult(const hardware::audio::V4_0::Result& result);
+
+    void emitError(const char* funcName, const char* description);
+};
+
+}  // namespace V4_0
+}  // namespace android
+
+#endif // ANDROID_HARDWARE_CONVERSION_HELPER_HIDL_4_0_H

diff --git a/media/libaudiohal/4.0/DeviceHalHidl.cpp b/media/libaudiohal/4.0/DeviceHalHidl.cpp
new file mode 100644
index 0000000..6facca9
--- /dev/null
+++ b/media/libaudiohal/4.0/DeviceHalHidl.cpp

@@ -0,0 +1,389 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <stdio.h>
+
+#define LOG_TAG "DeviceHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <android/hardware/audio/4.0/IPrimaryDevice.h>
+#include <cutils/native_handle.h>
+#include <hwbinder/IPCThreadState.h>
+#include <utils/Log.h>
+
+#include <common/all-versions/VersionUtils.h>
+
+#include "DeviceHalHidl.h"
+#include "HidlUtils.h"
+#include "StreamHalHidl.h"
+#include "VersionUtils.h"
+
+using ::android::hardware::audio::common::V4_0::AudioConfig;
+using ::android::hardware::audio::common::V4_0::AudioDevice;
+using ::android::hardware::audio::common::V4_0::AudioInputFlag;
+using ::android::hardware::audio::common::V4_0::AudioOutputFlag;
+using ::android::hardware::audio::common::V4_0::AudioPatchHandle;
+using ::android::hardware::audio::common::V4_0::AudioPort;
+using ::android::hardware::audio::common::V4_0::AudioPortConfig;
+using ::android::hardware::audio::common::V4_0::AudioMode;
+using ::android::hardware::audio::common::V4_0::AudioSource;
+using ::android::hardware::audio::common::V4_0::HidlUtils;
+using ::android::hardware::audio::common::utils::mkEnumConverter;
+using ::android::hardware::audio::V4_0::DeviceAddress;
+using ::android::hardware::audio::V4_0::IPrimaryDevice;
+using ::android::hardware::audio::V4_0::ParameterValue;
+using ::android::hardware::audio::V4_0::Result;
+using ::android::hardware::audio::V4_0::SinkMetadata;
+using ::android::hardware::hidl_string;
+using ::android::hardware::hidl_vec;
+
+namespace android {
+namespace V4_0 {
+
+namespace {
+
+status_t deviceAddressFromHal(
+        audio_devices_t device, const char* halAddress, DeviceAddress* address) {
+    address->device = AudioDevice(device);
+
+    if (halAddress == nullptr || strnlen(halAddress, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) {
+        return OK;
+    }
+    const bool isInput = (device & AUDIO_DEVICE_BIT_IN) != 0;
+    if (isInput) device &= ~AUDIO_DEVICE_BIT_IN;
+    if ((!isInput && (device & AUDIO_DEVICE_OUT_ALL_A2DP) != 0)
+            || (isInput && (device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) != 0)) {
+        int status = sscanf(halAddress,
+                "%hhX:%hhX:%hhX:%hhX:%hhX:%hhX",
+                &address->address.mac[0], &address->address.mac[1], &address->address.mac[2],
+                &address->address.mac[3], &address->address.mac[4], &address->address.mac[5]);
+        return status == 6 ? OK : BAD_VALUE;
+    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_IP) != 0)
+            || (isInput && (device & AUDIO_DEVICE_IN_IP) != 0)) {
+        int status = sscanf(halAddress,
+                "%hhu.%hhu.%hhu.%hhu",
+                &address->address.ipv4[0], &address->address.ipv4[1],
+                &address->address.ipv4[2], &address->address.ipv4[3]);
+        return status == 4 ? OK : BAD_VALUE;
+    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_ALL_USB)) != 0
+            || (isInput && (device & AUDIO_DEVICE_IN_ALL_USB)) != 0) {
+        int status = sscanf(halAddress,
+                "card=%d;device=%d",
+                &address->address.alsa.card, &address->address.alsa.device);
+        return status == 2 ? OK : BAD_VALUE;
+    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_BUS) != 0)
+            || (isInput && (device & AUDIO_DEVICE_IN_BUS) != 0)) {
+        if (halAddress != NULL) {
+            address->busAddress = halAddress;
+            return OK;
+        }
+        return BAD_VALUE;
+    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) != 0
+            || (isInput && (device & AUDIO_DEVICE_IN_REMOTE_SUBMIX) != 0)) {
+        if (halAddress != NULL) {
+            address->rSubmixAddress = halAddress;
+            return OK;
+        }
+        return BAD_VALUE;
+    }
+    return OK;
+}
+
+}  // namespace
+
+DeviceHalHidl::DeviceHalHidl(const sp<IDevice>& device)
+        : ConversionHelperHidl("Device"), mDevice(device),
+          mPrimaryDevice(IPrimaryDevice::castFrom(device)) {
+}
+
+DeviceHalHidl::~DeviceHalHidl() {
+    if (mDevice != 0) {
+        mDevice.clear();
+        hardware::IPCThreadState::self()->flushCommands();
+    }
+}
+
+status_t DeviceHalHidl::getSupportedDevices(uint32_t*) {
+    // Obsolete.
+    return INVALID_OPERATION;
+}
+
+status_t DeviceHalHidl::initCheck() {
+    if (mDevice == 0) return NO_INIT;
+    return processReturn("initCheck", mDevice->initCheck());
+}
+
+status_t DeviceHalHidl::setVoiceVolume(float volume) {
+    if (mDevice == 0) return NO_INIT;
+    if (mPrimaryDevice == 0) return INVALID_OPERATION;
+    return processReturn("setVoiceVolume", mPrimaryDevice->setVoiceVolume(volume));
+}
+
+status_t DeviceHalHidl::setMasterVolume(float volume) {
+    if (mDevice == 0) return NO_INIT;
+    if (mPrimaryDevice == 0) return INVALID_OPERATION;
+    return processReturn("setMasterVolume", mPrimaryDevice->setMasterVolume(volume));
+}
+
+status_t DeviceHalHidl::getMasterVolume(float *volume) {
+    if (mDevice == 0) return NO_INIT;
+    if (mPrimaryDevice == 0) return INVALID_OPERATION;
+    Result retval;
+    Return<void> ret = mPrimaryDevice->getMasterVolume(
+            [&](Result r, float v) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *volume = v;
+                }
+            });
+    return processReturn("getMasterVolume", ret, retval);
+}
+
+status_t DeviceHalHidl::setMode(audio_mode_t mode) {
+    if (mDevice == 0) return NO_INIT;
+    if (mPrimaryDevice == 0) return INVALID_OPERATION;
+    return processReturn("setMode", mPrimaryDevice->setMode(AudioMode(mode)));
+}
+
+status_t DeviceHalHidl::setMicMute(bool state) {
+    if (mDevice == 0) return NO_INIT;
+    return processReturn("setMicMute", mDevice->setMicMute(state));
+}
+
+status_t DeviceHalHidl::getMicMute(bool *state) {
+    if (mDevice == 0) return NO_INIT;
+    Result retval;
+    Return<void> ret = mDevice->getMicMute(
+            [&](Result r, bool mute) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *state = mute;
+                }
+            });
+    return processReturn("getMicMute", ret, retval);
+}
+
+status_t DeviceHalHidl::setMasterMute(bool state) {
+    if (mDevice == 0) return NO_INIT;
+    return processReturn("setMasterMute", mDevice->setMasterMute(state));
+}
+
+status_t DeviceHalHidl::getMasterMute(bool *state) {
+    if (mDevice == 0) return NO_INIT;
+    Result retval;
+    Return<void> ret = mDevice->getMasterMute(
+            [&](Result r, bool mute) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *state = mute;
+                }
+            });
+    return processReturn("getMasterMute", ret, retval);
+}
+
+status_t DeviceHalHidl::setParameters(const String8& kvPairs) {
+    if (mDevice == 0) return NO_INIT;
+    hidl_vec<ParameterValue> hidlParams;
+    status_t status = parametersFromHal(kvPairs, &hidlParams);
+    if (status != OK) return status;
+    // TODO: change the API so that context and kvPairs are separated
+    return processReturn("setParameters",
+                         utils::setParameters(mDevice, {} /* context */, hidlParams));
+}
+
+status_t DeviceHalHidl::getParameters(const String8& keys, String8 *values) {
+    values->clear();
+    if (mDevice == 0) return NO_INIT;
+    hidl_vec<hidl_string> hidlKeys;
+    status_t status = keysFromHal(keys, &hidlKeys);
+    if (status != OK) return status;
+    Result retval;
+    Return<void> ret = utils::getParameters(mDevice,
+            {} /* context */,
+            hidlKeys,
+            [&](Result r, const hidl_vec<ParameterValue>& parameters) {
+                retval = r;
+                if (retval == Result::OK) {
+                    parametersToHal(parameters, values);
+                }
+            });
+    return processReturn("getParameters", ret, retval);
+}
+
+status_t DeviceHalHidl::getInputBufferSize(
+        const struct audio_config *config, size_t *size) {
+    if (mDevice == 0) return NO_INIT;
+    AudioConfig hidlConfig;
+    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
+    Result retval;
+    Return<void> ret = mDevice->getInputBufferSize(
+            hidlConfig,
+            [&](Result r, uint64_t bufferSize) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *size = static_cast<size_t>(bufferSize);
+                }
+            });
+    return processReturn("getInputBufferSize", ret, retval);
+}
+
+status_t DeviceHalHidl::openOutputStream(
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        audio_output_flags_t flags,
+        struct audio_config *config,
+        const char *address,
+        sp<StreamOutHalInterface> *outStream) {
+    if (mDevice == 0) return NO_INIT;
+    DeviceAddress hidlDevice;
+    status_t status = deviceAddressFromHal(devices, address, &hidlDevice);
+    if (status != OK) return status;
+    AudioConfig hidlConfig;
+    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mDevice->openOutputStream(
+            handle,
+            hidlDevice,
+            hidlConfig,
+            mkEnumConverter<AudioOutputFlag>(flags),
+            {} /* metadata */,
+            [&](Result r, const sp<IStreamOut>& result, const AudioConfig& suggestedConfig) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *outStream = new StreamOutHalHidl(result);
+                }
+                HidlUtils::audioConfigToHal(suggestedConfig, config);
+            });
+    return processReturn("openOutputStream", ret, retval);
+}
+
+status_t DeviceHalHidl::openInputStream(
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        struct audio_config *config,
+        audio_input_flags_t flags,
+        const char *address,
+        audio_source_t source,
+        sp<StreamInHalInterface> *inStream) {
+    if (mDevice == 0) return NO_INIT;
+    DeviceAddress hidlDevice;
+    status_t status = deviceAddressFromHal(devices, address, &hidlDevice);
+    if (status != OK) return status;
+    AudioConfig hidlConfig;
+    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
+    Result retval = Result::NOT_INITIALIZED;
+    // TODO: correctly propagate the tracks sources and volume
+    //       for now, only send the main source at 1dbfs
+    SinkMetadata metadata = {{{AudioSource(source), 1}}};
+    Return<void> ret = mDevice->openInputStream(
+            handle,
+            hidlDevice,
+            hidlConfig,
+            flags,
+            metadata,
+            [&](Result r, const sp<IStreamIn>& result, const AudioConfig& suggestedConfig) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *inStream = new StreamInHalHidl(result);
+                }
+                HidlUtils::audioConfigToHal(suggestedConfig, config);
+            });
+    return processReturn("openInputStream", ret, retval);
+}
+
+status_t DeviceHalHidl::supportsAudioPatches(bool *supportsPatches) {
+    if (mDevice == 0) return NO_INIT;
+    return processReturn("supportsAudioPatches", mDevice->supportsAudioPatches(), supportsPatches);
+}
+
+status_t DeviceHalHidl::createAudioPatch(
+        unsigned int num_sources,
+        const struct audio_port_config *sources,
+        unsigned int num_sinks,
+        const struct audio_port_config *sinks,
+        audio_patch_handle_t *patch) {
+    if (mDevice == 0) return NO_INIT;
+    hidl_vec<AudioPortConfig> hidlSources, hidlSinks;
+    HidlUtils::audioPortConfigsFromHal(num_sources, sources, &hidlSources);
+    HidlUtils::audioPortConfigsFromHal(num_sinks, sinks, &hidlSinks);
+    Result retval;
+    Return<void> ret = mDevice->createAudioPatch(
+            hidlSources, hidlSinks,
+            [&](Result r, AudioPatchHandle hidlPatch) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *patch = static_cast<audio_patch_handle_t>(hidlPatch);
+                }
+            });
+    return processReturn("createAudioPatch", ret, retval);
+}
+
+status_t DeviceHalHidl::releaseAudioPatch(audio_patch_handle_t patch) {
+    if (mDevice == 0) return NO_INIT;
+    return processReturn("releaseAudioPatch", mDevice->releaseAudioPatch(patch));
+}
+
+status_t DeviceHalHidl::getAudioPort(struct audio_port *port) {
+    if (mDevice == 0) return NO_INIT;
+    AudioPort hidlPort;
+    HidlUtils::audioPortFromHal(*port, &hidlPort);
+    Result retval;
+    Return<void> ret = mDevice->getAudioPort(
+            hidlPort,
+            [&](Result r, const AudioPort& p) {
+                retval = r;
+                if (retval == Result::OK) {
+                    HidlUtils::audioPortToHal(p, port);
+                }
+            });
+    return processReturn("getAudioPort", ret, retval);
+}
+
+status_t DeviceHalHidl::setAudioPortConfig(const struct audio_port_config *config) {
+    if (mDevice == 0) return NO_INIT;
+    AudioPortConfig hidlConfig;
+    HidlUtils::audioPortConfigFromHal(*config, &hidlConfig);
+    return processReturn("setAudioPortConfig", mDevice->setAudioPortConfig(hidlConfig));
+}
+
+status_t DeviceHalHidl::getMicrophones(std::vector<media::MicrophoneInfo> *microphonesInfo) {
+    if (mDevice == 0) return NO_INIT;
+    Result retval;
+    Return<void> ret = mDevice->getMicrophones(
+            [&](Result r, hidl_vec<MicrophoneInfo> micArrayHal) {
+        retval = r;
+        for (size_t k = 0; k < micArrayHal.size(); k++) {
+            audio_microphone_characteristic_t dst;
+            //convert
+            microphoneInfoToHal(micArrayHal[k], &dst);
+            media::MicrophoneInfo microphone = media::MicrophoneInfo(dst);
+            microphonesInfo->push_back(microphone);
+        }
+    });
+    return processReturn("getMicrophones", ret, retval);
+}
+
+status_t DeviceHalHidl::dump(int fd) {
+    if (mDevice == 0) return NO_INIT;
+    native_handle_t* hidlHandle = native_handle_create(1, 0);
+    hidlHandle->data[0] = fd;
+    Return<void> ret = mDevice->debug(hidlHandle, {} /* options */);
+    native_handle_delete(hidlHandle);
+    return processReturn("dump", ret);
+}
+
+} // namespace V4_0
+} // namespace android

diff --git a/media/libaudiohal/4.0/DeviceHalHidl.h b/media/libaudiohal/4.0/DeviceHalHidl.h
new file mode 100644
index 0000000..0bd2175
--- /dev/null
+++ b/media/libaudiohal/4.0/DeviceHalHidl.h

@@ -0,0 +1,131 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_DEVICE_HAL_HIDL_4_0_H
+#define ANDROID_HARDWARE_DEVICE_HAL_HIDL_4_0_H
+
+#include <android/hardware/audio/4.0/IDevice.h>
+#include <android/hardware/audio/4.0/IPrimaryDevice.h>
+#include <media/audiohal/DeviceHalInterface.h>
+
+#include "ConversionHelperHidl.h"
+
+using ::android::hardware::audio::V4_0::IDevice;
+using ::android::hardware::audio::V4_0::IPrimaryDevice;
+using ::android::hardware::Return;
+
+namespace android {
+namespace V4_0 {
+
+class DeviceHalHidl : public DeviceHalInterface, public ConversionHelperHidl
+{
+  public:
+    // Sets the value of 'devices' to a bitmask of 1 or more values of audio_devices_t.
+    virtual status_t getSupportedDevices(uint32_t *devices);
+
+    // Check to see if the audio hardware interface has been initialized.
+    virtual status_t initCheck();
+
+    // Set the audio volume of a voice call. Range is between 0.0 and 1.0.
+    virtual status_t setVoiceVolume(float volume);
+
+    // Set the audio volume for all audio activities other than voice call.
+    virtual status_t setMasterVolume(float volume);
+
+    // Get the current master volume value for the HAL.
+    virtual status_t getMasterVolume(float *volume);
+
+    // Called when the audio mode changes.
+    virtual status_t setMode(audio_mode_t mode);
+
+    // Muting control.
+    virtual status_t setMicMute(bool state);
+    virtual status_t getMicMute(bool *state);
+    virtual status_t setMasterMute(bool state);
+    virtual status_t getMasterMute(bool *state);
+
+    // Set global audio parameters.
+    virtual status_t setParameters(const String8& kvPairs);
+
+    // Get global audio parameters.
+    virtual status_t getParameters(const String8& keys, String8 *values);
+
+    // Returns audio input buffer size according to parameters passed.
+    virtual status_t getInputBufferSize(const struct audio_config *config,
+            size_t *size);
+
+    // Creates and opens the audio hardware output stream. The stream is closed
+    // by releasing all references to the returned object.
+    virtual status_t openOutputStream(
+            audio_io_handle_t handle,
+            audio_devices_t devices,
+            audio_output_flags_t flags,
+            struct audio_config *config,
+            const char *address,
+            sp<StreamOutHalInterface> *outStream);
+
+    // Creates and opens the audio hardware input stream. The stream is closed
+    // by releasing all references to the returned object.
+    virtual status_t openInputStream(
+            audio_io_handle_t handle,
+            audio_devices_t devices,
+            struct audio_config *config,
+            audio_input_flags_t flags,
+            const char *address,
+            audio_source_t source,
+            sp<StreamInHalInterface> *inStream);
+
+    // Returns whether createAudioPatch and releaseAudioPatch operations are supported.
+    virtual status_t supportsAudioPatches(bool *supportsPatches);
+
+    // Creates an audio patch between several source and sink ports.
+    virtual status_t createAudioPatch(
+            unsigned int num_sources,
+            const struct audio_port_config *sources,
+            unsigned int num_sinks,
+            const struct audio_port_config *sinks,
+            audio_patch_handle_t *patch);
+
+    // Releases an audio patch.
+    virtual status_t releaseAudioPatch(audio_patch_handle_t patch);
+
+    // Fills the list of supported attributes for a given audio port.
+    virtual status_t getAudioPort(struct audio_port *port);
+
+    // Set audio port configuration.
+    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
+
+    // List microphones
+    virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+
+    virtual status_t dump(int fd);
+
+  private:
+    friend class DevicesFactoryHalHidl;
+    sp<IDevice> mDevice;
+    sp<IPrimaryDevice> mPrimaryDevice;  // Null if it's not a primary device.
+
+    // Can not be constructed directly by clients.
+    explicit DeviceHalHidl(const sp<IDevice>& device);
+
+    // The destructor automatically closes the device.
+    virtual ~DeviceHalHidl();
+};
+
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_DEVICE_HAL_HIDL_4_0_H

diff --git a/media/libaudiohal/4.0/DeviceHalLocal.cpp b/media/libaudiohal/4.0/DeviceHalLocal.cpp
new file mode 100644
index 0000000..a245dd9
--- /dev/null
+++ b/media/libaudiohal/4.0/DeviceHalLocal.cpp

@@ -0,0 +1,213 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "DeviceHalLocal"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+
+#include "DeviceHalLocal.h"
+#include "StreamHalLocal.h"
+
+namespace android {
+namespace V4_0 {
+
+DeviceHalLocal::DeviceHalLocal(audio_hw_device_t *dev)
+        : mDev(dev) {
+}
+
+DeviceHalLocal::~DeviceHalLocal() {
+    int status = audio_hw_device_close(mDev);
+    ALOGW_IF(status, "Error closing audio hw device %p: %s", mDev, strerror(-status));
+    mDev = 0;
+}
+
+status_t DeviceHalLocal::getSupportedDevices(uint32_t *devices) {
+    if (mDev->get_supported_devices == NULL) return INVALID_OPERATION;
+    *devices = mDev->get_supported_devices(mDev);
+    return OK;
+}
+
+status_t DeviceHalLocal::initCheck() {
+    return mDev->init_check(mDev);
+}
+
+status_t DeviceHalLocal::setVoiceVolume(float volume) {
+    return mDev->set_voice_volume(mDev, volume);
+}
+
+status_t DeviceHalLocal::setMasterVolume(float volume) {
+    if (mDev->set_master_volume == NULL) return INVALID_OPERATION;
+    return mDev->set_master_volume(mDev, volume);
+}
+
+status_t DeviceHalLocal::getMasterVolume(float *volume) {
+    if (mDev->get_master_volume == NULL) return INVALID_OPERATION;
+    return mDev->get_master_volume(mDev, volume);
+}
+
+status_t DeviceHalLocal::setMode(audio_mode_t mode) {
+    return mDev->set_mode(mDev, mode);
+}
+
+status_t DeviceHalLocal::setMicMute(bool state) {
+    return mDev->set_mic_mute(mDev, state);
+}
+
+status_t DeviceHalLocal::getMicMute(bool *state) {
+    return mDev->get_mic_mute(mDev, state);
+}
+
+status_t DeviceHalLocal::setMasterMute(bool state) {
+    if (mDev->set_master_mute == NULL) return INVALID_OPERATION;
+    return mDev->set_master_mute(mDev, state);
+}
+
+status_t DeviceHalLocal::getMasterMute(bool *state) {
+    if (mDev->get_master_mute == NULL) return INVALID_OPERATION;
+    return mDev->get_master_mute(mDev, state);
+}
+
+status_t DeviceHalLocal::setParameters(const String8& kvPairs) {
+    return mDev->set_parameters(mDev, kvPairs.string());
+}
+
+status_t DeviceHalLocal::getParameters(const String8& keys, String8 *values) {
+    char *halValues = mDev->get_parameters(mDev, keys.string());
+    if (halValues != NULL) {
+        values->setTo(halValues);
+        free(halValues);
+    } else {
+        values->clear();
+    }
+    return OK;
+}
+
+status_t DeviceHalLocal::getInputBufferSize(
+        const struct audio_config *config, size_t *size) {
+    *size = mDev->get_input_buffer_size(mDev, config);
+    return OK;
+}
+
+status_t DeviceHalLocal::openOutputStream(
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        audio_output_flags_t flags,
+        struct audio_config *config,
+        const char *address,
+        sp<StreamOutHalInterface> *outStream) {
+    audio_stream_out_t *halStream;
+    ALOGV("open_output_stream handle: %d devices: %x flags: %#x"
+            "srate: %d format %#x channels %x address %s",
+            handle, devices, flags,
+            config->sample_rate, config->format, config->channel_mask,
+            address);
+    int openResut = mDev->open_output_stream(
+            mDev, handle, devices, flags, config, &halStream, address);
+    if (openResut == OK) {
+        *outStream = new StreamOutHalLocal(halStream, this);
+    }
+    ALOGV("open_output_stream status %d stream %p", openResut, halStream);
+    return openResut;
+}
+
+status_t DeviceHalLocal::openInputStream(
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        struct audio_config *config,
+        audio_input_flags_t flags,
+        const char *address,
+        audio_source_t source,
+        sp<StreamInHalInterface> *inStream) {
+    audio_stream_in_t *halStream;
+    ALOGV("open_input_stream handle: %d devices: %x flags: %#x "
+            "srate: %d format %#x channels %x address %s source %d",
+            handle, devices, flags,
+            config->sample_rate, config->format, config->channel_mask,
+            address, source);
+    int openResult = mDev->open_input_stream(
+            mDev, handle, devices, config, &halStream, flags, address, source);
+    if (openResult == OK) {
+        *inStream = new StreamInHalLocal(halStream, this);
+    }
+    ALOGV("open_input_stream status %d stream %p", openResult, inStream);
+    return openResult;
+}
+
+status_t DeviceHalLocal::supportsAudioPatches(bool *supportsPatches) {
+    *supportsPatches = version() >= AUDIO_DEVICE_API_VERSION_3_0;
+    return OK;
+}
+
+status_t DeviceHalLocal::createAudioPatch(
+        unsigned int num_sources,
+        const struct audio_port_config *sources,
+        unsigned int num_sinks,
+        const struct audio_port_config *sinks,
+        audio_patch_handle_t *patch) {
+    if (version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+        return mDev->create_audio_patch(
+                mDev, num_sources, sources, num_sinks, sinks, patch);
+    } else {
+        return INVALID_OPERATION;
+    }
+}
+
+status_t DeviceHalLocal::releaseAudioPatch(audio_patch_handle_t patch) {
+    if (version() >= AUDIO_DEVICE_API_VERSION_3_0) {
+        return mDev->release_audio_patch(mDev, patch);
+    } else {
+        return INVALID_OPERATION;
+    }
+}
+
+status_t DeviceHalLocal::getAudioPort(struct audio_port *port) {
+    return mDev->get_audio_port(mDev, port);
+}
+
+status_t DeviceHalLocal::setAudioPortConfig(const struct audio_port_config *config) {
+    if (version() >= AUDIO_DEVICE_API_VERSION_3_0)
+        return mDev->set_audio_port_config(mDev, config);
+    else
+        return INVALID_OPERATION;
+}
+
+status_t DeviceHalLocal::getMicrophones(std::vector<media::MicrophoneInfo> *microphones) {
+    if (mDev->get_microphones == NULL) return INVALID_OPERATION;
+    size_t actual_mics = AUDIO_MICROPHONE_MAX_COUNT;
+    audio_microphone_characteristic_t mic_array[AUDIO_MICROPHONE_MAX_COUNT];
+    status_t status = mDev->get_microphones(mDev, &mic_array[0], &actual_mics);
+    for (size_t i = 0; i < actual_mics; i++) {
+        media::MicrophoneInfo microphoneInfo = media::MicrophoneInfo(mic_array[i]);
+        microphones->push_back(microphoneInfo);
+    }
+    return status;
+}
+
+status_t DeviceHalLocal::dump(int fd) {
+    return mDev->dump(mDev, fd);
+}
+
+void DeviceHalLocal::closeOutputStream(struct audio_stream_out *stream_out) {
+    mDev->close_output_stream(mDev, stream_out);
+}
+
+void DeviceHalLocal::closeInputStream(struct audio_stream_in *stream_in) {
+    mDev->close_input_stream(mDev, stream_in);
+}
+
+} // namespace V4_0
+} // namespace android

diff --git a/media/libaudiohal/4.0/DeviceHalLocal.h b/media/libaudiohal/4.0/DeviceHalLocal.h
new file mode 100644
index 0000000..08341a4
--- /dev/null
+++ b/media/libaudiohal/4.0/DeviceHalLocal.h

@@ -0,0 +1,129 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_DEVICE_HAL_LOCAL_4_0_H
+#define ANDROID_HARDWARE_DEVICE_HAL_LOCAL_4_0_H
+
+#include <hardware/audio.h>
+#include <media/audiohal/DeviceHalInterface.h>
+
+namespace android {
+namespace V4_0 {
+
+class DeviceHalLocal : public DeviceHalInterface
+{
+  public:
+    // Sets the value of 'devices' to a bitmask of 1 or more values of audio_devices_t.
+    virtual status_t getSupportedDevices(uint32_t *devices);
+
+    // Check to see if the audio hardware interface has been initialized.
+    virtual status_t initCheck();
+
+    // Set the audio volume of a voice call. Range is between 0.0 and 1.0.
+    virtual status_t setVoiceVolume(float volume);
+
+    // Set the audio volume for all audio activities other than voice call.
+    virtual status_t setMasterVolume(float volume);
+
+    // Get the current master volume value for the HAL.
+    virtual status_t getMasterVolume(float *volume);
+
+    // Called when the audio mode changes.
+    virtual status_t setMode(audio_mode_t mode);
+
+    // Muting control.
+    virtual status_t setMicMute(bool state);
+    virtual status_t getMicMute(bool *state);
+    virtual status_t setMasterMute(bool state);
+    virtual status_t getMasterMute(bool *state);
+
+    // Set global audio parameters.
+    virtual status_t setParameters(const String8& kvPairs);
+
+    // Get global audio parameters.
+    virtual status_t getParameters(const String8& keys, String8 *values);
+
+    // Returns audio input buffer size according to parameters passed.
+    virtual status_t getInputBufferSize(const struct audio_config *config,
+            size_t *size);
+
+    // Creates and opens the audio hardware output stream. The stream is closed
+    // by releasing all references to the returned object.
+    virtual status_t openOutputStream(
+            audio_io_handle_t handle,
+            audio_devices_t devices,
+            audio_output_flags_t flags,
+            struct audio_config *config,
+            const char *address,
+            sp<StreamOutHalInterface> *outStream);
+
+    // Creates and opens the audio hardware input stream. The stream is closed
+    // by releasing all references to the returned object.
+    virtual status_t openInputStream(
+            audio_io_handle_t handle,
+            audio_devices_t devices,
+            struct audio_config *config,
+            audio_input_flags_t flags,
+            const char *address,
+            audio_source_t source,
+            sp<StreamInHalInterface> *inStream);
+
+    // Returns whether createAudioPatch and releaseAudioPatch operations are supported.
+    virtual status_t supportsAudioPatches(bool *supportsPatches);
+
+    // Creates an audio patch between several source and sink ports.
+    virtual status_t createAudioPatch(
+            unsigned int num_sources,
+            const struct audio_port_config *sources,
+            unsigned int num_sinks,
+            const struct audio_port_config *sinks,
+            audio_patch_handle_t *patch);
+
+    // Releases an audio patch.
+    virtual status_t releaseAudioPatch(audio_patch_handle_t patch);
+
+    // Fills the list of supported attributes for a given audio port.
+    virtual status_t getAudioPort(struct audio_port *port);
+
+    // Set audio port configuration.
+    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
+
+    // List microphones
+    virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+
+    virtual status_t dump(int fd);
+
+    void closeOutputStream(struct audio_stream_out *stream_out);
+    void closeInputStream(struct audio_stream_in *stream_in);
+
+  private:
+    audio_hw_device_t *mDev;
+
+    friend class DevicesFactoryHalLocal;
+
+    // Can not be constructed directly by clients.
+    explicit DeviceHalLocal(audio_hw_device_t *dev);
+
+    // The destructor automatically closes the device.
+    virtual ~DeviceHalLocal();
+
+    uint32_t version() const { return mDev->common.version; }
+};
+
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_DEVICE_HAL_LOCAL_4_0_H

diff --git a/media/libaudiohal/4.0/DevicesFactoryHalHidl.cpp b/media/libaudiohal/4.0/DevicesFactoryHalHidl.cpp
new file mode 100644
index 0000000..c83194e
--- /dev/null
+++ b/media/libaudiohal/4.0/DevicesFactoryHalHidl.cpp

@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <string.h>
+
+#define LOG_TAG "DevicesFactoryHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <android/hardware/audio/4.0/IDevice.h>
+#include <media/audiohal/hidl/HalDeathHandler.h>
+#include <utils/Log.h>
+
+#include "ConversionHelperHidl.h"
+#include "DeviceHalHidl.h"
+#include "DevicesFactoryHalHidl.h"
+
+using ::android::hardware::audio::V4_0::IDevice;
+using ::android::hardware::audio::V4_0::Result;
+using ::android::hardware::Return;
+
+namespace android {
+namespace V4_0 {
+
+DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
+    mDevicesFactory = IDevicesFactory::getService();
+    if (mDevicesFactory != 0) {
+        // It is assumed that DevicesFactory is owned by AudioFlinger
+        // and thus have the same lifespan.
+        mDevicesFactory->linkToDeath(HalDeathHandler::getInstance(), 0 /*cookie*/);
+    } else {
+        ALOGE("Failed to obtain IDevicesFactory service, terminating process.");
+        exit(1);
+    }
+    // The MSD factory is optional
+    mDevicesFactoryMsd = IDevicesFactory::getService(AUDIO_HAL_SERVICE_NAME_MSD);
+    // TODO: Register death handler, and add 'restart' directive to audioserver.rc
+}
+
+DevicesFactoryHalHidl::~DevicesFactoryHalHidl() {
+}
+
+status_t DevicesFactoryHalHidl::openDevice(const char *name, sp<DeviceHalInterface> *device) {
+    if (mDevicesFactory == 0) return NO_INIT;
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mDevicesFactory->openDevice(
+            name,
+            [&](Result r, const sp<IDevice>& result) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *device = new DeviceHalHidl(result);
+                }
+            });
+    if (ret.isOk()) {
+        if (retval == Result::OK) return OK;
+        else if (retval == Result::INVALID_ARGUMENTS) return BAD_VALUE;
+        else return NO_INIT;
+    }
+    return FAILED_TRANSACTION;
+}
+
+} // namespace V4_0
+} // namespace android

diff --git a/media/libaudiohal/4.0/DevicesFactoryHalHidl.h b/media/libaudiohal/4.0/DevicesFactoryHalHidl.h
new file mode 100644
index 0000000..114889b
--- /dev/null
+++ b/media/libaudiohal/4.0/DevicesFactoryHalHidl.h

@@ -0,0 +1,54 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_4_0_H
+#define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_4_0_H
+
+#include <android/hardware/audio/4.0/IDevicesFactory.h>
+#include <media/audiohal/DevicesFactoryHalInterface.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+#include "DeviceHalHidl.h"
+
+using ::android::hardware::audio::V4_0::IDevicesFactory;
+
+namespace android {
+namespace V4_0 {
+
+class DevicesFactoryHalHidl : public DevicesFactoryHalInterface
+{
+  public:
+    // Opens a device with the specified name. To close the device, it is
+    // necessary to release references to the returned object.
+    virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
+
+  private:
+    friend class DevicesFactoryHalHybrid;
+
+    sp<IDevicesFactory> mDevicesFactory;
+    sp<IDevicesFactory> mDevicesFactoryMsd;
+
+    // Can not be constructed directly by clients.
+    DevicesFactoryHalHidl();
+
+    virtual ~DevicesFactoryHalHidl();
+};
+
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_4_0_H

diff --git a/media/libaudiohal/4.0/DevicesFactoryHalHybrid.cpp b/media/libaudiohal/4.0/DevicesFactoryHalHybrid.cpp
new file mode 100644
index 0000000..7ff1ec7d
--- /dev/null
+++ b/media/libaudiohal/4.0/DevicesFactoryHalHybrid.cpp

@@ -0,0 +1,44 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "DevicesFactoryHalHybrid"
+//#define LOG_NDEBUG 0
+
+#include <libaudiohal/4.0/DevicesFactoryHalHybrid.h>
+#include "DevicesFactoryHalLocal.h"
+#include "DevicesFactoryHalHidl.h"
+
+namespace android {
+namespace V4_0 {
+
+DevicesFactoryHalHybrid::DevicesFactoryHalHybrid()
+        : mLocalFactory(new DevicesFactoryHalLocal()),
+          mHidlFactory(new DevicesFactoryHalHidl()) {
+}
+
+DevicesFactoryHalHybrid::~DevicesFactoryHalHybrid() {
+}
+
+status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
+    if (mHidlFactory != 0 && strcmp(AUDIO_HARDWARE_MODULE_ID_A2DP, name) != 0 &&
+        strcmp(AUDIO_HARDWARE_MODULE_ID_HEARING_AID, name) != 0) {
+        return mHidlFactory->openDevice(name, device);
+    }
+    return mLocalFactory->openDevice(name, device);
+}
+
+} // namespace V4_0
+} // namespace android

diff --git a/media/libaudiohal/4.0/DevicesFactoryHalLocal.cpp b/media/libaudiohal/4.0/DevicesFactoryHalLocal.cpp
new file mode 100644
index 0000000..e54edd4
--- /dev/null
+++ b/media/libaudiohal/4.0/DevicesFactoryHalLocal.cpp

@@ -0,0 +1,71 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "DevicesFactoryHalLocal"
+//#define LOG_NDEBUG 0
+
+#include <string.h>
+
+#include <hardware/audio.h>
+#include <utils/Log.h>
+
+#include "DeviceHalLocal.h"
+#include "DevicesFactoryHalLocal.h"
+
+namespace android {
+namespace V4_0 {
+
+static status_t load_audio_interface(const char *if_name, audio_hw_device_t **dev)
+{
+    const hw_module_t *mod;
+    int rc;
+
+    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
+    if (rc) {
+        ALOGE("%s couldn't load audio hw module %s.%s (%s)", __func__,
+                AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
+        goto out;
+    }
+    rc = audio_hw_device_open(mod, dev);
+    if (rc) {
+        ALOGE("%s couldn't open audio hw device in %s.%s (%s)", __func__,
+                AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
+        goto out;
+    }
+    if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
+        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
+        rc = BAD_VALUE;
+        audio_hw_device_close(*dev);
+        goto out;
+    }
+    return OK;
+
+out:
+    *dev = NULL;
+    return rc;
+}
+
+status_t DevicesFactoryHalLocal::openDevice(const char *name, sp<DeviceHalInterface> *device) {
+    audio_hw_device_t *dev;
+    status_t rc = load_audio_interface(name, &dev);
+    if (rc == OK) {
+        *device = new DeviceHalLocal(dev);
+    }
+    return rc;
+}
+
+} // namespace V4_0
+} // namespace android

diff --git a/media/libaudiohal/4.0/DevicesFactoryHalLocal.h b/media/libaudiohal/4.0/DevicesFactoryHalLocal.h
new file mode 100644
index 0000000..bc1c521
--- /dev/null
+++ b/media/libaudiohal/4.0/DevicesFactoryHalLocal.h

@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_DEVICES_FACTORY_HAL_LOCAL_4_0_H
+#define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_LOCAL_4_0_H
+
+#include <media/audiohal/DevicesFactoryHalInterface.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+#include "DeviceHalLocal.h"
+
+namespace android {
+namespace V4_0 {
+
+class DevicesFactoryHalLocal : public DevicesFactoryHalInterface
+{
+  public:
+    // Opens a device with the specified name. To close the device, it is
+    // necessary to release references to the returned object.
+    virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
+
+  private:
+    friend class DevicesFactoryHalHybrid;
+
+    // Can not be constructed directly by clients.
+    DevicesFactoryHalLocal() {}
+
+    virtual ~DevicesFactoryHalLocal() {}
+};
+
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_LOCAL_4_0_H

diff --git a/media/libaudiohal/4.0/EffectBufferHalHidl.cpp b/media/libaudiohal/4.0/EffectBufferHalHidl.cpp
new file mode 100644
index 0000000..957c89f
--- /dev/null
+++ b/media/libaudiohal/4.0/EffectBufferHalHidl.cpp

@@ -0,0 +1,146 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <atomic>
+
+#define LOG_TAG "EffectBufferHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <android/hidl/allocator/1.0/IAllocator.h>
+#include <hidlmemory/mapping.h>
+#include <utils/Log.h>
+
+#include "ConversionHelperHidl.h"
+#include "EffectBufferHalHidl.h"
+
+using ::android::hardware::Return;
+using ::android::hidl::allocator::V1_0::IAllocator;
+
+namespace android {
+namespace V4_0 {
+
+// static
+uint64_t EffectBufferHalHidl::makeUniqueId() {
+    static std::atomic<uint64_t> counter{1};
+    return counter++;
+}
+
+status_t EffectBufferHalHidl::allocate(
+        size_t size, sp<EffectBufferHalInterface>* buffer) {
+    return mirror(nullptr, size, buffer);
+}
+
+status_t EffectBufferHalHidl::mirror(
+        void* external, size_t size, sp<EffectBufferHalInterface>* buffer) {
+    sp<EffectBufferHalInterface> tempBuffer = new EffectBufferHalHidl(size);
+    status_t result = static_cast<EffectBufferHalHidl*>(tempBuffer.get())->init();
+    if (result == OK) {
+        tempBuffer->setExternalData(external);
+        *buffer = tempBuffer;
+    }
+    return result;
+}
+
+EffectBufferHalHidl::EffectBufferHalHidl(size_t size)
+        : mBufferSize(size), mFrameCountChanged(false),
+          mExternalData(nullptr), mAudioBuffer{0, {nullptr}} {
+    mHidlBuffer.id = makeUniqueId();
+    mHidlBuffer.frameCount = 0;
+}
+
+EffectBufferHalHidl::~EffectBufferHalHidl() {
+}
+
+status_t EffectBufferHalHidl::init() {
+    sp<IAllocator> ashmem = IAllocator::getService("ashmem");
+    if (ashmem == 0) {
+        ALOGE("Failed to retrieve ashmem allocator service");
+        return NO_INIT;
+    }
+    status_t retval = NO_MEMORY;
+    Return<void> result = ashmem->allocate(
+            mBufferSize,
+            [&](bool success, const hidl_memory& memory) {
+                if (success) {
+                    mHidlBuffer.data = memory;
+                    retval = OK;
+                }
+            });
+    if (result.isOk() && retval == OK) {
+        mMemory = hardware::mapMemory(mHidlBuffer.data);
+        if (mMemory != 0) {
+            mMemory->update();
+            mAudioBuffer.raw = static_cast<void*>(mMemory->getPointer());
+            memset(mAudioBuffer.raw, 0, mMemory->getSize());
+            mMemory->commit();
+        } else {
+            ALOGE("Failed to map allocated ashmem");
+            retval = NO_MEMORY;
+        }
+    } else {
+        ALOGE("Failed to allocate %d bytes from ashmem", (int)mBufferSize);
+    }
+    return result.isOk() ? retval : FAILED_TRANSACTION;
+}
+
+audio_buffer_t* EffectBufferHalHidl::audioBuffer() {
+    return &mAudioBuffer;
+}
+
+void* EffectBufferHalHidl::externalData() const {
+    return mExternalData;
+}
+
+void EffectBufferHalHidl::setFrameCount(size_t frameCount) {
+    mHidlBuffer.frameCount = frameCount;
+    mAudioBuffer.frameCount = frameCount;
+    mFrameCountChanged = true;
+}
+
+bool EffectBufferHalHidl::checkFrameCountChange() {
+    bool result = mFrameCountChanged;
+    mFrameCountChanged = false;
+    return result;
+}
+
+void EffectBufferHalHidl::setExternalData(void* external) {
+    mExternalData = external;
+}
+
+void EffectBufferHalHidl::update() {
+    update(mBufferSize);
+}
+
+void EffectBufferHalHidl::commit() {
+    commit(mBufferSize);
+}
+
+void EffectBufferHalHidl::update(size_t size) {
+    if (mExternalData == nullptr) return;
+    mMemory->update();
+    if (size > mBufferSize) size = mBufferSize;
+    memcpy(mAudioBuffer.raw, mExternalData, size);
+    mMemory->commit();
+}
+
+void EffectBufferHalHidl::commit(size_t size) {
+    if (mExternalData == nullptr) return;
+    if (size > mBufferSize) size = mBufferSize;
+    memcpy(mExternalData, mAudioBuffer.raw, size);
+}
+
+} // namespace V4_0
+} // namespace android

diff --git a/media/libaudiohal/4.0/EffectBufferHalHidl.h b/media/libaudiohal/4.0/EffectBufferHalHidl.h
new file mode 100644
index 0000000..6d578c6
--- /dev/null
+++ b/media/libaudiohal/4.0/EffectBufferHalHidl.h

@@ -0,0 +1,78 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_4_0_H
+#define ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_4_0_H
+
+#include <android/hardware/audio/effect/4.0/types.h>
+#include <android/hidl/memory/1.0/IMemory.h>
+#include <hidl/HidlSupport.h>
+#include <media/audiohal/EffectBufferHalInterface.h>
+#include <system/audio_effect.h>
+
+using android::hardware::audio::effect::V4_0::AudioBuffer;
+using android::hardware::hidl_memory;
+using android::hidl::memory::V1_0::IMemory;
+
+namespace android {
+namespace V4_0 {
+
+class EffectBufferHalHidl : public EffectBufferHalInterface
+{
+  public:
+    static status_t allocate(size_t size, sp<EffectBufferHalInterface>* buffer);
+    static status_t mirror(void* external, size_t size, sp<EffectBufferHalInterface>* buffer);
+
+    virtual audio_buffer_t* audioBuffer();
+    virtual void* externalData() const;
+
+    virtual size_t getSize() const override { return mBufferSize; }
+
+    virtual void setExternalData(void* external);
+    virtual void setFrameCount(size_t frameCount);
+    virtual bool checkFrameCountChange();
+
+    virtual void update();
+    virtual void commit();
+    virtual void update(size_t size);
+    virtual void commit(size_t size);
+
+    const AudioBuffer& hidlBuffer() const { return mHidlBuffer; }
+
+  private:
+    friend class EffectBufferHalInterface;
+
+    static uint64_t makeUniqueId();
+
+    const size_t mBufferSize;
+    bool mFrameCountChanged;
+    void* mExternalData;
+    AudioBuffer mHidlBuffer;
+    sp<IMemory> mMemory;
+    audio_buffer_t mAudioBuffer;
+
+    // Can not be constructed directly by clients.
+    explicit EffectBufferHalHidl(size_t size);
+
+    virtual ~EffectBufferHalHidl();
+
+    status_t init();
+};
+
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_4_0_H

diff --git a/media/libaudiohal/4.0/EffectHalHidl.cpp b/media/libaudiohal/4.0/EffectHalHidl.cpp
new file mode 100644
index 0000000..c99c4c8
--- /dev/null
+++ b/media/libaudiohal/4.0/EffectHalHidl.cpp

@@ -0,0 +1,342 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "EffectHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <common/all-versions/VersionUtils.h>
+#include <hwbinder/IPCThreadState.h>
+#include <media/EffectsFactoryApi.h>
+#include <utils/Log.h>
+
+#include "ConversionHelperHidl.h"
+#include "EffectBufferHalHidl.h"
+#include "EffectHalHidl.h"
+#include "HidlUtils.h"
+
+using ::android::hardware::audio::effect::V4_0::AudioBuffer;
+using ::android::hardware::audio::effect::V4_0::EffectBufferAccess;
+using ::android::hardware::audio::effect::V4_0::EffectConfigParameters;
+using ::android::hardware::audio::effect::V4_0::MessageQueueFlagBits;
+using ::android::hardware::audio::effect::V4_0::Result;
+using ::android::hardware::audio::common::V4_0::HidlUtils;
+using ::android::hardware::audio::common::V4_0::AudioChannelMask;
+using ::android::hardware::audio::common::V4_0::AudioFormat;
+using ::android::hardware::audio::common::utils::mkEnumConverter;
+using ::android::hardware::hidl_vec;
+using ::android::hardware::MQDescriptorSync;
+using ::android::hardware::Return;
+
+namespace android {
+namespace V4_0 {
+
+EffectHalHidl::EffectHalHidl(const sp<IEffect>& effect, uint64_t effectId)
+        : mEffect(effect), mEffectId(effectId), mBuffersChanged(true), mEfGroup(nullptr) {
+}
+
+EffectHalHidl::~EffectHalHidl() {
+    if (mEffect != 0) {
+        close();
+        mEffect.clear();
+        hardware::IPCThreadState::self()->flushCommands();
+    }
+    if (mEfGroup) {
+        EventFlag::deleteEventFlag(&mEfGroup);
+    }
+}
+
+// static
+void EffectHalHidl::effectDescriptorToHal(
+        const EffectDescriptor& descriptor, effect_descriptor_t* halDescriptor) {
+    HidlUtils::uuidToHal(descriptor.type, &halDescriptor->type);
+    HidlUtils::uuidToHal(descriptor.uuid, &halDescriptor->uuid);
+    halDescriptor->flags = static_cast<uint32_t>(descriptor.flags);
+    halDescriptor->cpuLoad = descriptor.cpuLoad;
+    halDescriptor->memoryUsage = descriptor.memoryUsage;
+    memcpy(halDescriptor->name, descriptor.name.data(), descriptor.name.size());
+    memcpy(halDescriptor->implementor,
+            descriptor.implementor.data(), descriptor.implementor.size());
+}
+
+// TODO(mnaganov): These buffer conversion functions should be shared with Effect wrapper
+// via HidlUtils. Move them there when hardware/interfaces will get un-frozen again.
+
+// static
+void EffectHalHidl::effectBufferConfigFromHal(
+        const buffer_config_t& halConfig, EffectBufferConfig* config) {
+    config->samplingRateHz = halConfig.samplingRate;
+    config->channels = mkEnumConverter<AudioChannelMask>(halConfig.channels);
+    config->format = AudioFormat(halConfig.format);
+    config->accessMode = EffectBufferAccess(halConfig.accessMode);
+    config->mask = mkEnumConverter<EffectConfigParameters>(halConfig.mask);
+}
+
+// static
+void EffectHalHidl::effectBufferConfigToHal(
+        const EffectBufferConfig& config, buffer_config_t* halConfig) {
+    halConfig->buffer.frameCount = 0;
+    halConfig->buffer.raw = NULL;
+    halConfig->samplingRate = config.samplingRateHz;
+    halConfig->channels = static_cast<uint32_t>(config.channels);
+    halConfig->bufferProvider.cookie = NULL;
+    halConfig->bufferProvider.getBuffer = NULL;
+    halConfig->bufferProvider.releaseBuffer = NULL;
+    halConfig->format = static_cast<uint8_t>(config.format);
+    halConfig->accessMode = static_cast<uint8_t>(config.accessMode);
+    halConfig->mask = static_cast<uint8_t>(config.mask);
+}
+
+// static
+void EffectHalHidl::effectConfigFromHal(const effect_config_t& halConfig, EffectConfig* config) {
+    effectBufferConfigFromHal(halConfig.inputCfg, &config->inputCfg);
+    effectBufferConfigFromHal(halConfig.outputCfg, &config->outputCfg);
+}
+
+// static
+void EffectHalHidl::effectConfigToHal(const EffectConfig& config, effect_config_t* halConfig) {
+    effectBufferConfigToHal(config.inputCfg, &halConfig->inputCfg);
+    effectBufferConfigToHal(config.outputCfg, &halConfig->outputCfg);
+}
+
+// static
+status_t EffectHalHidl::analyzeResult(const Result& result) {
+    switch (result) {
+        case Result::OK: return OK;
+        case Result::INVALID_ARGUMENTS: return BAD_VALUE;
+        case Result::INVALID_STATE: return NOT_ENOUGH_DATA;
+        case Result::NOT_INITIALIZED: return NO_INIT;
+        case Result::NOT_SUPPORTED: return INVALID_OPERATION;
+        case Result::RESULT_TOO_BIG: return NO_MEMORY;
+        default: return NO_INIT;
+    }
+}
+
+status_t EffectHalHidl::setInBuffer(const sp<EffectBufferHalInterface>& buffer) {
+    if (!mBuffersChanged) {
+        if (buffer.get() == nullptr || mInBuffer.get() == nullptr) {
+            mBuffersChanged = buffer.get() != mInBuffer.get();
+        } else {
+            mBuffersChanged = buffer->audioBuffer() != mInBuffer->audioBuffer();
+        }
+    }
+    mInBuffer = buffer;
+    return OK;
+}
+
+status_t EffectHalHidl::setOutBuffer(const sp<EffectBufferHalInterface>& buffer) {
+    if (!mBuffersChanged) {
+        if (buffer.get() == nullptr || mOutBuffer.get() == nullptr) {
+            mBuffersChanged = buffer.get() != mOutBuffer.get();
+        } else {
+            mBuffersChanged = buffer->audioBuffer() != mOutBuffer->audioBuffer();
+        }
+    }
+    mOutBuffer = buffer;
+    return OK;
+}
+
+status_t EffectHalHidl::process() {
+    return processImpl(static_cast<uint32_t>(MessageQueueFlagBits::REQUEST_PROCESS));
+}
+
+status_t EffectHalHidl::processReverse() {
+    return processImpl(static_cast<uint32_t>(MessageQueueFlagBits::REQUEST_PROCESS_REVERSE));
+}
+
+status_t EffectHalHidl::prepareForProcessing() {
+    std::unique_ptr<StatusMQ> tempStatusMQ;
+    Result retval;
+    Return<void> ret = mEffect->prepareForProcessing(
+            [&](Result r, const MQDescriptorSync<Result>& statusMQ) {
+                retval = r;
+                if (retval == Result::OK) {
+                    tempStatusMQ.reset(new StatusMQ(statusMQ));
+                    if (tempStatusMQ->isValid() && tempStatusMQ->getEventFlagWord()) {
+                        EventFlag::createEventFlag(tempStatusMQ->getEventFlagWord(), &mEfGroup);
+                    }
+                }
+            });
+    if (!ret.isOk() || retval != Result::OK) {
+        return ret.isOk() ? analyzeResult(retval) : FAILED_TRANSACTION;
+    }
+    if (!tempStatusMQ || !tempStatusMQ->isValid() || !mEfGroup) {
+        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for effects");
+        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
+                "Status message queue for effects is invalid");
+        ALOGE_IF(!mEfGroup, "Event flag creation for effects failed");
+        return NO_INIT;
+    }
+    mStatusMQ = std::move(tempStatusMQ);
+    return OK;
+}
+
+bool EffectHalHidl::needToResetBuffers() {
+    if (mBuffersChanged) return true;
+    bool inBufferFrameCountUpdated = mInBuffer->checkFrameCountChange();
+    bool outBufferFrameCountUpdated = mOutBuffer->checkFrameCountChange();
+    return inBufferFrameCountUpdated || outBufferFrameCountUpdated;
+}
+
+status_t EffectHalHidl::processImpl(uint32_t mqFlag) {
+    if (mEffect == 0 || mInBuffer == 0 || mOutBuffer == 0) return NO_INIT;
+    status_t status;
+    if (!mStatusMQ && (status = prepareForProcessing()) != OK) {
+        return status;
+    }
+    if (needToResetBuffers() && (status = setProcessBuffers()) != OK) {
+        return status;
+    }
+    // The data is already in the buffers, just need to flush it and wake up the server side.
+    std::atomic_thread_fence(std::memory_order_release);
+    mEfGroup->wake(mqFlag);
+    uint32_t efState = 0;
+retry:
+    status_t ret = mEfGroup->wait(
+            static_cast<uint32_t>(MessageQueueFlagBits::DONE_PROCESSING), &efState);
+    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::DONE_PROCESSING)) {
+        Result retval = Result::NOT_INITIALIZED;
+        mStatusMQ->read(&retval);
+        if (retval == Result::OK || retval == Result::INVALID_STATE) {
+            // Sync back the changed contents of the buffer.
+            std::atomic_thread_fence(std::memory_order_acquire);
+        }
+        return analyzeResult(retval);
+    }
+    if (ret == -EAGAIN || ret == -EINTR) {
+        // Spurious wakeup. This normally retries no more than once.
+        goto retry;
+    }
+    return ret;
+}
+
+status_t EffectHalHidl::setProcessBuffers() {
+    Return<Result> ret = mEffect->setProcessBuffers(
+            static_cast<EffectBufferHalHidl*>(mInBuffer.get())->hidlBuffer(),
+            static_cast<EffectBufferHalHidl*>(mOutBuffer.get())->hidlBuffer());
+    if (ret.isOk() && ret == Result::OK) {
+        mBuffersChanged = false;
+        return OK;
+    }
+    return ret.isOk() ? analyzeResult(ret) : FAILED_TRANSACTION;
+}
+
+status_t EffectHalHidl::command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData,
+        uint32_t *replySize, void *pReplyData) {
+    if (mEffect == 0) return NO_INIT;
+
+    // Special cases.
+    if (cmdCode == EFFECT_CMD_SET_CONFIG || cmdCode == EFFECT_CMD_SET_CONFIG_REVERSE) {
+        return setConfigImpl(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
+    } else if (cmdCode == EFFECT_CMD_GET_CONFIG || cmdCode == EFFECT_CMD_GET_CONFIG_REVERSE) {
+        return getConfigImpl(cmdCode, replySize, pReplyData);
+    }
+
+    // Common case.
+    hidl_vec<uint8_t> hidlData;
+    if (pCmdData != nullptr && cmdSize > 0) {
+        hidlData.setToExternal(reinterpret_cast<uint8_t*>(pCmdData), cmdSize);
+    }
+    status_t status;
+    uint32_t replySizeStub = 0;
+    if (replySize == nullptr || pReplyData == nullptr) replySize = &replySizeStub;
+    Return<void> ret = mEffect->command(cmdCode, hidlData, *replySize,
+            [&](int32_t s, const hidl_vec<uint8_t>& result) {
+                status = s;
+                if (status == 0) {
+                    if (*replySize > result.size()) *replySize = result.size();
+                    if (pReplyData != nullptr && *replySize > 0) {
+                        memcpy(pReplyData, &result[0], *replySize);
+                    }
+                }
+            });
+    return ret.isOk() ? status : FAILED_TRANSACTION;
+}
+
+status_t EffectHalHidl::getDescriptor(effect_descriptor_t *pDescriptor) {
+    if (mEffect == 0) return NO_INIT;
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mEffect->getDescriptor(
+            [&](Result r, const EffectDescriptor& result) {
+                retval = r;
+                if (retval == Result::OK) {
+                    effectDescriptorToHal(result, pDescriptor);
+                }
+            });
+    return ret.isOk() ? analyzeResult(retval) : FAILED_TRANSACTION;
+}
+
+status_t EffectHalHidl::close() {
+    if (mEffect == 0) return NO_INIT;
+    Return<Result> ret = mEffect->close();
+    return ret.isOk() ? analyzeResult(ret) : FAILED_TRANSACTION;
+}
+
+status_t EffectHalHidl::getConfigImpl(
+        uint32_t cmdCode, uint32_t *replySize, void *pReplyData) {
+    if (replySize == NULL || *replySize != sizeof(effect_config_t) || pReplyData == NULL) {
+        return BAD_VALUE;
+    }
+    status_t result = FAILED_TRANSACTION;
+    Return<void> ret;
+    if (cmdCode == EFFECT_CMD_GET_CONFIG) {
+        ret = mEffect->getConfig([&] (Result r, const EffectConfig &hidlConfig) {
+            result = analyzeResult(r);
+            if (r == Result::OK) {
+                effectConfigToHal(hidlConfig, static_cast<effect_config_t*>(pReplyData));
+            }
+        });
+    } else {
+        ret = mEffect->getConfigReverse([&] (Result r, const EffectConfig &hidlConfig) {
+            result = analyzeResult(r);
+            if (r == Result::OK) {
+                effectConfigToHal(hidlConfig, static_cast<effect_config_t*>(pReplyData));
+            }
+        });
+    }
+    if (!ret.isOk()) {
+        result = FAILED_TRANSACTION;
+    }
+    return result;
+}
+
+status_t EffectHalHidl::setConfigImpl(
+        uint32_t cmdCode, uint32_t cmdSize, void *pCmdData, uint32_t *replySize, void *pReplyData) {
+    if (pCmdData == NULL || cmdSize != sizeof(effect_config_t) ||
+            replySize == NULL || *replySize != sizeof(int32_t) || pReplyData == NULL) {
+        return BAD_VALUE;
+    }
+    const effect_config_t *halConfig = static_cast<effect_config_t*>(pCmdData);
+    if (halConfig->inputCfg.bufferProvider.getBuffer != NULL ||
+            halConfig->inputCfg.bufferProvider.releaseBuffer != NULL ||
+            halConfig->outputCfg.bufferProvider.getBuffer != NULL ||
+            halConfig->outputCfg.bufferProvider.releaseBuffer != NULL) {
+        ALOGE("Buffer provider callbacks are not supported");
+    }
+    EffectConfig hidlConfig;
+    effectConfigFromHal(*halConfig, &hidlConfig);
+    Return<Result> ret = cmdCode == EFFECT_CMD_SET_CONFIG ?
+            mEffect->setConfig(hidlConfig, nullptr, nullptr) :
+            mEffect->setConfigReverse(hidlConfig, nullptr, nullptr);
+    status_t result = FAILED_TRANSACTION;
+    if (ret.isOk()) {
+        result = analyzeResult(ret);
+        *static_cast<int32_t*>(pReplyData) = result;
+    }
+    return result;
+}
+
+} // namespace V4_0
+} // namespace android

diff --git a/media/libaudiohal/4.0/EffectHalHidl.h b/media/libaudiohal/4.0/EffectHalHidl.h
new file mode 100644
index 0000000..5a4dab1
--- /dev/null
+++ b/media/libaudiohal/4.0/EffectHalHidl.h

@@ -0,0 +1,110 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_EFFECT_HAL_HIDL_4_0_H
+#define ANDROID_HARDWARE_EFFECT_HAL_HIDL_4_0_H
+
+#include <android/hardware/audio/effect/4.0/IEffect.h>
+#include <media/audiohal/EffectHalInterface.h>
+#include <fmq/EventFlag.h>
+#include <fmq/MessageQueue.h>
+#include <system/audio_effect.h>
+
+using ::android::hardware::audio::effect::V4_0::EffectBufferConfig;
+using ::android::hardware::audio::effect::V4_0::EffectConfig;
+using ::android::hardware::audio::effect::V4_0::EffectDescriptor;
+using ::android::hardware::audio::effect::V4_0::IEffect;
+using ::android::hardware::EventFlag;
+using ::android::hardware::MessageQueue;
+
+namespace android {
+namespace V4_0 {
+
+class EffectHalHidl : public EffectHalInterface
+{
+  public:
+    // Set the input buffer.
+    virtual status_t setInBuffer(const sp<EffectBufferHalInterface>& buffer);
+
+    // Set the output buffer.
+    virtual status_t setOutBuffer(const sp<EffectBufferHalInterface>& buffer);
+
+    // Effect process function.
+    virtual status_t process();
+
+    // Process reverse stream function. This function is used to pass
+    // a reference stream to the effect engine.
+    virtual status_t processReverse();
+
+    // Send a command and receive a response to/from effect engine.
+    virtual status_t command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData,
+            uint32_t *replySize, void *pReplyData);
+
+    // Returns the effect descriptor.
+    virtual status_t getDescriptor(effect_descriptor_t *pDescriptor);
+
+    // Free resources on the remote side.
+    virtual status_t close();
+
+    // Whether it's a local implementation.
+    virtual bool isLocal() const { return false; }
+
+    uint64_t effectId() const { return mEffectId; }
+
+    static void effectDescriptorToHal(
+            const EffectDescriptor& descriptor, effect_descriptor_t* halDescriptor);
+
+  private:
+    friend class EffectsFactoryHalHidl;
+    typedef MessageQueue<
+        hardware::audio::effect::V4_0::Result, hardware::kSynchronizedReadWrite> StatusMQ;
+
+    sp<IEffect> mEffect;
+    const uint64_t mEffectId;
+    sp<EffectBufferHalInterface> mInBuffer;
+    sp<EffectBufferHalInterface> mOutBuffer;
+    bool mBuffersChanged;
+    std::unique_ptr<StatusMQ> mStatusMQ;
+    EventFlag* mEfGroup;
+
+    static status_t analyzeResult(const hardware::audio::effect::V4_0::Result& result);
+    static void effectBufferConfigFromHal(
+            const buffer_config_t& halConfig, EffectBufferConfig* config);
+    static void effectBufferConfigToHal(
+            const EffectBufferConfig& config, buffer_config_t* halConfig);
+    static void effectConfigFromHal(const effect_config_t& halConfig, EffectConfig* config);
+    static void effectConfigToHal(const EffectConfig& config, effect_config_t* halConfig);
+
+    // Can not be constructed directly by clients.
+    EffectHalHidl(const sp<IEffect>& effect, uint64_t effectId);
+
+    // The destructor automatically releases the effect.
+    virtual ~EffectHalHidl();
+
+    status_t getConfigImpl(uint32_t cmdCode, uint32_t *replySize, void *pReplyData);
+    status_t prepareForProcessing();
+    bool needToResetBuffers();
+    status_t processImpl(uint32_t mqFlag);
+    status_t setConfigImpl(
+            uint32_t cmdCode, uint32_t cmdSize, void *pCmdData,
+            uint32_t *replySize, void *pReplyData);
+    status_t setProcessBuffers();
+};
+
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_EFFECT_HAL_HIDL_4_0_H

diff --git a/media/libaudiohal/4.0/EffectsFactoryHalHidl.cpp b/media/libaudiohal/4.0/EffectsFactoryHalHidl.cpp
new file mode 100644
index 0000000..dfed784
--- /dev/null
+++ b/media/libaudiohal/4.0/EffectsFactoryHalHidl.cpp

@@ -0,0 +1,152 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "EffectsFactoryHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <cutils/native_handle.h>
+#include <libaudiohal/4.0/EffectsFactoryHalHidl.h>
+
+#include "ConversionHelperHidl.h"
+#include "EffectBufferHalHidl.h"
+#include "EffectHalHidl.h"
+#include "HidlUtils.h"
+
+using ::android::hardware::audio::common::V4_0::HidlUtils;
+using ::android::hardware::audio::common::V4_0::Uuid;
+using ::android::hardware::audio::effect::V4_0::IEffect;
+using ::android::hardware::audio::effect::V4_0::Result;
+using ::android::hardware::Return;
+
+namespace android {
+namespace V4_0 {
+
+EffectsFactoryHalHidl::EffectsFactoryHalHidl() : ConversionHelperHidl("EffectsFactory") {
+    mEffectsFactory = IEffectsFactory::getService();
+    if (mEffectsFactory == 0) {
+        ALOGE("Failed to obtain IEffectsFactory service, terminating process.");
+        exit(1);
+    }
+}
+
+EffectsFactoryHalHidl::~EffectsFactoryHalHidl() {
+}
+
+status_t EffectsFactoryHalHidl::queryAllDescriptors() {
+    if (mEffectsFactory == 0) return NO_INIT;
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mEffectsFactory->getAllDescriptors(
+            [&](Result r, const hidl_vec<EffectDescriptor>& result) {
+                retval = r;
+                if (retval == Result::OK) {
+                    mLastDescriptors = result;
+                }
+            });
+    if (ret.isOk()) {
+        return retval == Result::OK ? OK : NO_INIT;
+    }
+    mLastDescriptors.resize(0);
+    return processReturn(__FUNCTION__, ret);
+}
+
+status_t EffectsFactoryHalHidl::queryNumberEffects(uint32_t *pNumEffects) {
+    status_t queryResult = queryAllDescriptors();
+    if (queryResult == OK) {
+        *pNumEffects = mLastDescriptors.size();
+    }
+    return queryResult;
+}
+
+status_t EffectsFactoryHalHidl::getDescriptor(
+        uint32_t index, effect_descriptor_t *pDescriptor) {
+    // TODO: We need somehow to track the changes on the server side
+    // or figure out how to convert everybody to query all the descriptors at once.
+    // TODO: check for nullptr
+    if (mLastDescriptors.size() == 0) {
+        status_t queryResult = queryAllDescriptors();
+        if (queryResult != OK) return queryResult;
+    }
+    if (index >= mLastDescriptors.size()) return NAME_NOT_FOUND;
+    EffectHalHidl::effectDescriptorToHal(mLastDescriptors[index], pDescriptor);
+    return OK;
+}
+
+status_t EffectsFactoryHalHidl::getDescriptor(
+        const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor) {
+    // TODO: check for nullptr
+    if (mEffectsFactory == 0) return NO_INIT;
+    Uuid hidlUuid;
+    HidlUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mEffectsFactory->getDescriptor(hidlUuid,
+            [&](Result r, const EffectDescriptor& result) {
+                retval = r;
+                if (retval == Result::OK) {
+                    EffectHalHidl::effectDescriptorToHal(result, pDescriptor);
+                }
+            });
+    if (ret.isOk()) {
+        if (retval == Result::OK) return OK;
+        else if (retval == Result::INVALID_ARGUMENTS) return NAME_NOT_FOUND;
+        else return NO_INIT;
+    }
+    return processReturn(__FUNCTION__, ret);
+}
+
+status_t EffectsFactoryHalHidl::createEffect(
+        const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId,
+        sp<EffectHalInterface> *effect) {
+    if (mEffectsFactory == 0) return NO_INIT;
+    Uuid hidlUuid;
+    HidlUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
+    Result retval = Result::NOT_INITIALIZED;
+    Return<void> ret = mEffectsFactory->createEffect(
+            hidlUuid, sessionId, ioId,
+            [&](Result r, const sp<IEffect>& result, uint64_t effectId) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *effect = new EffectHalHidl(result, effectId);
+                }
+            });
+    if (ret.isOk()) {
+        if (retval == Result::OK) return OK;
+        else if (retval == Result::INVALID_ARGUMENTS) return NAME_NOT_FOUND;
+        else return NO_INIT;
+    }
+    return processReturn(__FUNCTION__, ret);
+}
+
+status_t EffectsFactoryHalHidl::dumpEffects(int fd) {
+    if (mEffectsFactory == 0) return NO_INIT;
+    native_handle_t* hidlHandle = native_handle_create(1, 0);
+    hidlHandle->data[0] = fd;
+    Return<void> ret = mEffectsFactory->debug(hidlHandle, {} /* options */);
+    native_handle_delete(hidlHandle);
+    return processReturn(__FUNCTION__, ret);
+}
+
+status_t EffectsFactoryHalHidl::allocateBuffer(size_t size, sp<EffectBufferHalInterface>* buffer) {
+    return EffectBufferHalHidl::allocate(size, buffer);
+}
+
+status_t EffectsFactoryHalHidl::mirrorBuffer(void* external, size_t size,
+                          sp<EffectBufferHalInterface>* buffer) {
+    return EffectBufferHalHidl::mirror(external, size, buffer);
+}
+
+
+} // namespace V4_0
+} // namespace android

diff --git a/media/libaudiohal/4.0/StreamHalHidl.cpp b/media/libaudiohal/4.0/StreamHalHidl.cpp
new file mode 100644
index 0000000..1c2fdb0
--- /dev/null
+++ b/media/libaudiohal/4.0/StreamHalHidl.cpp

@@ -0,0 +1,817 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "StreamHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <android/hardware/audio/4.0/IStreamOutCallback.h>
+#include <hwbinder/IPCThreadState.h>
+#include <mediautils/SchedulingPolicyService.h>
+#include <utils/Log.h>
+
+#include "DeviceHalHidl.h"
+#include "EffectHalHidl.h"
+#include "StreamHalHidl.h"
+#include "VersionUtils.h"
+
+using ::android::hardware::audio::common::V4_0::AudioChannelMask;
+using ::android::hardware::audio::common::V4_0::AudioContentType;
+using ::android::hardware::audio::common::V4_0::AudioFormat;
+using ::android::hardware::audio::common::V4_0::AudioSource;
+using ::android::hardware::audio::common::V4_0::AudioUsage;
+using ::android::hardware::audio::common::V4_0::ThreadInfo;
+using ::android::hardware::audio::V4_0::AudioDrain;
+using ::android::hardware::audio::V4_0::IStreamOutCallback;
+using ::android::hardware::audio::V4_0::MessageQueueFlagBits;
+using ::android::hardware::audio::V4_0::MicrophoneInfo;
+using ::android::hardware::audio::V4_0::MmapBufferInfo;
+using ::android::hardware::audio::V4_0::MmapPosition;
+using ::android::hardware::audio::V4_0::ParameterValue;
+using ::android::hardware::audio::V4_0::PlaybackTrackMetadata;
+using ::android::hardware::audio::V4_0::RecordTrackMetadata;
+using ::android::hardware::audio::V4_0::Result;
+using ::android::hardware::audio::V4_0::TimeSpec;
+using ::android::hardware::MQDescriptorSync;
+using ::android::hardware::Return;
+using ::android::hardware::Void;
+using ReadCommand = ::android::hardware::audio::V4_0::IStreamIn::ReadCommand;
+
+namespace android {
+namespace V4_0 {
+
+StreamHalHidl::StreamHalHidl(IStream *stream)
+        : ConversionHelperHidl("Stream"),
+          mStream(stream),
+          mHalThreadPriority(HAL_THREAD_PRIORITY_DEFAULT),
+          mCachedBufferSize(0){
+
+    // Instrument audio signal power logging.
+    // Note: This assumes channel mask, format, and sample rate do not change after creation.
+    if (mStream != nullptr && mStreamPowerLog.isUserDebugOrEngBuild()) {
+        // Obtain audio properties (see StreamHalHidl::getAudioProperties() below).
+        Return<void> ret = mStream->getAudioProperties(
+                [&](auto sr, auto m, auto f) {
+                mStreamPowerLog.init(sr,
+                        static_cast<audio_channel_mask_t>(m),
+                        static_cast<audio_format_t>(f));
+            });
+    }
+}
+
+StreamHalHidl::~StreamHalHidl() {
+    mStream = nullptr;
+}
+
+status_t StreamHalHidl::getSampleRate(uint32_t *rate) {
+    if (!mStream) return NO_INIT;
+    return processReturn("getSampleRate", mStream->getSampleRate(), rate);
+}
+
+status_t StreamHalHidl::getBufferSize(size_t *size) {
+    if (!mStream) return NO_INIT;
+    status_t status = processReturn("getBufferSize", mStream->getBufferSize(), size);
+    if (status == OK) {
+        mCachedBufferSize = *size;
+    }
+    return status;
+}
+
+status_t StreamHalHidl::getChannelMask(audio_channel_mask_t *mask) {
+    if (!mStream) return NO_INIT;
+    return processReturn("getChannelMask", mStream->getChannelMask(), mask);
+}
+
+status_t StreamHalHidl::getFormat(audio_format_t *format) {
+    if (!mStream) return NO_INIT;
+    return processReturn("getFormat", mStream->getFormat(), format);
+}
+
+status_t StreamHalHidl::getAudioProperties(
+        uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format) {
+    if (!mStream) return NO_INIT;
+    Return<void> ret = mStream->getAudioProperties(
+            [&](uint32_t sr, auto m, auto f) {
+                *sampleRate = sr;
+                *mask = static_cast<audio_channel_mask_t>(m);
+                *format = static_cast<audio_format_t>(f);
+            });
+    return processReturn("getAudioProperties", ret);
+}
+
+status_t StreamHalHidl::setParameters(const String8& kvPairs) {
+    if (!mStream) return NO_INIT;
+    hidl_vec<ParameterValue> hidlParams;
+    status_t status = parametersFromHal(kvPairs, &hidlParams);
+    if (status != OK) return status;
+    return processReturn("setParameters",
+                         utils::setParameters(mStream, hidlParams, {} /* options */));
+}
+
+status_t StreamHalHidl::getParameters(const String8& keys, String8 *values) {
+    values->clear();
+    if (!mStream) return NO_INIT;
+    hidl_vec<hidl_string> hidlKeys;
+    status_t status = keysFromHal(keys, &hidlKeys);
+    if (status != OK) return status;
+    Result retval;
+    Return<void> ret = utils::getParameters(
+            mStream,
+            {} /* context */,
+            hidlKeys,
+            [&](Result r, const hidl_vec<ParameterValue>& parameters) {
+                retval = r;
+                if (retval == Result::OK) {
+                    parametersToHal(parameters, values);
+                }
+            });
+    return processReturn("getParameters", ret, retval);
+}
+
+status_t StreamHalHidl::addEffect(sp<EffectHalInterface> effect) {
+    if (!mStream) return NO_INIT;
+    return processReturn("addEffect", mStream->addEffect(
+                    static_cast<EffectHalHidl*>(effect.get())->effectId()));
+}
+
+status_t StreamHalHidl::removeEffect(sp<EffectHalInterface> effect) {
+    if (!mStream) return NO_INIT;
+    return processReturn("removeEffect", mStream->removeEffect(
+                    static_cast<EffectHalHidl*>(effect.get())->effectId()));
+}
+
+status_t StreamHalHidl::standby() {
+    if (!mStream) return NO_INIT;
+    return processReturn("standby", mStream->standby());
+}
+
+status_t StreamHalHidl::dump(int fd) {
+    if (!mStream) return NO_INIT;
+    native_handle_t* hidlHandle = native_handle_create(1, 0);
+    hidlHandle->data[0] = fd;
+    Return<void> ret = mStream->debug(hidlHandle, {} /* options */);
+    native_handle_delete(hidlHandle);
+    mStreamPowerLog.dump(fd);
+    return processReturn("dump", ret);
+}
+
+status_t StreamHalHidl::start() {
+    if (!mStream) return NO_INIT;
+    return processReturn("start", mStream->start());
+}
+
+status_t StreamHalHidl::stop() {
+    if (!mStream) return NO_INIT;
+    return processReturn("stop", mStream->stop());
+}
+
+status_t StreamHalHidl::createMmapBuffer(int32_t minSizeFrames,
+                                  struct audio_mmap_buffer_info *info) {
+    Result retval;
+    Return<void> ret = mStream->createMmapBuffer(
+            minSizeFrames,
+            [&](Result r, const MmapBufferInfo& hidlInfo) {
+                retval = r;
+                if (retval == Result::OK) {
+                    const native_handle *handle = hidlInfo.sharedMemory.handle();
+                    if (handle->numFds > 0) {
+                        info->shared_memory_fd = handle->data[0];
+                        info->buffer_size_frames = hidlInfo.bufferSizeFrames;
+                        info->burst_size_frames = hidlInfo.burstSizeFrames;
+                        // info->shared_memory_address is not needed in HIDL context
+                        info->shared_memory_address = NULL;
+                    } else {
+                        retval = Result::NOT_INITIALIZED;
+                    }
+                }
+            });
+    return processReturn("createMmapBuffer", ret, retval);
+}
+
+status_t StreamHalHidl::getMmapPosition(struct audio_mmap_position *position) {
+    Result retval;
+    Return<void> ret = mStream->getMmapPosition(
+            [&](Result r, const MmapPosition& hidlPosition) {
+                retval = r;
+                if (retval == Result::OK) {
+                    position->time_nanoseconds = hidlPosition.timeNanoseconds;
+                    position->position_frames = hidlPosition.positionFrames;
+                }
+            });
+    return processReturn("getMmapPosition", ret, retval);
+}
+
+status_t StreamHalHidl::setHalThreadPriority(int priority) {
+    mHalThreadPriority = priority;
+    return OK;
+}
+
+status_t StreamHalHidl::getCachedBufferSize(size_t *size) {
+    if (mCachedBufferSize != 0) {
+        *size = mCachedBufferSize;
+        return OK;
+    }
+    return getBufferSize(size);
+}
+
+bool StreamHalHidl::requestHalThreadPriority(pid_t threadPid, pid_t threadId) {
+    if (mHalThreadPriority == HAL_THREAD_PRIORITY_DEFAULT) {
+        return true;
+    }
+    int err = requestPriority(
+            threadPid, threadId,
+            mHalThreadPriority, false /*isForApp*/, true /*asynchronous*/);
+    ALOGE_IF(err, "failed to set priority %d for pid %d tid %d; error %d",
+            mHalThreadPriority, threadPid, threadId, err);
+    // Audio will still work, but latency will be higher and sometimes unacceptable.
+    return err == 0;
+}
+
+namespace {
+
+/* Notes on callback ownership.
+
+This is how (Hw)Binder ownership model looks like. The server implementation
+is owned by Binder framework (via sp<>). Proxies are owned by clients.
+When the last proxy disappears, Binder framework releases the server impl.
+
+Thus, it is not needed to keep any references to StreamOutCallback (this is
+the server impl) -- it will live as long as HAL server holds a strong ref to
+IStreamOutCallback proxy. We clear that reference by calling 'clearCallback'
+from the destructor of StreamOutHalHidl.
+
+The callback only keeps a weak reference to the stream. The stream is owned
+by AudioFlinger.
+
+*/
+
+struct StreamOutCallback : public IStreamOutCallback {
+    StreamOutCallback(const wp<StreamOutHalHidl>& stream) : mStream(stream) {}
+
+    // IStreamOutCallback implementation
+    Return<void> onWriteReady()  override {
+        sp<StreamOutHalHidl> stream = mStream.promote();
+        if (stream != 0) {
+            stream->onWriteReady();
+        }
+        return Void();
+    }
+
+    Return<void> onDrainReady()  override {
+        sp<StreamOutHalHidl> stream = mStream.promote();
+        if (stream != 0) {
+            stream->onDrainReady();
+        }
+        return Void();
+    }
+
+    Return<void> onError()  override {
+        sp<StreamOutHalHidl> stream = mStream.promote();
+        if (stream != 0) {
+            stream->onError();
+        }
+        return Void();
+    }
+
+  private:
+    wp<StreamOutHalHidl> mStream;
+};
+
+}  // namespace
+
+StreamOutHalHidl::StreamOutHalHidl(const sp<IStreamOut>& stream)
+        : StreamHalHidl(stream.get()), mStream(stream), mWriterClient(0), mEfGroup(nullptr) {
+}
+
+StreamOutHalHidl::~StreamOutHalHidl() {
+    if (mStream != 0) {
+        if (mCallback.unsafe_get()) {
+            processReturn("clearCallback", mStream->clearCallback());
+        }
+        processReturn("close", mStream->close());
+        mStream.clear();
+    }
+    mCallback.clear();
+    hardware::IPCThreadState::self()->flushCommands();
+    if (mEfGroup) {
+        EventFlag::deleteEventFlag(&mEfGroup);
+    }
+}
+
+status_t StreamOutHalHidl::getFrameSize(size_t *size) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("getFrameSize", mStream->getFrameSize(), size);
+}
+
+status_t StreamOutHalHidl::getLatency(uint32_t *latency) {
+    if (mStream == 0) return NO_INIT;
+    if (mWriterClient == gettid() && mCommandMQ) {
+        return callWriterThread(
+                WriteCommand::GET_LATENCY, "getLatency", nullptr, 0,
+                [&](const WriteStatus& writeStatus) {
+                    *latency = writeStatus.reply.latencyMs;
+                });
+    } else {
+        return processReturn("getLatency", mStream->getLatency(), latency);
+    }
+}
+
+status_t StreamOutHalHidl::setVolume(float left, float right) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("setVolume", mStream->setVolume(left, right));
+}
+
+status_t StreamOutHalHidl::write(const void *buffer, size_t bytes, size_t *written) {
+    if (mStream == 0) return NO_INIT;
+    *written = 0;
+
+    if (bytes == 0 && !mDataMQ) {
+        // Can't determine the size for the MQ buffer. Wait for a non-empty write request.
+        ALOGW_IF(mCallback.unsafe_get(), "First call to async write with 0 bytes");
+        return OK;
+    }
+
+    status_t status;
+    if (!mDataMQ) {
+        // In case if playback starts close to the end of a compressed track, the bytes
+        // that need to be written is less than the actual buffer size. Need to use
+        // full buffer size for the MQ since otherwise after seeking back to the middle
+        // data will be truncated.
+        size_t bufferSize;
+        if ((status = getCachedBufferSize(&bufferSize)) != OK) {
+            return status;
+        }
+        if (bytes > bufferSize) bufferSize = bytes;
+        if ((status = prepareForWriting(bufferSize)) != OK) {
+            return status;
+        }
+    }
+
+    status = callWriterThread(
+            WriteCommand::WRITE, "write", static_cast<const uint8_t*>(buffer), bytes,
+            [&] (const WriteStatus& writeStatus) {
+                *written = writeStatus.reply.written;
+                // Diagnostics of the cause of b/35813113.
+                ALOGE_IF(*written > bytes,
+                        "hal reports more bytes written than asked for: %lld > %lld",
+                        (long long)*written, (long long)bytes);
+            });
+    mStreamPowerLog.log(buffer, *written);
+    return status;
+}
+
+status_t StreamOutHalHidl::callWriterThread(
+        WriteCommand cmd, const char* cmdName,
+        const uint8_t* data, size_t dataSize, StreamOutHalHidl::WriterCallback callback) {
+    if (!mCommandMQ->write(&cmd)) {
+        ALOGE("command message queue write failed for \"%s\"", cmdName);
+        return -EAGAIN;
+    }
+    if (data != nullptr) {
+        size_t availableToWrite = mDataMQ->availableToWrite();
+        if (dataSize > availableToWrite) {
+            ALOGW("truncating write data from %lld to %lld due to insufficient data queue space",
+                    (long long)dataSize, (long long)availableToWrite);
+            dataSize = availableToWrite;
+        }
+        if (!mDataMQ->write(data, dataSize)) {
+            ALOGE("data message queue write failed for \"%s\"", cmdName);
+        }
+    }
+    mEfGroup->wake(static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY));
+
+    // TODO: Remove manual event flag handling once blocking MQ is implemented. b/33815422
+    uint32_t efState = 0;
+retry:
+    status_t ret = mEfGroup->wait(static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL), &efState);
+    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL)) {
+        WriteStatus writeStatus;
+        writeStatus.retval = Result::NOT_INITIALIZED;
+        if (!mStatusMQ->read(&writeStatus)) {
+            ALOGE("status message read failed for \"%s\"", cmdName);
+        }
+        if (writeStatus.retval == Result::OK) {
+            ret = OK;
+            callback(writeStatus);
+        } else {
+            ret = processReturn(cmdName, writeStatus.retval);
+        }
+        return ret;
+    }
+    if (ret == -EAGAIN || ret == -EINTR) {
+        // Spurious wakeup. This normally retries no more than once.
+        goto retry;
+    }
+    return ret;
+}
+
+status_t StreamOutHalHidl::prepareForWriting(size_t bufferSize) {
+    std::unique_ptr<CommandMQ> tempCommandMQ;
+    std::unique_ptr<DataMQ> tempDataMQ;
+    std::unique_ptr<StatusMQ> tempStatusMQ;
+    Result retval;
+    pid_t halThreadPid, halThreadTid;
+    Return<void> ret = mStream->prepareForWriting(
+            1, bufferSize,
+            [&](Result r,
+                    const CommandMQ::Descriptor& commandMQ,
+                    const DataMQ::Descriptor& dataMQ,
+                    const StatusMQ::Descriptor& statusMQ,
+                    const ThreadInfo& halThreadInfo) {
+                retval = r;
+                if (retval == Result::OK) {
+                    tempCommandMQ.reset(new CommandMQ(commandMQ));
+                    tempDataMQ.reset(new DataMQ(dataMQ));
+                    tempStatusMQ.reset(new StatusMQ(statusMQ));
+                    if (tempDataMQ->isValid() && tempDataMQ->getEventFlagWord()) {
+                        EventFlag::createEventFlag(tempDataMQ->getEventFlagWord(), &mEfGroup);
+                    }
+                    halThreadPid = halThreadInfo.pid;
+                    halThreadTid = halThreadInfo.tid;
+                }
+            });
+    if (!ret.isOk() || retval != Result::OK) {
+        return processReturn("prepareForWriting", ret, retval);
+    }
+    if (!tempCommandMQ || !tempCommandMQ->isValid() ||
+            !tempDataMQ || !tempDataMQ->isValid() ||
+            !tempStatusMQ || !tempStatusMQ->isValid() ||
+            !mEfGroup) {
+        ALOGE_IF(!tempCommandMQ, "Failed to obtain command message queue for writing");
+        ALOGE_IF(tempCommandMQ && !tempCommandMQ->isValid(),
+                "Command message queue for writing is invalid");
+        ALOGE_IF(!tempDataMQ, "Failed to obtain data message queue for writing");
+        ALOGE_IF(tempDataMQ && !tempDataMQ->isValid(), "Data message queue for writing is invalid");
+        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for writing");
+        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
+                "Status message queue for writing is invalid");
+        ALOGE_IF(!mEfGroup, "Event flag creation for writing failed");
+        return NO_INIT;
+    }
+    requestHalThreadPriority(halThreadPid, halThreadTid);
+
+    mCommandMQ = std::move(tempCommandMQ);
+    mDataMQ = std::move(tempDataMQ);
+    mStatusMQ = std::move(tempStatusMQ);
+    mWriterClient = gettid();
+    return OK;
+}
+
+status_t StreamOutHalHidl::getRenderPosition(uint32_t *dspFrames) {
+    if (mStream == 0) return NO_INIT;
+    Result retval;
+    Return<void> ret = mStream->getRenderPosition(
+            [&](Result r, uint32_t d) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *dspFrames = d;
+                }
+            });
+    return processReturn("getRenderPosition", ret, retval);
+}
+
+status_t StreamOutHalHidl::getNextWriteTimestamp(int64_t *timestamp) {
+    if (mStream == 0) return NO_INIT;
+    Result retval;
+    Return<void> ret = mStream->getNextWriteTimestamp(
+            [&](Result r, int64_t t) {
+                retval = r;
+                if (retval == Result::OK) {
+                    *timestamp = t;
+                }
+            });
+    return processReturn("getRenderPosition", ret, retval);
+}
+
+status_t StreamOutHalHidl::setCallback(wp<StreamOutHalInterfaceCallback> callback) {
+    if (mStream == 0) return NO_INIT;
+    status_t status = processReturn(
+            "setCallback", mStream->setCallback(new StreamOutCallback(this)));
+    if (status == OK) {
+        mCallback = callback;
+    }
+    return status;
+}
+
+status_t StreamOutHalHidl::supportsPauseAndResume(bool *supportsPause, bool *supportsResume) {
+    if (mStream == 0) return NO_INIT;
+    Return<void> ret = mStream->supportsPauseAndResume(
+            [&](bool p, bool r) {
+                *supportsPause = p;
+                *supportsResume = r;
+            });
+    return processReturn("supportsPauseAndResume", ret);
+}
+
+status_t StreamOutHalHidl::pause() {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("pause", mStream->pause());
+}
+
+status_t StreamOutHalHidl::resume() {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("pause", mStream->resume());
+}
+
+status_t StreamOutHalHidl::supportsDrain(bool *supportsDrain) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("supportsDrain", mStream->supportsDrain(), supportsDrain);
+}
+
+status_t StreamOutHalHidl::drain(bool earlyNotify) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn(
+            "drain", mStream->drain(earlyNotify ? AudioDrain::EARLY_NOTIFY : AudioDrain::ALL));
+}
+
+status_t StreamOutHalHidl::flush() {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("pause", mStream->flush());
+}
+
+status_t StreamOutHalHidl::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) {
+    if (mStream == 0) return NO_INIT;
+    if (mWriterClient == gettid() && mCommandMQ) {
+        return callWriterThread(
+                WriteCommand::GET_PRESENTATION_POSITION, "getPresentationPosition", nullptr, 0,
+                [&](const WriteStatus& writeStatus) {
+                    *frames = writeStatus.reply.presentationPosition.frames;
+                    timestamp->tv_sec = writeStatus.reply.presentationPosition.timeStamp.tvSec;
+                    timestamp->tv_nsec = writeStatus.reply.presentationPosition.timeStamp.tvNSec;
+                });
+    } else {
+        Result retval;
+        Return<void> ret = mStream->getPresentationPosition(
+                [&](Result r, uint64_t hidlFrames, const TimeSpec& hidlTimeStamp) {
+                    retval = r;
+                    if (retval == Result::OK) {
+                        *frames = hidlFrames;
+                        timestamp->tv_sec = hidlTimeStamp.tvSec;
+                        timestamp->tv_nsec = hidlTimeStamp.tvNSec;
+                    }
+                });
+        return processReturn("getPresentationPosition", ret, retval);
+    }
+}
+
+/** Transform a standard collection to an HIDL vector. */
+template <class Values, class ElementConverter>
+static auto transformToHidlVec(const Values& values, ElementConverter converter) {
+    hidl_vec<decltype(converter(*values.begin()))> result{values.size()};
+    using namespace std;
+    transform(begin(values), end(values), begin(result), converter);
+    return result;
+}
+
+status_t StreamOutHalHidl::updateSourceMetadata(const SourceMetadata& sourceMetadata) {
+    hardware::audio::V4_0::SourceMetadata halMetadata = {
+        .tracks = transformToHidlVec(sourceMetadata.tracks,
+              [](const playback_track_metadata& metadata) -> PlaybackTrackMetadata {
+                  return {
+                    .usage=static_cast<AudioUsage>(metadata.usage),
+                    .contentType=static_cast<AudioContentType>(metadata.content_type),
+                    .gain=metadata.gain,
+                  };
+              })};
+    return processReturn("updateSourceMetadata", mStream->updateSourceMetadata(halMetadata));
+}
+
+void StreamOutHalHidl::onWriteReady() {
+    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
+    if (callback == 0) return;
+    ALOGV("asyncCallback onWriteReady");
+    callback->onWriteReady();
+}
+
+void StreamOutHalHidl::onDrainReady() {
+    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
+    if (callback == 0) return;
+    ALOGV("asyncCallback onDrainReady");
+    callback->onDrainReady();
+}
+
+void StreamOutHalHidl::onError() {
+    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
+    if (callback == 0) return;
+    ALOGV("asyncCallback onError");
+    callback->onError();
+}
+
+
+StreamInHalHidl::StreamInHalHidl(const sp<IStreamIn>& stream)
+        : StreamHalHidl(stream.get()), mStream(stream), mReaderClient(0), mEfGroup(nullptr) {
+}
+
+StreamInHalHidl::~StreamInHalHidl() {
+    if (mStream != 0) {
+        processReturn("close", mStream->close());
+        mStream.clear();
+        hardware::IPCThreadState::self()->flushCommands();
+    }
+    if (mEfGroup) {
+        EventFlag::deleteEventFlag(&mEfGroup);
+    }
+}
+
+status_t StreamInHalHidl::getFrameSize(size_t *size) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("getFrameSize", mStream->getFrameSize(), size);
+}
+
+status_t StreamInHalHidl::setGain(float gain) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("setGain", mStream->setGain(gain));
+}
+
+status_t StreamInHalHidl::read(void *buffer, size_t bytes, size_t *read) {
+    if (mStream == 0) return NO_INIT;
+    *read = 0;
+
+    if (bytes == 0 && !mDataMQ) {
+        // Can't determine the size for the MQ buffer. Wait for a non-empty read request.
+        return OK;
+    }
+
+    status_t status;
+    if (!mDataMQ && (status = prepareForReading(bytes)) != OK) {
+        return status;
+    }
+
+    ReadParameters params;
+    params.command = ReadCommand::READ;
+    params.params.read = bytes;
+    status = callReaderThread(params, "read",
+            [&](const ReadStatus& readStatus) {
+                const size_t availToRead = mDataMQ->availableToRead();
+                if (!mDataMQ->read(static_cast<uint8_t*>(buffer), std::min(bytes, availToRead))) {
+                    ALOGE("data message queue read failed for \"read\"");
+                }
+                ALOGW_IF(availToRead != readStatus.reply.read,
+                        "HAL read report inconsistent: mq = %d, status = %d",
+                        (int32_t)availToRead, (int32_t)readStatus.reply.read);
+                *read = readStatus.reply.read;
+            });
+    mStreamPowerLog.log(buffer, *read);
+    return status;
+}
+
+status_t StreamInHalHidl::callReaderThread(
+        const ReadParameters& params, const char* cmdName,
+        StreamInHalHidl::ReaderCallback callback) {
+    if (!mCommandMQ->write(&params)) {
+        ALOGW("command message queue write failed");
+        return -EAGAIN;
+    }
+    mEfGroup->wake(static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL));
+
+    // TODO: Remove manual event flag handling once blocking MQ is implemented. b/33815422
+    uint32_t efState = 0;
+retry:
+    status_t ret = mEfGroup->wait(static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY), &efState);
+    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY)) {
+        ReadStatus readStatus;
+        readStatus.retval = Result::NOT_INITIALIZED;
+        if (!mStatusMQ->read(&readStatus)) {
+            ALOGE("status message read failed for \"%s\"", cmdName);
+        }
+         if (readStatus.retval == Result::OK) {
+            ret = OK;
+            callback(readStatus);
+        } else {
+            ret = processReturn(cmdName, readStatus.retval);
+        }
+        return ret;
+    }
+    if (ret == -EAGAIN || ret == -EINTR) {
+        // Spurious wakeup. This normally retries no more than once.
+        goto retry;
+    }
+    return ret;
+}
+
+status_t StreamInHalHidl::prepareForReading(size_t bufferSize) {
+    std::unique_ptr<CommandMQ> tempCommandMQ;
+    std::unique_ptr<DataMQ> tempDataMQ;
+    std::unique_ptr<StatusMQ> tempStatusMQ;
+    Result retval;
+    pid_t halThreadPid, halThreadTid;
+    Return<void> ret = mStream->prepareForReading(
+            1, bufferSize,
+            [&](Result r,
+                    const CommandMQ::Descriptor& commandMQ,
+                    const DataMQ::Descriptor& dataMQ,
+                    const StatusMQ::Descriptor& statusMQ,
+                    const ThreadInfo& halThreadInfo) {
+                retval = r;
+                if (retval == Result::OK) {
+                    tempCommandMQ.reset(new CommandMQ(commandMQ));
+                    tempDataMQ.reset(new DataMQ(dataMQ));
+                    tempStatusMQ.reset(new StatusMQ(statusMQ));
+                    if (tempDataMQ->isValid() && tempDataMQ->getEventFlagWord()) {
+                        EventFlag::createEventFlag(tempDataMQ->getEventFlagWord(), &mEfGroup);
+                    }
+                    halThreadPid = halThreadInfo.pid;
+                    halThreadTid = halThreadInfo.tid;
+                }
+            });
+    if (!ret.isOk() || retval != Result::OK) {
+        return processReturn("prepareForReading", ret, retval);
+    }
+    if (!tempCommandMQ || !tempCommandMQ->isValid() ||
+            !tempDataMQ || !tempDataMQ->isValid() ||
+            !tempStatusMQ || !tempStatusMQ->isValid() ||
+            !mEfGroup) {
+        ALOGE_IF(!tempCommandMQ, "Failed to obtain command message queue for writing");
+        ALOGE_IF(tempCommandMQ && !tempCommandMQ->isValid(),
+                "Command message queue for writing is invalid");
+        ALOGE_IF(!tempDataMQ, "Failed to obtain data message queue for reading");
+        ALOGE_IF(tempDataMQ && !tempDataMQ->isValid(), "Data message queue for reading is invalid");
+        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for reading");
+        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
+                "Status message queue for reading is invalid");
+        ALOGE_IF(!mEfGroup, "Event flag creation for reading failed");
+        return NO_INIT;
+    }
+    requestHalThreadPriority(halThreadPid, halThreadTid);
+
+    mCommandMQ = std::move(tempCommandMQ);
+    mDataMQ = std::move(tempDataMQ);
+    mStatusMQ = std::move(tempStatusMQ);
+    mReaderClient = gettid();
+    return OK;
+}
+
+status_t StreamInHalHidl::getInputFramesLost(uint32_t *framesLost) {
+    if (mStream == 0) return NO_INIT;
+    return processReturn("getInputFramesLost", mStream->getInputFramesLost(), framesLost);
+}
+
+status_t StreamInHalHidl::getCapturePosition(int64_t *frames, int64_t *time) {
+    if (mStream == 0) return NO_INIT;
+    if (mReaderClient == gettid() && mCommandMQ) {
+        ReadParameters params;
+        params.command = ReadCommand::GET_CAPTURE_POSITION;
+        return callReaderThread(params, "getCapturePosition",
+                [&](const ReadStatus& readStatus) {
+                    *frames = readStatus.reply.capturePosition.frames;
+                    *time = readStatus.reply.capturePosition.time;
+                });
+    } else {
+        Result retval;
+        Return<void> ret = mStream->getCapturePosition(
+                [&](Result r, uint64_t hidlFrames, uint64_t hidlTime) {
+                    retval = r;
+                    if (retval == Result::OK) {
+                        *frames = hidlFrames;
+                        *time = hidlTime;
+                    }
+                });
+        return processReturn("getCapturePosition", ret, retval);
+    }
+}
+
+
+status_t StreamInHalHidl::getActiveMicrophones(
+        std::vector<media::MicrophoneInfo> *microphonesInfo) {
+    if (!mStream) return NO_INIT;
+    Result retval;
+    Return<void> ret = mStream->getActiveMicrophones(
+            [&](Result r, hidl_vec<MicrophoneInfo> micArrayHal) {
+        retval = r;
+        for (size_t k = 0; k < micArrayHal.size(); k++) {
+            audio_microphone_characteristic_t dst;
+            // convert
+            microphoneInfoToHal(micArrayHal[k], &dst);
+            media::MicrophoneInfo microphone = media::MicrophoneInfo(dst);
+            microphonesInfo->push_back(microphone);
+        }
+    });
+    return processReturn("getActiveMicrophones", ret, retval);
+}
+
+status_t StreamInHalHidl::updateSinkMetadata(const SinkMetadata& sinkMetadata) {
+    hardware::audio::V4_0::SinkMetadata halMetadata = {
+        .tracks = transformToHidlVec(sinkMetadata.tracks,
+              [](const record_track_metadata& metadata) -> RecordTrackMetadata {
+                  return {
+                    .source=static_cast<AudioSource>(metadata.source),
+                    .gain=metadata.gain,
+                  };
+              })};
+    return processReturn("updateSinkMetadata", mStream->updateSinkMetadata(halMetadata));
+}
+
+} // namespace V4_0
+} // namespace android

diff --git a/media/libaudiohal/4.0/StreamHalHidl.h b/media/libaudiohal/4.0/StreamHalHidl.h
new file mode 100644
index 0000000..2dda0f8
--- /dev/null
+++ b/media/libaudiohal/4.0/StreamHalHidl.h

@@ -0,0 +1,250 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_STREAM_HAL_HIDL_4_0_H
+#define ANDROID_HARDWARE_STREAM_HAL_HIDL_4_0_H
+
+#include <atomic>
+
+#include <android/hardware/audio/4.0/IStream.h>
+#include <android/hardware/audio/4.0/IStreamIn.h>
+#include <android/hardware/audio/4.0/IStreamOut.h>
+#include <fmq/EventFlag.h>
+#include <fmq/MessageQueue.h>
+#include <media/audiohal/StreamHalInterface.h>
+
+#include "ConversionHelperHidl.h"
+#include "StreamPowerLog.h"
+
+using ::android::hardware::audio::V4_0::IStream;
+using ::android::hardware::audio::V4_0::IStreamIn;
+using ::android::hardware::audio::V4_0::IStreamOut;
+using ::android::hardware::EventFlag;
+using ::android::hardware::MessageQueue;
+using ::android::hardware::Return;
+using ReadParameters = ::android::hardware::audio::V4_0::IStreamIn::ReadParameters;
+using ReadStatus = ::android::hardware::audio::V4_0::IStreamIn::ReadStatus;
+using WriteCommand = ::android::hardware::audio::V4_0::IStreamOut::WriteCommand;
+using WriteStatus = ::android::hardware::audio::V4_0::IStreamOut::WriteStatus;
+
+namespace android {
+namespace V4_0 {
+
+class DeviceHalHidl;
+
+class StreamHalHidl : public virtual StreamHalInterface, public ConversionHelperHidl
+{
+  public:
+    // Return the sampling rate in Hz - eg. 44100.
+    virtual status_t getSampleRate(uint32_t *rate);
+
+    // Return size of input/output buffer in bytes for this stream - eg. 4800.
+    virtual status_t getBufferSize(size_t *size);
+
+    // Return the channel mask.
+    virtual status_t getChannelMask(audio_channel_mask_t *mask);
+
+    // Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT.
+    virtual status_t getFormat(audio_format_t *format);
+
+    // Convenience method.
+    virtual status_t getAudioProperties(
+            uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format);
+
+    // Set audio stream parameters.
+    virtual status_t setParameters(const String8& kvPairs);
+
+    // Get audio stream parameters.
+    virtual status_t getParameters(const String8& keys, String8 *values);
+
+    // Add or remove the effect on the stream.
+    virtual status_t addEffect(sp<EffectHalInterface> effect);
+    virtual status_t removeEffect(sp<EffectHalInterface> effect);
+
+    // Put the audio hardware input/output into standby mode.
+    virtual status_t standby();
+
+    virtual status_t dump(int fd);
+
+    // Start a stream operating in mmap mode.
+    virtual status_t start();
+
+    // Stop a stream operating in mmap mode.
+    virtual status_t stop();
+
+    // Retrieve information on the data buffer in mmap mode.
+    virtual status_t createMmapBuffer(int32_t minSizeFrames,
+                                      struct audio_mmap_buffer_info *info);
+
+    // Get current read/write position in the mmap buffer
+    virtual status_t getMmapPosition(struct audio_mmap_position *position);
+
+    // Set the priority of the thread that interacts with the HAL
+    // (must match the priority of the audioflinger's thread that calls 'read' / 'write')
+    virtual status_t setHalThreadPriority(int priority);
+
+  protected:
+    // Subclasses can not be constructed directly by clients.
+    explicit StreamHalHidl(IStream *stream);
+
+    // The destructor automatically closes the stream.
+    virtual ~StreamHalHidl();
+
+    status_t getCachedBufferSize(size_t *size);
+
+    bool requestHalThreadPriority(pid_t threadPid, pid_t threadId);
+
+    // mStreamPowerLog is used for audio signal power logging.
+    StreamPowerLog mStreamPowerLog;
+
+  private:
+    const int HAL_THREAD_PRIORITY_DEFAULT = -1;
+    IStream *mStream;
+    int mHalThreadPriority;
+    size_t mCachedBufferSize;
+};
+
+class StreamOutHalHidl : public StreamOutHalInterface, public StreamHalHidl {
+  public:
+    // Return the frame size (number of bytes per sample) of a stream.
+    virtual status_t getFrameSize(size_t *size);
+
+    // Return the audio hardware driver estimated latency in milliseconds.
+    virtual status_t getLatency(uint32_t *latency);
+
+    // Use this method in situations where audio mixing is done in the hardware.
+    virtual status_t setVolume(float left, float right);
+
+    // Write audio buffer to driver.
+    virtual status_t write(const void *buffer, size_t bytes, size_t *written);
+
+    // Return the number of audio frames written by the audio dsp to DAC since
+    // the output has exited standby.
+    virtual status_t getRenderPosition(uint32_t *dspFrames);
+
+    // Get the local time at which the next write to the audio driver will be presented.
+    virtual status_t getNextWriteTimestamp(int64_t *timestamp);
+
+    // Set the callback for notifying completion of non-blocking write and drain.
+    virtual status_t setCallback(wp<StreamOutHalInterfaceCallback> callback);
+
+    // Returns whether pause and resume operations are supported.
+    virtual status_t supportsPauseAndResume(bool *supportsPause, bool *supportsResume);
+
+    // Notifies to the audio driver to resume playback following a pause.
+    virtual status_t pause();
+
+    // Notifies to the audio driver to resume playback following a pause.
+    virtual status_t resume();
+
+    // Returns whether drain operation is supported.
+    virtual status_t supportsDrain(bool *supportsDrain);
+
+    // Requests notification when data buffered by the driver/hardware has been played.
+    virtual status_t drain(bool earlyNotify);
+
+    // Notifies to the audio driver to flush the queued data.
+    virtual status_t flush();
+
+    // Return a recent count of the number of audio frames presented to an external observer.
+    virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
+
+    // Called when the metadata of the stream's source has been changed.
+    status_t updateSourceMetadata(const SourceMetadata& sourceMetadata) override;
+
+    // Methods used by StreamOutCallback (HIDL).
+    void onWriteReady();
+    void onDrainReady();
+    void onError();
+
+  private:
+    friend class DeviceHalHidl;
+    typedef MessageQueue<WriteCommand, hardware::kSynchronizedReadWrite> CommandMQ;
+    typedef MessageQueue<uint8_t, hardware::kSynchronizedReadWrite> DataMQ;
+    typedef MessageQueue<WriteStatus, hardware::kSynchronizedReadWrite> StatusMQ;
+
+    wp<StreamOutHalInterfaceCallback> mCallback;
+    sp<IStreamOut> mStream;
+    std::unique_ptr<CommandMQ> mCommandMQ;
+    std::unique_ptr<DataMQ> mDataMQ;
+    std::unique_ptr<StatusMQ> mStatusMQ;
+    std::atomic<pid_t> mWriterClient;
+    EventFlag* mEfGroup;
+
+    // Can not be constructed directly by clients.
+    StreamOutHalHidl(const sp<IStreamOut>& stream);
+
+    virtual ~StreamOutHalHidl();
+
+    using WriterCallback = std::function<void(const WriteStatus& writeStatus)>;
+    status_t callWriterThread(
+            WriteCommand cmd, const char* cmdName,
+            const uint8_t* data, size_t dataSize, WriterCallback callback);
+    status_t prepareForWriting(size_t bufferSize);
+};
+
+class StreamInHalHidl : public StreamInHalInterface, public StreamHalHidl {
+  public:
+    // Return the frame size (number of bytes per sample) of a stream.
+    virtual status_t getFrameSize(size_t *size);
+
+    // Set the input gain for the audio driver.
+    virtual status_t setGain(float gain);
+
+    // Read audio buffer in from driver.
+    virtual status_t read(void *buffer, size_t bytes, size_t *read);
+
+    // Return the amount of input frames lost in the audio driver.
+    virtual status_t getInputFramesLost(uint32_t *framesLost);
+
+    // Return a recent count of the number of audio frames received and
+    // the clock time associated with that frame count.
+    virtual status_t getCapturePosition(int64_t *frames, int64_t *time);
+
+    // Get active microphones
+    virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+
+    // Called when the metadata of the stream's sink has been changed.
+    status_t updateSinkMetadata(const SinkMetadata& sinkMetadata) override;
+
+  private:
+    friend class DeviceHalHidl;
+    typedef MessageQueue<ReadParameters, hardware::kSynchronizedReadWrite> CommandMQ;
+    typedef MessageQueue<uint8_t, hardware::kSynchronizedReadWrite> DataMQ;
+    typedef MessageQueue<ReadStatus, hardware::kSynchronizedReadWrite> StatusMQ;
+
+    sp<IStreamIn> mStream;
+    std::unique_ptr<CommandMQ> mCommandMQ;
+    std::unique_ptr<DataMQ> mDataMQ;
+    std::unique_ptr<StatusMQ> mStatusMQ;
+    std::atomic<pid_t> mReaderClient;
+    EventFlag* mEfGroup;
+
+    // Can not be constructed directly by clients.
+    StreamInHalHidl(const sp<IStreamIn>& stream);
+
+    virtual ~StreamInHalHidl();
+
+    using ReaderCallback = std::function<void(const ReadStatus& readStatus)>;
+    status_t callReaderThread(
+            const ReadParameters& params, const char* cmdName, ReaderCallback callback);
+    status_t prepareForReading(size_t bufferSize);
+};
+
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_STREAM_HAL_HIDL_4_0_H

diff --git a/media/libaudiohal/4.0/StreamHalLocal.cpp b/media/libaudiohal/4.0/StreamHalLocal.cpp
new file mode 100644
index 0000000..e9d96bf
--- /dev/null
+++ b/media/libaudiohal/4.0/StreamHalLocal.cpp

@@ -0,0 +1,357 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "StreamHalLocal"
+//#define LOG_NDEBUG 0
+
+#include <hardware/audio.h>
+#include <utils/Log.h>
+
+#include "DeviceHalLocal.h"
+#include "StreamHalLocal.h"
+#include "VersionUtils.h"
+
+namespace android {
+namespace V4_0 {
+
+StreamHalLocal::StreamHalLocal(audio_stream_t *stream, sp<DeviceHalLocal> device)
+        : mDevice(device),
+          mStream(stream) {
+    // Instrument audio signal power logging.
+    // Note: This assumes channel mask, format, and sample rate do not change after creation.
+    if (mStream != nullptr && mStreamPowerLog.isUserDebugOrEngBuild()) {
+        mStreamPowerLog.init(mStream->get_sample_rate(mStream),
+                mStream->get_channels(mStream),
+                mStream->get_format(mStream));
+    }
+}
+
+StreamHalLocal::~StreamHalLocal() {
+    mStream = 0;
+    mDevice.clear();
+}
+
+status_t StreamHalLocal::getSampleRate(uint32_t *rate) {
+    *rate = mStream->get_sample_rate(mStream);
+    return OK;
+}
+
+status_t StreamHalLocal::getBufferSize(size_t *size) {
+    *size = mStream->get_buffer_size(mStream);
+    return OK;
+}
+
+status_t StreamHalLocal::getChannelMask(audio_channel_mask_t *mask) {
+    *mask = mStream->get_channels(mStream);
+    return OK;
+}
+
+status_t StreamHalLocal::getFormat(audio_format_t *format) {
+    *format = mStream->get_format(mStream);
+    return OK;
+}
+
+status_t StreamHalLocal::getAudioProperties(
+        uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format) {
+    *sampleRate = mStream->get_sample_rate(mStream);
+    *mask = mStream->get_channels(mStream);
+    *format = mStream->get_format(mStream);
+    return OK;
+}
+
+status_t StreamHalLocal::setParameters(const String8& kvPairs) {
+    return mStream->set_parameters(mStream, kvPairs.string());
+}
+
+status_t StreamHalLocal::getParameters(const String8& keys, String8 *values) {
+    char *halValues = mStream->get_parameters(mStream, keys.string());
+    if (halValues != NULL) {
+        values->setTo(halValues);
+        free(halValues);
+    } else {
+        values->clear();
+    }
+    return OK;
+}
+
+status_t StreamHalLocal::addEffect(sp<EffectHalInterface>) {
+    LOG_ALWAYS_FATAL("Local streams can not have effects");
+    return INVALID_OPERATION;
+}
+
+status_t StreamHalLocal::removeEffect(sp<EffectHalInterface>) {
+    LOG_ALWAYS_FATAL("Local streams can not have effects");
+    return INVALID_OPERATION;
+}
+
+status_t StreamHalLocal::standby() {
+    return mStream->standby(mStream);
+}
+
+status_t StreamHalLocal::dump(int fd) {
+    status_t status = mStream->dump(mStream, fd);
+    mStreamPowerLog.dump(fd);
+    return status;
+}
+
+status_t StreamHalLocal::setHalThreadPriority(int) {
+    // Don't need to do anything as local hal is executed by audioflinger directly
+    // on the same thread.
+    return OK;
+}
+
+StreamOutHalLocal::StreamOutHalLocal(audio_stream_out_t *stream, sp<DeviceHalLocal> device)
+        : StreamHalLocal(&stream->common, device), mStream(stream) {
+}
+
+StreamOutHalLocal::~StreamOutHalLocal() {
+    mCallback.clear();
+    mDevice->closeOutputStream(mStream);
+    mStream = 0;
+}
+
+status_t StreamOutHalLocal::getFrameSize(size_t *size) {
+    *size = audio_stream_out_frame_size(mStream);
+    return OK;
+}
+
+status_t StreamOutHalLocal::getLatency(uint32_t *latency) {
+    *latency = mStream->get_latency(mStream);
+    return OK;
+}
+
+status_t StreamOutHalLocal::setVolume(float left, float right) {
+    if (mStream->set_volume == NULL) return INVALID_OPERATION;
+    return mStream->set_volume(mStream, left, right);
+}
+
+status_t StreamOutHalLocal::write(const void *buffer, size_t bytes, size_t *written) {
+    ssize_t writeResult = mStream->write(mStream, buffer, bytes);
+    if (writeResult > 0) {
+        *written = writeResult;
+        mStreamPowerLog.log(buffer, *written);
+        return OK;
+    } else {
+        *written = 0;
+        return writeResult;
+    }
+}
+
+status_t StreamOutHalLocal::getRenderPosition(uint32_t *dspFrames) {
+    return mStream->get_render_position(mStream, dspFrames);
+}
+
+status_t StreamOutHalLocal::getNextWriteTimestamp(int64_t *timestamp) {
+    if (mStream->get_next_write_timestamp == NULL) return INVALID_OPERATION;
+    return mStream->get_next_write_timestamp(mStream, timestamp);
+}
+
+status_t StreamOutHalLocal::setCallback(wp<StreamOutHalInterfaceCallback> callback) {
+    if (mStream->set_callback == NULL) return INVALID_OPERATION;
+    status_t result = mStream->set_callback(mStream, StreamOutHalLocal::asyncCallback, this);
+    if (result == OK) {
+        mCallback = callback;
+    }
+    return result;
+}
+
+// static
+int StreamOutHalLocal::asyncCallback(stream_callback_event_t event, void*, void *cookie) {
+    // We act as if we gave a wp<StreamOutHalLocal> to HAL. This way we should handle
+    // correctly the case when the callback is invoked while StreamOutHalLocal's destructor is
+    // already running, because the destructor is invoked after the refcount has been atomically
+    // decremented.
+    wp<StreamOutHalLocal> weakSelf(static_cast<StreamOutHalLocal*>(cookie));
+    sp<StreamOutHalLocal> self = weakSelf.promote();
+    if (self == 0) return 0;
+    sp<StreamOutHalInterfaceCallback> callback = self->mCallback.promote();
+    if (callback == 0) return 0;
+    ALOGV("asyncCallback() event %d", event);
+    switch (event) {
+        case STREAM_CBK_EVENT_WRITE_READY:
+            callback->onWriteReady();
+            break;
+        case STREAM_CBK_EVENT_DRAIN_READY:
+            callback->onDrainReady();
+            break;
+        case STREAM_CBK_EVENT_ERROR:
+            callback->onError();
+            break;
+        default:
+            ALOGW("asyncCallback() unknown event %d", event);
+            break;
+    }
+    return 0;
+}
+
+status_t StreamOutHalLocal::supportsPauseAndResume(bool *supportsPause, bool *supportsResume) {
+    *supportsPause = mStream->pause != NULL;
+    *supportsResume = mStream->resume != NULL;
+    return OK;
+}
+
+status_t StreamOutHalLocal::pause() {
+    if (mStream->pause == NULL) return INVALID_OPERATION;
+    return mStream->pause(mStream);
+}
+
+status_t StreamOutHalLocal::resume() {
+    if (mStream->resume == NULL) return INVALID_OPERATION;
+    return mStream->resume(mStream);
+}
+
+status_t StreamOutHalLocal::supportsDrain(bool *supportsDrain) {
+    *supportsDrain = mStream->drain != NULL;
+    return OK;
+}
+
+status_t StreamOutHalLocal::drain(bool earlyNotify) {
+    if (mStream->drain == NULL) return INVALID_OPERATION;
+    return mStream->drain(mStream, earlyNotify ? AUDIO_DRAIN_EARLY_NOTIFY : AUDIO_DRAIN_ALL);
+}
+
+status_t StreamOutHalLocal::flush() {
+    if (mStream->flush == NULL) return INVALID_OPERATION;
+    return mStream->flush(mStream);
+}
+
+status_t StreamOutHalLocal::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) {
+    if (mStream->get_presentation_position == NULL) return INVALID_OPERATION;
+    return mStream->get_presentation_position(mStream, frames, timestamp);
+}
+
+status_t StreamOutHalLocal::updateSourceMetadata(const SourceMetadata& sourceMetadata) {
+    if (mStream->update_source_metadata == nullptr) {
+        return INVALID_OPERATION;
+    }
+    const source_metadata_t metadata {
+        .track_count = sourceMetadata.tracks.size(),
+        // const cast is fine as it is in a const structure
+        .tracks = const_cast<playback_track_metadata*>(sourceMetadata.tracks.data()),
+    };
+    mStream->update_source_metadata(mStream, &metadata);
+    return OK;
+}
+
+status_t StreamOutHalLocal::start() {
+    if (mStream->start == NULL) return INVALID_OPERATION;
+    return mStream->start(mStream);
+}
+
+status_t StreamOutHalLocal::stop() {
+    if (mStream->stop == NULL) return INVALID_OPERATION;
+    return mStream->stop(mStream);
+}
+
+status_t StreamOutHalLocal::createMmapBuffer(int32_t minSizeFrames,
+                                  struct audio_mmap_buffer_info *info) {
+    if (mStream->create_mmap_buffer == NULL) return INVALID_OPERATION;
+    return mStream->create_mmap_buffer(mStream, minSizeFrames, info);
+}
+
+status_t StreamOutHalLocal::getMmapPosition(struct audio_mmap_position *position) {
+    if (mStream->get_mmap_position == NULL) return INVALID_OPERATION;
+    return mStream->get_mmap_position(mStream, position);
+}
+
+StreamInHalLocal::StreamInHalLocal(audio_stream_in_t *stream, sp<DeviceHalLocal> device)
+        : StreamHalLocal(&stream->common, device), mStream(stream) {
+}
+
+StreamInHalLocal::~StreamInHalLocal() {
+    mDevice->closeInputStream(mStream);
+    mStream = 0;
+}
+
+status_t StreamInHalLocal::getFrameSize(size_t *size) {
+    *size = audio_stream_in_frame_size(mStream);
+    return OK;
+}
+
+status_t StreamInHalLocal::setGain(float gain) {
+    return mStream->set_gain(mStream, gain);
+}
+
+status_t StreamInHalLocal::read(void *buffer, size_t bytes, size_t *read) {
+    ssize_t readResult = mStream->read(mStream, buffer, bytes);
+    if (readResult > 0) {
+        *read = readResult;
+        mStreamPowerLog.log( buffer, *read);
+        return OK;
+    } else {
+        *read = 0;
+        return readResult;
+    }
+}
+
+status_t StreamInHalLocal::getInputFramesLost(uint32_t *framesLost) {
+    *framesLost = mStream->get_input_frames_lost(mStream);
+    return OK;
+}
+
+status_t StreamInHalLocal::getCapturePosition(int64_t *frames, int64_t *time) {
+    if (mStream->get_capture_position == NULL) return INVALID_OPERATION;
+    return mStream->get_capture_position(mStream, frames, time);
+}
+
+status_t StreamInHalLocal::updateSinkMetadata(const SinkMetadata& sinkMetadata) {
+    if (mStream->update_sink_metadata == nullptr) {
+        return INVALID_OPERATION;
+    }
+    const sink_metadata_t metadata {
+        .track_count = sinkMetadata.tracks.size(),
+        // const cast is fine as it is in a const structure
+        .tracks = const_cast<record_track_metadata*>(sinkMetadata.tracks.data()),
+    };
+    mStream->update_sink_metadata(mStream, &metadata);
+    return OK;
+}
+
+status_t StreamInHalLocal::start() {
+    if (mStream->start == NULL) return INVALID_OPERATION;
+    return mStream->start(mStream);
+}
+
+status_t StreamInHalLocal::stop() {
+    if (mStream->stop == NULL) return INVALID_OPERATION;
+    return mStream->stop(mStream);
+}
+
+status_t StreamInHalLocal::createMmapBuffer(int32_t minSizeFrames,
+                                  struct audio_mmap_buffer_info *info) {
+    if (mStream->create_mmap_buffer == NULL) return INVALID_OPERATION;
+    return mStream->create_mmap_buffer(mStream, minSizeFrames, info);
+}
+
+status_t StreamInHalLocal::getMmapPosition(struct audio_mmap_position *position) {
+    if (mStream->get_mmap_position == NULL) return INVALID_OPERATION;
+    return mStream->get_mmap_position(mStream, position);
+}
+
+status_t StreamInHalLocal::getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones) {
+    if (mStream->get_active_microphones == NULL) return INVALID_OPERATION;
+    size_t actual_mics = AUDIO_MICROPHONE_MAX_COUNT;
+    audio_microphone_characteristic_t mic_array[AUDIO_MICROPHONE_MAX_COUNT];
+    status_t status = mStream->get_active_microphones(mStream, &mic_array[0], &actual_mics);
+    for (size_t i = 0; i < actual_mics; i++) {
+        media::MicrophoneInfo microphoneInfo = media::MicrophoneInfo(mic_array[i]);
+        microphones->push_back(microphoneInfo);
+    }
+    return status;
+}
+
+} // namespace V4_0
+} // namespace android

diff --git a/media/libaudiohal/4.0/StreamHalLocal.h b/media/libaudiohal/4.0/StreamHalLocal.h
new file mode 100644
index 0000000..7237509
--- /dev/null
+++ b/media/libaudiohal/4.0/StreamHalLocal.h

@@ -0,0 +1,221 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_STREAM_HAL_LOCAL_4_0_H
+#define ANDROID_HARDWARE_STREAM_HAL_LOCAL_4_0_H
+
+#include <media/audiohal/StreamHalInterface.h>
+#include "StreamPowerLog.h"
+
+namespace android {
+namespace V4_0 {
+
+class DeviceHalLocal;
+
+class StreamHalLocal : public virtual StreamHalInterface
+{
+  public:
+    // Return the sampling rate in Hz - eg. 44100.
+    virtual status_t getSampleRate(uint32_t *rate);
+
+    // Return size of input/output buffer in bytes for this stream - eg. 4800.
+    virtual status_t getBufferSize(size_t *size);
+
+    // Return the channel mask.
+    virtual status_t getChannelMask(audio_channel_mask_t *mask);
+
+    // Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT.
+    virtual status_t getFormat(audio_format_t *format);
+
+    // Convenience method.
+    virtual status_t getAudioProperties(
+            uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format);
+
+    // Set audio stream parameters.
+    virtual status_t setParameters(const String8& kvPairs);
+
+    // Get audio stream parameters.
+    virtual status_t getParameters(const String8& keys, String8 *values);
+
+    // Add or remove the effect on the stream.
+    virtual status_t addEffect(sp<EffectHalInterface> effect);
+    virtual status_t removeEffect(sp<EffectHalInterface> effect);
+
+    // Put the audio hardware input/output into standby mode.
+    virtual status_t standby();
+
+    virtual status_t dump(int fd);
+
+    // Start a stream operating in mmap mode.
+    virtual status_t start() = 0;
+
+    // Stop a stream operating in mmap mode.
+    virtual status_t stop() = 0;
+
+    // Retrieve information on the data buffer in mmap mode.
+    virtual status_t createMmapBuffer(int32_t minSizeFrames,
+                                      struct audio_mmap_buffer_info *info) = 0;
+
+    // Get current read/write position in the mmap buffer
+    virtual status_t getMmapPosition(struct audio_mmap_position *position) = 0;
+
+    // Set the priority of the thread that interacts with the HAL
+    // (must match the priority of the audioflinger's thread that calls 'read' / 'write')
+    virtual status_t setHalThreadPriority(int priority);
+
+  protected:
+    // Subclasses can not be constructed directly by clients.
+    StreamHalLocal(audio_stream_t *stream, sp<DeviceHalLocal> device);
+
+    // The destructor automatically closes the stream.
+    virtual ~StreamHalLocal();
+
+    sp<DeviceHalLocal> mDevice;
+
+    // mStreamPowerLog is used for audio signal power logging.
+    StreamPowerLog mStreamPowerLog;
+
+  private:
+    audio_stream_t *mStream;
+};
+
+class StreamOutHalLocal : public StreamOutHalInterface, public StreamHalLocal {
+  public:
+    // Return the frame size (number of bytes per sample) of a stream.
+    virtual status_t getFrameSize(size_t *size);
+
+    // Return the audio hardware driver estimated latency in milliseconds.
+    virtual status_t getLatency(uint32_t *latency);
+
+    // Use this method in situations where audio mixing is done in the hardware.
+    virtual status_t setVolume(float left, float right);
+
+    // Write audio buffer to driver.
+    virtual status_t write(const void *buffer, size_t bytes, size_t *written);
+
+    // Return the number of audio frames written by the audio dsp to DAC since
+    // the output has exited standby.
+    virtual status_t getRenderPosition(uint32_t *dspFrames);
+
+    // Get the local time at which the next write to the audio driver will be presented.
+    virtual status_t getNextWriteTimestamp(int64_t *timestamp);
+
+    // Set the callback for notifying completion of non-blocking write and drain.
+    virtual status_t setCallback(wp<StreamOutHalInterfaceCallback> callback);
+
+    // Returns whether pause and resume operations are supported.
+    virtual status_t supportsPauseAndResume(bool *supportsPause, bool *supportsResume);
+
+    // Notifies to the audio driver to resume playback following a pause.
+    virtual status_t pause();
+
+    // Notifies to the audio driver to resume playback following a pause.
+    virtual status_t resume();
+
+    // Returns whether drain operation is supported.
+    virtual status_t supportsDrain(bool *supportsDrain);
+
+    // Requests notification when data buffered by the driver/hardware has been played.
+    virtual status_t drain(bool earlyNotify);
+
+    // Notifies to the audio driver to flush the queued data.
+    virtual status_t flush();
+
+    // Return a recent count of the number of audio frames presented to an external observer.
+    virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
+
+    // Start a stream operating in mmap mode.
+    virtual status_t start();
+
+    // Stop a stream operating in mmap mode.
+    virtual status_t stop();
+
+    // Retrieve information on the data buffer in mmap mode.
+    virtual status_t createMmapBuffer(int32_t minSizeFrames,
+                                      struct audio_mmap_buffer_info *info);
+
+    // Get current read/write position in the mmap buffer
+    virtual status_t getMmapPosition(struct audio_mmap_position *position);
+
+    // Called when the metadata of the stream's source has been changed.
+    status_t updateSourceMetadata(const SourceMetadata& sourceMetadata) override;
+
+  private:
+    audio_stream_out_t *mStream;
+    wp<StreamOutHalInterfaceCallback> mCallback;
+
+    friend class DeviceHalLocal;
+
+    // Can not be constructed directly by clients.
+    StreamOutHalLocal(audio_stream_out_t *stream, sp<DeviceHalLocal> device);
+
+    virtual ~StreamOutHalLocal();
+
+    static int asyncCallback(stream_callback_event_t event, void *param, void *cookie);
+};
+
+class StreamInHalLocal : public StreamInHalInterface, public StreamHalLocal {
+  public:
+    // Return the frame size (number of bytes per sample) of a stream.
+    virtual status_t getFrameSize(size_t *size);
+
+    // Set the input gain for the audio driver.
+    virtual status_t setGain(float gain);
+
+    // Read audio buffer in from driver.
+    virtual status_t read(void *buffer, size_t bytes, size_t *read);
+
+    // Return the amount of input frames lost in the audio driver.
+    virtual status_t getInputFramesLost(uint32_t *framesLost);
+
+    // Return a recent count of the number of audio frames received and
+    // the clock time associated with that frame count.
+    virtual status_t getCapturePosition(int64_t *frames, int64_t *time);
+
+    // Start a stream operating in mmap mode.
+    virtual status_t start();
+
+    // Stop a stream operating in mmap mode.
+    virtual status_t stop();
+
+    // Retrieve information on the data buffer in mmap mode.
+    virtual status_t createMmapBuffer(int32_t minSizeFrames,
+                                      struct audio_mmap_buffer_info *info);
+
+    // Get current read/write position in the mmap buffer
+    virtual status_t getMmapPosition(struct audio_mmap_position *position);
+
+    // Get active microphones
+    virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+
+    // Called when the metadata of the stream's sink has been changed.
+    status_t updateSinkMetadata(const SinkMetadata& sinkMetadata) override;
+
+  private:
+    audio_stream_in_t *mStream;
+
+    friend class DeviceHalLocal;
+
+    // Can not be constructed directly by clients.
+    StreamInHalLocal(audio_stream_in_t *stream, sp<DeviceHalLocal> device);
+
+    virtual ~StreamInHalLocal();
+};
+
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_STREAM_HAL_LOCAL_4_0_H

diff --git a/media/libaudiohal/4.0/StreamPowerLog.h b/media/libaudiohal/4.0/StreamPowerLog.h
new file mode 100644
index 0000000..57b7201
--- /dev/null
+++ b/media/libaudiohal/4.0/StreamPowerLog.h

@@ -0,0 +1,104 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_STREAM_POWER_LOG_4_0_H
+#define ANDROID_HARDWARE_STREAM_POWER_LOG_4_0_H
+
+#include <audio_utils/clock.h>
+#include <audio_utils/PowerLog.h>
+#include <cutils/properties.h>
+#include <system/audio.h>
+
+namespace android {
+namespace V4_0 {
+
+class StreamPowerLog {
+public:
+    StreamPowerLog() :
+        mIsUserDebugOrEngBuild(is_userdebug_or_eng_build()),
+        mPowerLog(nullptr),
+        mFrameSize(0) {
+        // use init() to set up the power log.
+    }
+
+    ~StreamPowerLog() {
+        power_log_destroy(mPowerLog); // OK for null mPowerLog
+        mPowerLog = nullptr;
+    }
+
+    // A one-time initialization (do not call twice) before using StreamPowerLog.
+    void init(uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format) {
+        if (mPowerLog == nullptr) {
+            // Note: A way to get channel count for both input and output channel masks
+            // but does not check validity of the channel mask.
+            const uint32_t channelCount = popcount(audio_channel_mask_get_bits(channelMask));
+            mFrameSize = channelCount * audio_bytes_per_sample(format);
+            if (mFrameSize > 0) {
+                const size_t kPowerLogFramesPerEntry =
+                        (long long)sampleRate * kPowerLogSamplingIntervalMs / 1000;
+                mPowerLog = power_log_create(
+                        sampleRate,
+                        channelCount,
+                        format,
+                        kPowerLogEntries,
+                        kPowerLogFramesPerEntry);
+            }
+        }
+        // mPowerLog may be NULL (not the right build, format not accepted, etc.).
+    }
+
+    // Dump the power log to fd.
+    void dump(int fd) const {
+        // OK for null mPowerLog
+        (void)power_log_dump(
+                mPowerLog, fd, "      " /* prefix */, kPowerLogLines, 0 /* limit_ns */);
+    }
+
+    // Log the audio data contained in buffer.
+    void log(const void *buffer, size_t sizeInBytes) const {
+        if (mPowerLog != nullptr) { // mFrameSize is always nonzero if mPowerLog exists.
+            power_log_log(
+                    mPowerLog, buffer, sizeInBytes / mFrameSize, audio_utils_get_real_time_ns());
+        }
+    }
+
+    bool isUserDebugOrEngBuild() const {
+        return mIsUserDebugOrEngBuild;
+    }
+
+private:
+
+    static inline bool is_userdebug_or_eng_build() {
+        char value[PROPERTY_VALUE_MAX];
+        (void)property_get("ro.build.type", value, "unknown"); // ignore actual length
+        return strcmp(value, "userdebug") == 0 || strcmp(value, "eng") == 0;
+    }
+
+    // Audio signal power log configuration.
+    static const size_t kPowerLogLines = 40;
+    static const size_t kPowerLogSamplingIntervalMs = 50;
+    static const size_t kPowerLogEntries = (1 /* minutes */ * 60 /* seconds */ * 1000 /* msec */
+            / kPowerLogSamplingIntervalMs);
+
+    const bool mIsUserDebugOrEngBuild;
+    power_log_t *mPowerLog;
+    size_t mFrameSize;
+};
+
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_STREAM_POWER_LOG_4_0_H

diff --git a/media/libaudiohal/4.0/VersionUtils.h b/media/libaudiohal/4.0/VersionUtils.h
new file mode 100644
index 0000000..1246c2e
--- /dev/null
+++ b/media/libaudiohal/4.0/VersionUtils.h

@@ -0,0 +1,49 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_VERSION_UTILS_4_0_H
+#define ANDROID_HARDWARE_VERSION_UTILS_4_0_H
+
+#include <android/hardware/audio/4.0/types.h>
+#include <hidl/HidlSupport.h>
+
+using ::android::hardware::audio::V4_0::ParameterValue;
+using ::android::hardware::audio::V4_0::Result;
+using ::android::hardware::Return;
+using ::android::hardware::hidl_vec;
+using ::android::hardware::hidl_string;
+
+namespace android {
+namespace V4_0 {
+namespace utils {
+
+template <class T, class Callback>
+Return<void> getParameters(T& object, hidl_vec<ParameterValue> context,
+                           hidl_vec<hidl_string> keys, Callback callback) {
+    return object->getParameters(context, keys, callback);
+}
+
+template <class T>
+Return<Result> setParameters(T& object, hidl_vec<ParameterValue> context,
+                             hidl_vec<ParameterValue> keys) {
+    return object->setParameters(context, keys);
+}
+
+} // namespace utils
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_VERSION_UTILS_4_0_H

diff --git a/media/libaudiohal/4.0/include/libaudiohal/4.0/DevicesFactoryHalHybrid.h b/media/libaudiohal/4.0/include/libaudiohal/4.0/DevicesFactoryHalHybrid.h
new file mode 100644
index 0000000..abf6de0
--- /dev/null
+++ b/media/libaudiohal/4.0/include/libaudiohal/4.0/DevicesFactoryHalHybrid.h

@@ -0,0 +1,49 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_4_0_H
+#define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_4_0_H
+
+#include <media/audiohal/DevicesFactoryHalInterface.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+namespace android {
+namespace V4_0 {
+
+class DevicesFactoryHalHybrid : public DevicesFactoryHalInterface
+{
+  public:
+    // Opens a device with the specified name. To close the device, it is
+    // necessary to release references to the returned object.
+    virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
+
+  private:
+    friend class DevicesFactoryHalInterface;
+
+    // Can not be constructed directly by clients.
+    DevicesFactoryHalHybrid();
+
+    virtual ~DevicesFactoryHalHybrid();
+
+    sp<DevicesFactoryHalInterface> mLocalFactory;
+    sp<DevicesFactoryHalInterface> mHidlFactory;
+};
+
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_4_0_H

diff --git a/media/libaudiohal/4.0/include/libaudiohal/4.0/EffectsFactoryHalHidl.h b/media/libaudiohal/4.0/include/libaudiohal/4.0/EffectsFactoryHalHidl.h
new file mode 100644
index 0000000..680b7a1
--- /dev/null
+++ b/media/libaudiohal/4.0/include/libaudiohal/4.0/EffectsFactoryHalHidl.h

@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_4_0_H
+#define ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_4_0_H
+
+#include <android/hardware/audio/effect/4.0/IEffectsFactory.h>
+#include <android/hardware/audio/effect/4.0/types.h>
+#include <media/audiohal/EffectsFactoryHalInterface.h>
+
+#include "ConversionHelperHidl.h"
+
+namespace android {
+namespace V4_0 {
+
+using ::android::hardware::audio::effect::V4_0::EffectDescriptor;
+using ::android::hardware::audio::effect::V4_0::IEffectsFactory;
+using ::android::hardware::hidl_vec;
+
+class EffectsFactoryHalHidl : public EffectsFactoryHalInterface, public ConversionHelperHidl
+{
+  public:
+    // Returns the number of different effects in all loaded libraries.
+    virtual status_t queryNumberEffects(uint32_t *pNumEffects);
+
+    // Returns a descriptor of the next available effect.
+    virtual status_t getDescriptor(uint32_t index,
+            effect_descriptor_t *pDescriptor);
+
+    virtual status_t getDescriptor(const effect_uuid_t *pEffectUuid,
+            effect_descriptor_t *pDescriptor);
+
+    // Creates an effect engine of the specified type.
+    // To release the effect engine, it is necessary to release references
+    // to the returned effect object.
+    virtual status_t createEffect(const effect_uuid_t *pEffectUuid,
+            int32_t sessionId, int32_t ioId,
+            sp<EffectHalInterface> *effect);
+
+    virtual status_t dumpEffects(int fd);
+
+    status_t allocateBuffer(size_t size, sp<EffectBufferHalInterface>* buffer) override;
+    status_t mirrorBuffer(void* external, size_t size,
+                          sp<EffectBufferHalInterface>* buffer) override;
+
+  private:
+    friend class EffectsFactoryHalInterface;
+
+    sp<IEffectsFactory> mEffectsFactory;
+    hidl_vec<EffectDescriptor> mLastDescriptors;
+
+    // Can not be constructed directly by clients.
+    EffectsFactoryHalHidl();
+    virtual ~EffectsFactoryHalHidl();
+
+    status_t queryAllDescriptors();
+};
+
+} // namespace V4_0
+} // namespace android
+
+#endif // ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_4_0_H

diff --git a/media/libaudiohal/Android.bp b/media/libaudiohal/Android.bp
new file mode 100644
index 0000000..3a5df27
--- /dev/null
+++ b/media/libaudiohal/Android.bp

@@ -0,0 +1,56 @@
+cc_library_shared {
+    name: "libaudiohal",
+
+    srcs: [
+        "DevicesFactoryHalInterface.cpp",
+        "EffectsFactoryHalInterface.cpp",
+    ],
+
+    cflags: [
+        "-Wall",
+        "-Werror",
+    ],
+
+    shared_libs: [
+        "android.hardware.audio.effect@2.0",
+        "android.hardware.audio.effect@4.0",
+        "android.hardware.audio@2.0",
+        "android.hardware.audio@4.0",
+        "libaudiohal@2.0",
+        "libaudiohal@4.0",
+        "libutils",
+    ],
+
+    header_libs: [
+        "libaudiohal_headers"
+    ]
+}
+
+cc_library_shared {
+    name: "libaudiohal_deathhandler",
+
+    srcs: [
+        "HalDeathHandlerHidl.cpp",
+    ],
+
+    cflags: [
+        "-Wall",
+        "-Werror",
+    ],
+
+    shared_libs: [
+        "libhidlbase",
+        "libutils",
+        "liblog",
+    ],
+
+    header_libs: [
+        "libaudiohal_headers"
+    ]
+}
+
+cc_library_headers {
+    name: "libaudiohal_headers",
+
+    export_include_dirs: ["include"],
+}

diff --git a/media/libaudiohal/Android.mk b/media/libaudiohal/Android.mk
deleted file mode 100644
index 827908e..0000000
--- a/media/libaudiohal/Android.mk
+++ /dev/null

@@ -1,71 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_SHARED_LIBRARIES := \
-    libaudioutils \
-    libcutils   \
-    liblog      \
-    libutils    \
-    libhardware
-
-LOCAL_SRC_FILES := \
-    DeviceHalLocal.cpp          \
-    DevicesFactoryHalHybrid.cpp \
-    DevicesFactoryHalLocal.cpp  \
-    StreamHalLocal.cpp
-
-LOCAL_CFLAGS := -Wall -Werror
-
-ifeq ($(USE_LEGACY_LOCAL_AUDIO_HAL), true)
-
-# Use audiohal directly w/o hwbinder middleware.
-# This is for performance comparison and debugging only.
-
-LOCAL_SRC_FILES += \
-    EffectBufferHalLocal.cpp    \
-    EffectsFactoryHalLocal.cpp  \
-    EffectHalLocal.cpp
-
-LOCAL_SHARED_LIBRARIES += \
-    libeffects
-
-LOCAL_CFLAGS += -DUSE_LEGACY_LOCAL_AUDIO_HAL
-
-else  # if !USE_LEGACY_LOCAL_AUDIO_HAL
-
-LOCAL_SRC_FILES += \
-    ConversionHelperHidl.cpp   \
-    HalDeathHandlerHidl.cpp    \
-    DeviceHalHidl.cpp          \
-    DevicesFactoryHalHidl.cpp  \
-    EffectBufferHalHidl.cpp    \
-    EffectHalHidl.cpp          \
-    EffectsFactoryHalHidl.cpp  \
-    StreamHalHidl.cpp
-
-LOCAL_SHARED_LIBRARIES += \
-    libbase          \
-    libfmq           \
-    libhwbinder      \
-    libhidlbase      \
-    libhidlmemory    \
-    libhidltransport \
-    android.hardware.audio@2.0             \
-    android.hardware.audio.common@2.0      \
-    android.hardware.audio.common@2.0-util \
-    android.hardware.audio.effect@2.0      \
-    android.hidl.allocator@1.0             \
-    android.hidl.memory@1.0                \
-    libmedia_helper  \
-    libmediautils
-
-endif  # USE_LEGACY_LOCAL_AUDIO_HAL
-
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/include
-
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/include
-
-LOCAL_MODULE := libaudiohal
-
-include $(BUILD_SHARED_LIBRARY)

diff --git a/media/libaudiohal/DeviceHalHidl.cpp b/media/libaudiohal/DeviceHalHidl.cpp
deleted file mode 100644
index 49ef991..0000000
--- a/media/libaudiohal/DeviceHalHidl.cpp
+++ /dev/null

@@ -1,357 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <stdio.h>
-
-#define LOG_TAG "DeviceHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <android/hardware/audio/2.0/IPrimaryDevice.h>
-#include <cutils/native_handle.h>
-#include <hwbinder/IPCThreadState.h>
-#include <utils/Log.h>
-
-#include "DeviceHalHidl.h"
-#include "HidlUtils.h"
-#include "StreamHalHidl.h"
-
-using ::android::hardware::audio::common::V2_0::AudioConfig;
-using ::android::hardware::audio::common::V2_0::AudioDevice;
-using ::android::hardware::audio::common::V2_0::AudioInputFlag;
-using ::android::hardware::audio::common::V2_0::AudioOutputFlag;
-using ::android::hardware::audio::common::V2_0::AudioPatchHandle;
-using ::android::hardware::audio::common::V2_0::AudioPort;
-using ::android::hardware::audio::common::V2_0::AudioPortConfig;
-using ::android::hardware::audio::common::V2_0::AudioMode;
-using ::android::hardware::audio::common::V2_0::AudioSource;
-using ::android::hardware::audio::V2_0::DeviceAddress;
-using ::android::hardware::audio::V2_0::IPrimaryDevice;
-using ::android::hardware::audio::V2_0::ParameterValue;
-using ::android::hardware::audio::V2_0::Result;
-using ::android::hardware::hidl_string;
-using ::android::hardware::hidl_vec;
-
-namespace android {
-
-namespace {
-
-status_t deviceAddressFromHal(
-        audio_devices_t device, const char* halAddress, DeviceAddress* address) {
-    address->device = AudioDevice(device);
-
-    if (address == nullptr || strnlen(halAddress, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) {
-        return OK;
-    }
-    const bool isInput = (device & AUDIO_DEVICE_BIT_IN) != 0;
-    if (isInput) device &= ~AUDIO_DEVICE_BIT_IN;
-    if ((!isInput && (device & AUDIO_DEVICE_OUT_ALL_A2DP) != 0)
-            || (isInput && (device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) != 0)) {
-        int status = sscanf(halAddress,
-                "%hhX:%hhX:%hhX:%hhX:%hhX:%hhX",
-                &address->address.mac[0], &address->address.mac[1], &address->address.mac[2],
-                &address->address.mac[3], &address->address.mac[4], &address->address.mac[5]);
-        return status == 6 ? OK : BAD_VALUE;
-    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_IP) != 0)
-            || (isInput && (device & AUDIO_DEVICE_IN_IP) != 0)) {
-        int status = sscanf(halAddress,
-                "%hhu.%hhu.%hhu.%hhu",
-                &address->address.ipv4[0], &address->address.ipv4[1],
-                &address->address.ipv4[2], &address->address.ipv4[3]);
-        return status == 4 ? OK : BAD_VALUE;
-    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_ALL_USB)) != 0
-            || (isInput && (device & AUDIO_DEVICE_IN_ALL_USB)) != 0) {
-        int status = sscanf(halAddress,
-                "card=%d;device=%d",
-                &address->address.alsa.card, &address->address.alsa.device);
-        return status == 2 ? OK : BAD_VALUE;
-    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_BUS) != 0)
-            || (isInput && (device & AUDIO_DEVICE_IN_BUS) != 0)) {
-        if (halAddress != NULL) {
-            address->busAddress = halAddress;
-            return OK;
-        }
-        return BAD_VALUE;
-    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) != 0
-            || (isInput && (device & AUDIO_DEVICE_IN_REMOTE_SUBMIX) != 0)) {
-        if (halAddress != NULL) {
-            address->rSubmixAddress = halAddress;
-            return OK;
-        }
-        return BAD_VALUE;
-    }
-    return OK;
-}
-
-}  // namespace
-
-DeviceHalHidl::DeviceHalHidl(const sp<IDevice>& device)
-        : ConversionHelperHidl("Device"), mDevice(device),
-          mPrimaryDevice(IPrimaryDevice::castFrom(device)) {
-}
-
-DeviceHalHidl::~DeviceHalHidl() {
-    if (mDevice != 0) {
-        mDevice.clear();
-        hardware::IPCThreadState::self()->flushCommands();
-    }
-}
-
-status_t DeviceHalHidl::getSupportedDevices(uint32_t*) {
-    // Obsolete.
-    return INVALID_OPERATION;
-}
-
-status_t DeviceHalHidl::initCheck() {
-    if (mDevice == 0) return NO_INIT;
-    return processReturn("initCheck", mDevice->initCheck());
-}
-
-status_t DeviceHalHidl::setVoiceVolume(float volume) {
-    if (mDevice == 0) return NO_INIT;
-    if (mPrimaryDevice == 0) return INVALID_OPERATION;
-    return processReturn("setVoiceVolume", mPrimaryDevice->setVoiceVolume(volume));
-}
-
-status_t DeviceHalHidl::setMasterVolume(float volume) {
-    if (mDevice == 0) return NO_INIT;
-    if (mPrimaryDevice == 0) return INVALID_OPERATION;
-    return processReturn("setMasterVolume", mPrimaryDevice->setMasterVolume(volume));
-}
-
-status_t DeviceHalHidl::getMasterVolume(float *volume) {
-    if (mDevice == 0) return NO_INIT;
-    if (mPrimaryDevice == 0) return INVALID_OPERATION;
-    Result retval;
-    Return<void> ret = mPrimaryDevice->getMasterVolume(
-            [&](Result r, float v) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *volume = v;
-                }
-            });
-    return processReturn("getMasterVolume", ret, retval);
-}
-
-status_t DeviceHalHidl::setMode(audio_mode_t mode) {
-    if (mDevice == 0) return NO_INIT;
-    if (mPrimaryDevice == 0) return INVALID_OPERATION;
-    return processReturn("setMode", mPrimaryDevice->setMode(AudioMode(mode)));
-}
-
-status_t DeviceHalHidl::setMicMute(bool state) {
-    if (mDevice == 0) return NO_INIT;
-    return processReturn("setMicMute", mDevice->setMicMute(state));
-}
-
-status_t DeviceHalHidl::getMicMute(bool *state) {
-    if (mDevice == 0) return NO_INIT;
-    Result retval;
-    Return<void> ret = mDevice->getMicMute(
-            [&](Result r, bool mute) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *state = mute;
-                }
-            });
-    return processReturn("getMicMute", ret, retval);
-}
-
-status_t DeviceHalHidl::setMasterMute(bool state) {
-    if (mDevice == 0) return NO_INIT;
-    return processReturn("setMasterMute", mDevice->setMasterMute(state));
-}
-
-status_t DeviceHalHidl::getMasterMute(bool *state) {
-    if (mDevice == 0) return NO_INIT;
-    Result retval;
-    Return<void> ret = mDevice->getMasterMute(
-            [&](Result r, bool mute) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *state = mute;
-                }
-            });
-    return processReturn("getMasterMute", ret, retval);
-}
-
-status_t DeviceHalHidl::setParameters(const String8& kvPairs) {
-    if (mDevice == 0) return NO_INIT;
-    hidl_vec<ParameterValue> hidlParams;
-    status_t status = parametersFromHal(kvPairs, &hidlParams);
-    if (status != OK) return status;
-    return processReturn("setParameters", mDevice->setParameters(hidlParams));
-}
-
-status_t DeviceHalHidl::getParameters(const String8& keys, String8 *values) {
-    values->clear();
-    if (mDevice == 0) return NO_INIT;
-    hidl_vec<hidl_string> hidlKeys;
-    status_t status = keysFromHal(keys, &hidlKeys);
-    if (status != OK) return status;
-    Result retval;
-    Return<void> ret = mDevice->getParameters(
-            hidlKeys,
-            [&](Result r, const hidl_vec<ParameterValue>& parameters) {
-                retval = r;
-                if (retval == Result::OK) {
-                    parametersToHal(parameters, values);
-                }
-            });
-    return processReturn("getParameters", ret, retval);
-}
-
-status_t DeviceHalHidl::getInputBufferSize(
-        const struct audio_config *config, size_t *size) {
-    if (mDevice == 0) return NO_INIT;
-    AudioConfig hidlConfig;
-    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
-    Result retval;
-    Return<void> ret = mDevice->getInputBufferSize(
-            hidlConfig,
-            [&](Result r, uint64_t bufferSize) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *size = static_cast<size_t>(bufferSize);
-                }
-            });
-    return processReturn("getInputBufferSize", ret, retval);
-}
-
-status_t DeviceHalHidl::openOutputStream(
-        audio_io_handle_t handle,
-        audio_devices_t devices,
-        audio_output_flags_t flags,
-        struct audio_config *config,
-        const char *address,
-        sp<StreamOutHalInterface> *outStream) {
-    if (mDevice == 0) return NO_INIT;
-    DeviceAddress hidlDevice;
-    status_t status = deviceAddressFromHal(devices, address, &hidlDevice);
-    if (status != OK) return status;
-    AudioConfig hidlConfig;
-    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mDevice->openOutputStream(
-            handle,
-            hidlDevice,
-            hidlConfig,
-            AudioOutputFlag(flags),
-            [&](Result r, const sp<IStreamOut>& result, const AudioConfig& suggestedConfig) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *outStream = new StreamOutHalHidl(result);
-                }
-                HidlUtils::audioConfigToHal(suggestedConfig, config);
-            });
-    return processReturn("openOutputStream", ret, retval);
-}
-
-status_t DeviceHalHidl::openInputStream(
-        audio_io_handle_t handle,
-        audio_devices_t devices,
-        struct audio_config *config,
-        audio_input_flags_t flags,
-        const char *address,
-        audio_source_t source,
-        sp<StreamInHalInterface> *inStream) {
-    if (mDevice == 0) return NO_INIT;
-    DeviceAddress hidlDevice;
-    status_t status = deviceAddressFromHal(devices, address, &hidlDevice);
-    if (status != OK) return status;
-    AudioConfig hidlConfig;
-    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mDevice->openInputStream(
-            handle,
-            hidlDevice,
-            hidlConfig,
-            AudioInputFlag(flags),
-            AudioSource(source),
-            [&](Result r, const sp<IStreamIn>& result, const AudioConfig& suggestedConfig) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *inStream = new StreamInHalHidl(result);
-                }
-                HidlUtils::audioConfigToHal(suggestedConfig, config);
-            });
-    return processReturn("openInputStream", ret, retval);
-}
-
-status_t DeviceHalHidl::supportsAudioPatches(bool *supportsPatches) {
-    if (mDevice == 0) return NO_INIT;
-    return processReturn("supportsAudioPatches", mDevice->supportsAudioPatches(), supportsPatches);
-}
-
-status_t DeviceHalHidl::createAudioPatch(
-        unsigned int num_sources,
-        const struct audio_port_config *sources,
-        unsigned int num_sinks,
-        const struct audio_port_config *sinks,
-        audio_patch_handle_t *patch) {
-    if (mDevice == 0) return NO_INIT;
-    hidl_vec<AudioPortConfig> hidlSources, hidlSinks;
-    HidlUtils::audioPortConfigsFromHal(num_sources, sources, &hidlSources);
-    HidlUtils::audioPortConfigsFromHal(num_sinks, sinks, &hidlSinks);
-    Result retval;
-    Return<void> ret = mDevice->createAudioPatch(
-            hidlSources, hidlSinks,
-            [&](Result r, AudioPatchHandle hidlPatch) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *patch = static_cast<audio_patch_handle_t>(hidlPatch);
-                }
-            });
-    return processReturn("createAudioPatch", ret, retval);
-}
-
-status_t DeviceHalHidl::releaseAudioPatch(audio_patch_handle_t patch) {
-    if (mDevice == 0) return NO_INIT;
-    return processReturn("releaseAudioPatch", mDevice->releaseAudioPatch(patch));
-}
-
-status_t DeviceHalHidl::getAudioPort(struct audio_port *port) {
-    if (mDevice == 0) return NO_INIT;
-    AudioPort hidlPort;
-    HidlUtils::audioPortFromHal(*port, &hidlPort);
-    Result retval;
-    Return<void> ret = mDevice->getAudioPort(
-            hidlPort,
-            [&](Result r, const AudioPort& p) {
-                retval = r;
-                if (retval == Result::OK) {
-                    HidlUtils::audioPortToHal(p, port);
-                }
-            });
-    return processReturn("getAudioPort", ret, retval);
-}
-
-status_t DeviceHalHidl::setAudioPortConfig(const struct audio_port_config *config) {
-    if (mDevice == 0) return NO_INIT;
-    AudioPortConfig hidlConfig;
-    HidlUtils::audioPortConfigFromHal(*config, &hidlConfig);
-    return processReturn("setAudioPortConfig", mDevice->setAudioPortConfig(hidlConfig));
-}
-
-status_t DeviceHalHidl::dump(int fd) {
-    if (mDevice == 0) return NO_INIT;
-    native_handle_t* hidlHandle = native_handle_create(1, 0);
-    hidlHandle->data[0] = fd;
-    Return<void> ret = mDevice->debugDump(hidlHandle);
-    native_handle_delete(hidlHandle);
-    return processReturn("dump", ret);
-}
-
-} // namespace android

diff --git a/media/libaudiohal/DeviceHalHidl.h b/media/libaudiohal/DeviceHalHidl.h
deleted file mode 100644
index 8651b51..0000000
--- a/media/libaudiohal/DeviceHalHidl.h
+++ /dev/null

@@ -1,126 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_DEVICE_HAL_HIDL_H
-#define ANDROID_HARDWARE_DEVICE_HAL_HIDL_H
-
-#include <android/hardware/audio/2.0/IDevice.h>
-#include <android/hardware/audio/2.0/IPrimaryDevice.h>
-#include <media/audiohal/DeviceHalInterface.h>
-
-#include "ConversionHelperHidl.h"
-
-using ::android::hardware::audio::V2_0::IDevice;
-using ::android::hardware::audio::V2_0::IPrimaryDevice;
-using ::android::hardware::Return;
-
-namespace android {
-
-class DeviceHalHidl : public DeviceHalInterface, public ConversionHelperHidl
-{
-  public:
-    // Sets the value of 'devices' to a bitmask of 1 or more values of audio_devices_t.
-    virtual status_t getSupportedDevices(uint32_t *devices);
-
-    // Check to see if the audio hardware interface has been initialized.
-    virtual status_t initCheck();
-
-    // Set the audio volume of a voice call. Range is between 0.0 and 1.0.
-    virtual status_t setVoiceVolume(float volume);
-
-    // Set the audio volume for all audio activities other than voice call.
-    virtual status_t setMasterVolume(float volume);
-
-    // Get the current master volume value for the HAL.
-    virtual status_t getMasterVolume(float *volume);
-
-    // Called when the audio mode changes.
-    virtual status_t setMode(audio_mode_t mode);
-
-    // Muting control.
-    virtual status_t setMicMute(bool state);
-    virtual status_t getMicMute(bool *state);
-    virtual status_t setMasterMute(bool state);
-    virtual status_t getMasterMute(bool *state);
-
-    // Set global audio parameters.
-    virtual status_t setParameters(const String8& kvPairs);
-
-    // Get global audio parameters.
-    virtual status_t getParameters(const String8& keys, String8 *values);
-
-    // Returns audio input buffer size according to parameters passed.
-    virtual status_t getInputBufferSize(const struct audio_config *config,
-            size_t *size);
-
-    // Creates and opens the audio hardware output stream. The stream is closed
-    // by releasing all references to the returned object.
-    virtual status_t openOutputStream(
-            audio_io_handle_t handle,
-            audio_devices_t devices,
-            audio_output_flags_t flags,
-            struct audio_config *config,
-            const char *address,
-            sp<StreamOutHalInterface> *outStream);
-
-    // Creates and opens the audio hardware input stream. The stream is closed
-    // by releasing all references to the returned object.
-    virtual status_t openInputStream(
-            audio_io_handle_t handle,
-            audio_devices_t devices,
-            struct audio_config *config,
-            audio_input_flags_t flags,
-            const char *address,
-            audio_source_t source,
-            sp<StreamInHalInterface> *inStream);
-
-    // Returns whether createAudioPatch and releaseAudioPatch operations are supported.
-    virtual status_t supportsAudioPatches(bool *supportsPatches);
-
-    // Creates an audio patch between several source and sink ports.
-    virtual status_t createAudioPatch(
-            unsigned int num_sources,
-            const struct audio_port_config *sources,
-            unsigned int num_sinks,
-            const struct audio_port_config *sinks,
-            audio_patch_handle_t *patch);
-
-    // Releases an audio patch.
-    virtual status_t releaseAudioPatch(audio_patch_handle_t patch);
-
-    // Fills the list of supported attributes for a given audio port.
-    virtual status_t getAudioPort(struct audio_port *port);
-
-    // Set audio port configuration.
-    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
-
-    virtual status_t dump(int fd);
-
-  private:
-    friend class DevicesFactoryHalHidl;
-    sp<IDevice> mDevice;
-    sp<IPrimaryDevice> mPrimaryDevice;  // Null if it's not a primary device.
-
-    // Can not be constructed directly by clients.
-    explicit DeviceHalHidl(const sp<IDevice>& device);
-
-    // The destructor automatically closes the device.
-    virtual ~DeviceHalHidl();
-};
-
-} // namespace android
-
-#endif // ANDROID_HARDWARE_DEVICE_HAL_HIDL_H

diff --git a/media/libaudiohal/DeviceHalLocal.cpp b/media/libaudiohal/DeviceHalLocal.cpp
deleted file mode 100644
index fc098f5..0000000
--- a/media/libaudiohal/DeviceHalLocal.cpp
+++ /dev/null

@@ -1,199 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "DeviceHalLocal"
-//#define LOG_NDEBUG 0
-
-#include <utils/Log.h>
-
-#include "DeviceHalLocal.h"
-#include "StreamHalLocal.h"
-
-namespace android {
-
-DeviceHalLocal::DeviceHalLocal(audio_hw_device_t *dev)
-        : mDev(dev) {
-}
-
-DeviceHalLocal::~DeviceHalLocal() {
-    int status = audio_hw_device_close(mDev);
-    ALOGW_IF(status, "Error closing audio hw device %p: %s", mDev, strerror(-status));
-    mDev = 0;
-}
-
-status_t DeviceHalLocal::getSupportedDevices(uint32_t *devices) {
-    if (mDev->get_supported_devices == NULL) return INVALID_OPERATION;
-    *devices = mDev->get_supported_devices(mDev);
-    return OK;
-}
-
-status_t DeviceHalLocal::initCheck() {
-    return mDev->init_check(mDev);
-}
-
-status_t DeviceHalLocal::setVoiceVolume(float volume) {
-    return mDev->set_voice_volume(mDev, volume);
-}
-
-status_t DeviceHalLocal::setMasterVolume(float volume) {
-    if (mDev->set_master_volume == NULL) return INVALID_OPERATION;
-    return mDev->set_master_volume(mDev, volume);
-}
-
-status_t DeviceHalLocal::getMasterVolume(float *volume) {
-    if (mDev->get_master_volume == NULL) return INVALID_OPERATION;
-    return mDev->get_master_volume(mDev, volume);
-}
-
-status_t DeviceHalLocal::setMode(audio_mode_t mode) {
-    return mDev->set_mode(mDev, mode);
-}
-
-status_t DeviceHalLocal::setMicMute(bool state) {
-    return mDev->set_mic_mute(mDev, state);
-}
-
-status_t DeviceHalLocal::getMicMute(bool *state) {
-    return mDev->get_mic_mute(mDev, state);
-}
-
-status_t DeviceHalLocal::setMasterMute(bool state) {
-    if (mDev->set_master_mute == NULL) return INVALID_OPERATION;
-    return mDev->set_master_mute(mDev, state);
-}
-
-status_t DeviceHalLocal::getMasterMute(bool *state) {
-    if (mDev->get_master_mute == NULL) return INVALID_OPERATION;
-    return mDev->get_master_mute(mDev, state);
-}
-
-status_t DeviceHalLocal::setParameters(const String8& kvPairs) {
-    return mDev->set_parameters(mDev, kvPairs.string());
-}
-
-status_t DeviceHalLocal::getParameters(const String8& keys, String8 *values) {
-    char *halValues = mDev->get_parameters(mDev, keys.string());
-    if (halValues != NULL) {
-        values->setTo(halValues);
-        free(halValues);
-    } else {
-        values->clear();
-    }
-    return OK;
-}
-
-status_t DeviceHalLocal::getInputBufferSize(
-        const struct audio_config *config, size_t *size) {
-    *size = mDev->get_input_buffer_size(mDev, config);
-    return OK;
-}
-
-status_t DeviceHalLocal::openOutputStream(
-        audio_io_handle_t handle,
-        audio_devices_t devices,
-        audio_output_flags_t flags,
-        struct audio_config *config,
-        const char *address,
-        sp<StreamOutHalInterface> *outStream) {
-    audio_stream_out_t *halStream;
-    ALOGV("open_output_stream handle: %d devices: %x flags: %#x"
-            "srate: %d format %#x channels %x address %s",
-            handle, devices, flags,
-            config->sample_rate, config->format, config->channel_mask,
-            address);
-    int openResut = mDev->open_output_stream(
-            mDev, handle, devices, flags, config, &halStream, address);
-    if (openResut == OK) {
-        *outStream = new StreamOutHalLocal(halStream, this);
-    }
-    ALOGV("open_output_stream status %d stream %p", openResut, halStream);
-    return openResut;
-}
-
-status_t DeviceHalLocal::openInputStream(
-        audio_io_handle_t handle,
-        audio_devices_t devices,
-        struct audio_config *config,
-        audio_input_flags_t flags,
-        const char *address,
-        audio_source_t source,
-        sp<StreamInHalInterface> *inStream) {
-    audio_stream_in_t *halStream;
-    ALOGV("open_input_stream handle: %d devices: %x flags: %#x "
-            "srate: %d format %#x channels %x address %s source %d",
-            handle, devices, flags,
-            config->sample_rate, config->format, config->channel_mask,
-            address, source);
-    int openResult = mDev->open_input_stream(
-            mDev, handle, devices, config, &halStream, flags, address, source);
-    if (openResult == OK) {
-        *inStream = new StreamInHalLocal(halStream, this);
-    }
-    ALOGV("open_input_stream status %d stream %p", openResult, inStream);
-    return openResult;
-}
-
-status_t DeviceHalLocal::supportsAudioPatches(bool *supportsPatches) {
-    *supportsPatches = version() >= AUDIO_DEVICE_API_VERSION_3_0;
-    return OK;
-}
-
-status_t DeviceHalLocal::createAudioPatch(
-        unsigned int num_sources,
-        const struct audio_port_config *sources,
-        unsigned int num_sinks,
-        const struct audio_port_config *sinks,
-        audio_patch_handle_t *patch) {
-    if (version() >= AUDIO_DEVICE_API_VERSION_3_0) {
-        return mDev->create_audio_patch(
-                mDev, num_sources, sources, num_sinks, sinks, patch);
-    } else {
-        return INVALID_OPERATION;
-    }
-}
-
-status_t DeviceHalLocal::releaseAudioPatch(audio_patch_handle_t patch) {
-    if (version() >= AUDIO_DEVICE_API_VERSION_3_0) {
-        return mDev->release_audio_patch(mDev, patch);
-    } else {
-        return INVALID_OPERATION;
-    }
-}
-
-status_t DeviceHalLocal::getAudioPort(struct audio_port *port) {
-    return mDev->get_audio_port(mDev, port);
-}
-
-status_t DeviceHalLocal::setAudioPortConfig(const struct audio_port_config *config) {
-    if (version() >= AUDIO_DEVICE_API_VERSION_3_0)
-        return mDev->set_audio_port_config(mDev, config);
-    else
-        return INVALID_OPERATION;
-}
-
-status_t DeviceHalLocal::dump(int fd) {
-    return mDev->dump(mDev, fd);
-}
-
-void DeviceHalLocal::closeOutputStream(struct audio_stream_out *stream_out) {
-    mDev->close_output_stream(mDev, stream_out);
-}
-
-void DeviceHalLocal::closeInputStream(struct audio_stream_in *stream_in) {
-    mDev->close_input_stream(mDev, stream_in);
-}
-
-} // namespace android

diff --git a/media/libaudiohal/DeviceHalLocal.h b/media/libaudiohal/DeviceHalLocal.h
deleted file mode 100644
index 865f296..0000000
--- a/media/libaudiohal/DeviceHalLocal.h
+++ /dev/null

@@ -1,124 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_DEVICE_HAL_LOCAL_H
-#define ANDROID_HARDWARE_DEVICE_HAL_LOCAL_H
-
-#include <hardware/audio.h>
-#include <media/audiohal/DeviceHalInterface.h>
-
-namespace android {
-
-class DeviceHalLocal : public DeviceHalInterface
-{
-  public:
-    // Sets the value of 'devices' to a bitmask of 1 or more values of audio_devices_t.
-    virtual status_t getSupportedDevices(uint32_t *devices);
-
-    // Check to see if the audio hardware interface has been initialized.
-    virtual status_t initCheck();
-
-    // Set the audio volume of a voice call. Range is between 0.0 and 1.0.
-    virtual status_t setVoiceVolume(float volume);
-
-    // Set the audio volume for all audio activities other than voice call.
-    virtual status_t setMasterVolume(float volume);
-
-    // Get the current master volume value for the HAL.
-    virtual status_t getMasterVolume(float *volume);
-
-    // Called when the audio mode changes.
-    virtual status_t setMode(audio_mode_t mode);
-
-    // Muting control.
-    virtual status_t setMicMute(bool state);
-    virtual status_t getMicMute(bool *state);
-    virtual status_t setMasterMute(bool state);
-    virtual status_t getMasterMute(bool *state);
-
-    // Set global audio parameters.
-    virtual status_t setParameters(const String8& kvPairs);
-
-    // Get global audio parameters.
-    virtual status_t getParameters(const String8& keys, String8 *values);
-
-    // Returns audio input buffer size according to parameters passed.
-    virtual status_t getInputBufferSize(const struct audio_config *config,
-            size_t *size);
-
-    // Creates and opens the audio hardware output stream. The stream is closed
-    // by releasing all references to the returned object.
-    virtual status_t openOutputStream(
-            audio_io_handle_t handle,
-            audio_devices_t devices,
-            audio_output_flags_t flags,
-            struct audio_config *config,
-            const char *address,
-            sp<StreamOutHalInterface> *outStream);
-
-    // Creates and opens the audio hardware input stream. The stream is closed
-    // by releasing all references to the returned object.
-    virtual status_t openInputStream(
-            audio_io_handle_t handle,
-            audio_devices_t devices,
-            struct audio_config *config,
-            audio_input_flags_t flags,
-            const char *address,
-            audio_source_t source,
-            sp<StreamInHalInterface> *inStream);
-
-    // Returns whether createAudioPatch and releaseAudioPatch operations are supported.
-    virtual status_t supportsAudioPatches(bool *supportsPatches);
-
-    // Creates an audio patch between several source and sink ports.
-    virtual status_t createAudioPatch(
-            unsigned int num_sources,
-            const struct audio_port_config *sources,
-            unsigned int num_sinks,
-            const struct audio_port_config *sinks,
-            audio_patch_handle_t *patch);
-
-    // Releases an audio patch.
-    virtual status_t releaseAudioPatch(audio_patch_handle_t patch);
-
-    // Fills the list of supported attributes for a given audio port.
-    virtual status_t getAudioPort(struct audio_port *port);
-
-    // Set audio port configuration.
-    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
-
-    virtual status_t dump(int fd);
-
-    void closeOutputStream(struct audio_stream_out *stream_out);
-    void closeInputStream(struct audio_stream_in *stream_in);
-
-  private:
-    audio_hw_device_t *mDev;
-
-    friend class DevicesFactoryHalLocal;
-
-    // Can not be constructed directly by clients.
-    explicit DeviceHalLocal(audio_hw_device_t *dev);
-
-    // The destructor automatically closes the device.
-    virtual ~DeviceHalLocal();
-
-    uint32_t version() const { return mDev->common.version; }
-};
-
-} // namespace android
-
-#endif // ANDROID_HARDWARE_DEVICE_HAL_LOCAL_H

diff --git a/media/libaudiohal/DevicesFactoryHalHidl.cpp b/media/libaudiohal/DevicesFactoryHalHidl.cpp
deleted file mode 100644
index 31da263..0000000
--- a/media/libaudiohal/DevicesFactoryHalHidl.cpp
+++ /dev/null

@@ -1,95 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <string.h>
-
-#define LOG_TAG "DevicesFactoryHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <android/hardware/audio/2.0/IDevice.h>
-#include <media/audiohal/hidl/HalDeathHandler.h>
-#include <utils/Log.h>
-
-#include "ConversionHelperHidl.h"
-#include "DeviceHalHidl.h"
-#include "DevicesFactoryHalHidl.h"
-
-using ::android::hardware::audio::V2_0::IDevice;
-using ::android::hardware::audio::V2_0::Result;
-using ::android::hardware::Return;
-
-namespace android {
-
-DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
-    mDevicesFactory = IDevicesFactory::getService();
-    if (mDevicesFactory != 0) {
-        // It is assumed that DevicesFactory is owned by AudioFlinger
-        // and thus have the same lifespan.
-        mDevicesFactory->linkToDeath(HalDeathHandler::getInstance(), 0 /*cookie*/);
-    } else {
-        ALOGE("Failed to obtain IDevicesFactory service, terminating process.");
-        exit(1);
-    }
-}
-
-DevicesFactoryHalHidl::~DevicesFactoryHalHidl() {
-}
-
-// static
-status_t DevicesFactoryHalHidl::nameFromHal(const char *name, IDevicesFactory::Device *device) {
-    if (strcmp(name, AUDIO_HARDWARE_MODULE_ID_PRIMARY) == 0) {
-        *device = IDevicesFactory::Device::PRIMARY;
-        return OK;
-    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) {
-        *device = IDevicesFactory::Device::A2DP;
-        return OK;
-    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) {
-        *device = IDevicesFactory::Device::USB;
-        return OK;
-    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) {
-        *device = IDevicesFactory::Device::R_SUBMIX;
-        return OK;
-    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_STUB) == 0) {
-        *device = IDevicesFactory::Device::STUB;
-        return OK;
-    }
-    ALOGE("Invalid device name %s", name);
-    return BAD_VALUE;
-}
-
-status_t DevicesFactoryHalHidl::openDevice(const char *name, sp<DeviceHalInterface> *device) {
-    if (mDevicesFactory == 0) return NO_INIT;
-    IDevicesFactory::Device hidlDevice;
-    status_t status = nameFromHal(name, &hidlDevice);
-    if (status != OK) return status;
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mDevicesFactory->openDevice(
-            hidlDevice,
-            [&](Result r, const sp<IDevice>& result) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *device = new DeviceHalHidl(result);
-                }
-            });
-    if (ret.isOk()) {
-        if (retval == Result::OK) return OK;
-        else if (retval == Result::INVALID_ARGUMENTS) return BAD_VALUE;
-        else return NO_INIT;
-    }
-    return FAILED_TRANSACTION;
-}
-
-} // namespace android

diff --git a/media/libaudiohal/DevicesFactoryHalHidl.h b/media/libaudiohal/DevicesFactoryHalHidl.h
deleted file mode 100644
index e2f1ad1..0000000
--- a/media/libaudiohal/DevicesFactoryHalHidl.h
+++ /dev/null

@@ -1,53 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_H
-#define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_H
-
-#include <android/hardware/audio/2.0/IDevicesFactory.h>
-#include <media/audiohal/DevicesFactoryHalInterface.h>
-#include <utils/Errors.h>
-#include <utils/RefBase.h>
-
-#include "DeviceHalHidl.h"
-
-using ::android::hardware::audio::V2_0::IDevicesFactory;
-
-namespace android {
-
-class DevicesFactoryHalHidl : public DevicesFactoryHalInterface
-{
-  public:
-    // Opens a device with the specified name. To close the device, it is
-    // necessary to release references to the returned object.
-    virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
-
-  private:
-    friend class DevicesFactoryHalHybrid;
-
-    sp<IDevicesFactory> mDevicesFactory;
-
-    static status_t nameFromHal(const char *name, IDevicesFactory::Device *device);
-
-    // Can not be constructed directly by clients.
-    DevicesFactoryHalHidl();
-
-    virtual ~DevicesFactoryHalHidl();
-};
-
-} // namespace android
-
-#endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_H

diff --git a/media/libaudiohal/DevicesFactoryHalHybrid.cpp b/media/libaudiohal/DevicesFactoryHalHybrid.cpp
deleted file mode 100644
index 454b03b..0000000
--- a/media/libaudiohal/DevicesFactoryHalHybrid.cpp
+++ /dev/null

@@ -1,54 +0,0 @@
-/*
- * Copyright (C) 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "DevicesFactoryHalHybrid"
-//#define LOG_NDEBUG 0
-
-#include "DevicesFactoryHalHybrid.h"
-#include "DevicesFactoryHalLocal.h"
-#ifndef USE_LEGACY_LOCAL_AUDIO_HAL
-#include "DevicesFactoryHalHidl.h"
-#endif
-
-namespace android {
-
-// static
-sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
-    return new DevicesFactoryHalHybrid();
-}
-
-DevicesFactoryHalHybrid::DevicesFactoryHalHybrid()
-        : mLocalFactory(new DevicesFactoryHalLocal()),
-          mHidlFactory(
-#ifdef USE_LEGACY_LOCAL_AUDIO_HAL
-                  nullptr
-#else
-                  new DevicesFactoryHalHidl()
-#endif
-                       ) {
-}
-
-DevicesFactoryHalHybrid::~DevicesFactoryHalHybrid() {
-}
-
-status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
-    if (mHidlFactory != 0 && strcmp(AUDIO_HARDWARE_MODULE_ID_A2DP, name) != 0) {
-        return mHidlFactory->openDevice(name, device);
-    }
-    return mLocalFactory->openDevice(name, device);
-}
-
-} // namespace android

diff --git a/media/libaudiohal/DevicesFactoryHalInterface.cpp b/media/libaudiohal/DevicesFactoryHalInterface.cpp
new file mode 100644
index 0000000..4c8eaf6
--- /dev/null
+++ b/media/libaudiohal/DevicesFactoryHalInterface.cpp

@@ -0,0 +1,36 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <android/hardware/audio/2.0/IDevicesFactory.h>
+#include <android/hardware/audio/4.0/IDevicesFactory.h>
+
+#include <DevicesFactoryHalHybrid.h>
+#include <libaudiohal/4.0/DevicesFactoryHalHybrid.h>
+
+namespace android {
+
+// static
+sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
+    if (hardware::audio::V4_0::IDevicesFactory::getService() != nullptr) {
+        return new V4_0::DevicesFactoryHalHybrid();
+    }
+    if (hardware::audio::V2_0::IDevicesFactory::getService() != nullptr) {
+        return new DevicesFactoryHalHybrid();
+    }
+    return nullptr;
+}
+
+} // namespace android

diff --git a/media/libaudiohal/EffectBufferHalHidl.cpp b/media/libaudiohal/EffectBufferHalHidl.cpp
deleted file mode 100644
index 8b5201b..0000000
--- a/media/libaudiohal/EffectBufferHalHidl.cpp
+++ /dev/null

@@ -1,146 +0,0 @@
-/*
- * Copyright (C) 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <atomic>
-
-#define LOG_TAG "EffectBufferHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <android/hidl/allocator/1.0/IAllocator.h>
-#include <hidlmemory/mapping.h>
-#include <utils/Log.h>
-
-#include "ConversionHelperHidl.h"
-#include "EffectBufferHalHidl.h"
-
-using ::android::hardware::Return;
-using ::android::hidl::allocator::V1_0::IAllocator;
-
-namespace android {
-
-// static
-uint64_t EffectBufferHalHidl::makeUniqueId() {
-    static std::atomic<uint64_t> counter{1};
-    return counter++;
-}
-
-// static
-status_t EffectBufferHalInterface::allocate(
-        size_t size, sp<EffectBufferHalInterface>* buffer) {
-    return mirror(nullptr, size, buffer);
-}
-
-// static
-status_t EffectBufferHalInterface::mirror(
-        void* external, size_t size, sp<EffectBufferHalInterface>* buffer) {
-    sp<EffectBufferHalInterface> tempBuffer = new EffectBufferHalHidl(size);
-    status_t result = static_cast<EffectBufferHalHidl*>(tempBuffer.get())->init();
-    if (result == OK) {
-        tempBuffer->setExternalData(external);
-        *buffer = tempBuffer;
-    }
-    return result;
-}
-
-EffectBufferHalHidl::EffectBufferHalHidl(size_t size)
-        : mBufferSize(size), mFrameCountChanged(false),
-          mExternalData(nullptr), mAudioBuffer{0, {nullptr}} {
-    mHidlBuffer.id = makeUniqueId();
-    mHidlBuffer.frameCount = 0;
-}
-
-EffectBufferHalHidl::~EffectBufferHalHidl() {
-}
-
-status_t EffectBufferHalHidl::init() {
-    sp<IAllocator> ashmem = IAllocator::getService("ashmem");
-    if (ashmem == 0) {
-        ALOGE("Failed to retrieve ashmem allocator service");
-        return NO_INIT;
-    }
-    status_t retval = NO_MEMORY;
-    Return<void> result = ashmem->allocate(
-            mBufferSize,
-            [&](bool success, const hidl_memory& memory) {
-                if (success) {
-                    mHidlBuffer.data = memory;
-                    retval = OK;
-                }
-            });
-    if (result.isOk() && retval == OK) {
-        mMemory = hardware::mapMemory(mHidlBuffer.data);
-        if (mMemory != 0) {
-            mMemory->update();
-            mAudioBuffer.raw = static_cast<void*>(mMemory->getPointer());
-            memset(mAudioBuffer.raw, 0, mMemory->getSize());
-            mMemory->commit();
-        } else {
-            ALOGE("Failed to map allocated ashmem");
-            retval = NO_MEMORY;
-        }
-    } else {
-        ALOGE("Failed to allocate %d bytes from ashmem", (int)mBufferSize);
-    }
-    return result.isOk() ? retval : FAILED_TRANSACTION;
-}
-
-audio_buffer_t* EffectBufferHalHidl::audioBuffer() {
-    return &mAudioBuffer;
-}
-
-void* EffectBufferHalHidl::externalData() const {
-    return mExternalData;
-}
-
-void EffectBufferHalHidl::setFrameCount(size_t frameCount) {
-    mHidlBuffer.frameCount = frameCount;
-    mAudioBuffer.frameCount = frameCount;
-    mFrameCountChanged = true;
-}
-
-bool EffectBufferHalHidl::checkFrameCountChange() {
-    bool result = mFrameCountChanged;
-    mFrameCountChanged = false;
-    return result;
-}
-
-void EffectBufferHalHidl::setExternalData(void* external) {
-    mExternalData = external;
-}
-
-void EffectBufferHalHidl::update() {
-    update(mBufferSize);
-}
-
-void EffectBufferHalHidl::commit() {
-    commit(mBufferSize);
-}
-
-void EffectBufferHalHidl::update(size_t size) {
-    if (mExternalData == nullptr) return;
-    mMemory->update();
-    if (size > mBufferSize) size = mBufferSize;
-    memcpy(mAudioBuffer.raw, mExternalData, size);
-    mMemory->commit();
-}
-
-void EffectBufferHalHidl::commit(size_t size) {
-    if (mExternalData == nullptr) return;
-    if (size > mBufferSize) size = mBufferSize;
-    memcpy(mExternalData, mAudioBuffer.raw, size);
-}
-
-} // namespace android

diff --git a/media/libaudiohal/EffectBufferHalHidl.h b/media/libaudiohal/EffectBufferHalHidl.h
deleted file mode 100644
index 66a81c2..0000000
--- a/media/libaudiohal/EffectBufferHalHidl.h
+++ /dev/null

@@ -1,71 +0,0 @@
-/*
- * Copyright (C) 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_H
-#define ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_H
-
-#include <android/hardware/audio/effect/2.0/types.h>
-#include <android/hidl/memory/1.0/IMemory.h>
-#include <hidl/HidlSupport.h>
-#include <media/audiohal/EffectBufferHalInterface.h>
-#include <system/audio_effect.h>
-
-using android::hardware::audio::effect::V2_0::AudioBuffer;
-using android::hardware::hidl_memory;
-using android::hidl::memory::V1_0::IMemory;
-
-namespace android {
-
-class EffectBufferHalHidl : public EffectBufferHalInterface
-{
-  public:
-    virtual audio_buffer_t* audioBuffer();
-    virtual void* externalData() const;
-
-    virtual void setExternalData(void* external);
-    virtual void setFrameCount(size_t frameCount);
-    virtual bool checkFrameCountChange();
-
-    virtual void update();
-    virtual void commit();
-    virtual void update(size_t size);
-    virtual void commit(size_t size);
-
-    const AudioBuffer& hidlBuffer() const { return mHidlBuffer; }
-
-  private:
-    friend class EffectBufferHalInterface;
-
-    static uint64_t makeUniqueId();
-
-    const size_t mBufferSize;
-    bool mFrameCountChanged;
-    void* mExternalData;
-    AudioBuffer mHidlBuffer;
-    sp<IMemory> mMemory;
-    audio_buffer_t mAudioBuffer;
-
-    // Can not be constructed directly by clients.
-    explicit EffectBufferHalHidl(size_t size);
-
-    virtual ~EffectBufferHalHidl();
-
-    status_t init();
-};
-
-} // namespace android
-
-#endif // ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_H

diff --git a/media/libaudiohal/EffectBufferHalLocal.cpp b/media/libaudiohal/EffectBufferHalLocal.cpp
deleted file mode 100644
index 7951c8e..0000000
--- a/media/libaudiohal/EffectBufferHalLocal.cpp
+++ /dev/null

@@ -1,91 +0,0 @@
-/*
- * Copyright (C) 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "EffectBufferHalLocal"
-//#define LOG_NDEBUG 0
-
-#include <utils/Log.h>
-
-#include "EffectBufferHalLocal.h"
-
-namespace android {
-
-// static
-status_t EffectBufferHalInterface::allocate(
-        size_t size, sp<EffectBufferHalInterface>* buffer) {
-    *buffer = new EffectBufferHalLocal(size);
-    return OK;
-}
-
-// static
-status_t EffectBufferHalInterface::mirror(
-        void* external, size_t size, sp<EffectBufferHalInterface>* buffer) {
-    *buffer = new EffectBufferHalLocal(external, size);
-    return OK;
-}
-
-EffectBufferHalLocal::EffectBufferHalLocal(size_t size)
-        : mOwnBuffer(new uint8_t[size]),
-          mBufferSize(size), mFrameCountChanged(false),
-          mAudioBuffer{0, {mOwnBuffer.get()}} {
-}
-
-EffectBufferHalLocal::EffectBufferHalLocal(void* external, size_t size)
-        : mOwnBuffer(nullptr),
-          mBufferSize(size), mFrameCountChanged(false),
-          mAudioBuffer{0, {external}} {
-}
-
-EffectBufferHalLocal::~EffectBufferHalLocal() {
-}
-
-audio_buffer_t* EffectBufferHalLocal::audioBuffer() {
-    return &mAudioBuffer;
-}
-
-void* EffectBufferHalLocal::externalData() const {
-    return mAudioBuffer.raw;
-}
-
-void EffectBufferHalLocal::setFrameCount(size_t frameCount) {
-    mAudioBuffer.frameCount = frameCount;
-    mFrameCountChanged = true;
-}
-
-void EffectBufferHalLocal::setExternalData(void* external) {
-    ALOGE_IF(mOwnBuffer != nullptr, "Attempt to set external data for allocated buffer");
-    mAudioBuffer.raw = external;
-}
-
-bool EffectBufferHalLocal::checkFrameCountChange() {
-    bool result = mFrameCountChanged;
-    mFrameCountChanged = false;
-    return result;
-}
-
-void EffectBufferHalLocal::update() {
-}
-
-void EffectBufferHalLocal::commit() {
-}
-
-void EffectBufferHalLocal::update(size_t) {
-}
-
-void EffectBufferHalLocal::commit(size_t) {
-}
-
-} // namespace android

diff --git a/media/libaudiohal/EffectBufferHalLocal.h b/media/libaudiohal/EffectBufferHalLocal.h
deleted file mode 100644
index d2b624b..0000000
--- a/media/libaudiohal/EffectBufferHalLocal.h
+++ /dev/null

@@ -1,61 +0,0 @@
-/*
- * Copyright (C) 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_EFFECT_BUFFER_HAL_LOCAL_H
-#define ANDROID_HARDWARE_EFFECT_BUFFER_HAL_LOCAL_H
-
-#include <memory>
-
-#include <media/audiohal/EffectBufferHalInterface.h>
-#include <system/audio_effect.h>
-
-namespace android {
-
-class EffectBufferHalLocal : public EffectBufferHalInterface
-{
-  public:
-    virtual audio_buffer_t* audioBuffer();
-    virtual void* externalData() const;
-
-    virtual void setExternalData(void* external);
-    virtual void setFrameCount(size_t frameCount);
-    virtual bool checkFrameCountChange();
-
-    virtual void update();
-    virtual void commit();
-    virtual void update(size_t size);
-    virtual void commit(size_t size);
-
-  private:
-    friend class EffectBufferHalInterface;
-
-    std::unique_ptr<uint8_t[]> mOwnBuffer;
-    const size_t mBufferSize;
-    bool mFrameCountChanged;
-    audio_buffer_t mAudioBuffer;
-
-    // Can not be constructed directly by clients.
-    explicit EffectBufferHalLocal(size_t size);
-    EffectBufferHalLocal(void* external, size_t size);
-
-    virtual ~EffectBufferHalLocal();
-
-    status_t init();
-};
-
-} // namespace android
-
-#endif // ANDROID_HARDWARE_EFFECT_BUFFER_HAL_LOCAL_H

diff --git a/media/libaudiohal/EffectHalHidl.cpp b/media/libaudiohal/EffectHalHidl.cpp
deleted file mode 100644
index 61fb6bab..0000000
--- a/media/libaudiohal/EffectHalHidl.cpp
+++ /dev/null

@@ -1,329 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "EffectHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <hwbinder/IPCThreadState.h>
-#include <media/EffectsFactoryApi.h>
-#include <utils/Log.h>
-
-#include "ConversionHelperHidl.h"
-#include "EffectBufferHalHidl.h"
-#include "EffectHalHidl.h"
-#include "HidlUtils.h"
-
-using ::android::hardware::audio::effect::V2_0::AudioBuffer;
-using ::android::hardware::audio::effect::V2_0::EffectBufferAccess;
-using ::android::hardware::audio::effect::V2_0::EffectConfigParameters;
-using ::android::hardware::audio::effect::V2_0::MessageQueueFlagBits;
-using ::android::hardware::audio::effect::V2_0::Result;
-using ::android::hardware::audio::common::V2_0::AudioChannelMask;
-using ::android::hardware::audio::common::V2_0::AudioFormat;
-using ::android::hardware::hidl_vec;
-using ::android::hardware::MQDescriptorSync;
-using ::android::hardware::Return;
-
-namespace android {
-
-EffectHalHidl::EffectHalHidl(const sp<IEffect>& effect, uint64_t effectId)
-        : mEffect(effect), mEffectId(effectId), mBuffersChanged(true), mEfGroup(nullptr) {
-}
-
-EffectHalHidl::~EffectHalHidl() {
-    if (mEffect != 0) {
-        close();
-        mEffect.clear();
-        hardware::IPCThreadState::self()->flushCommands();
-    }
-    if (mEfGroup) {
-        EventFlag::deleteEventFlag(&mEfGroup);
-    }
-}
-
-// static
-void EffectHalHidl::effectDescriptorToHal(
-        const EffectDescriptor& descriptor, effect_descriptor_t* halDescriptor) {
-    HidlUtils::uuidToHal(descriptor.type, &halDescriptor->type);
-    HidlUtils::uuidToHal(descriptor.uuid, &halDescriptor->uuid);
-    halDescriptor->flags = static_cast<uint32_t>(descriptor.flags);
-    halDescriptor->cpuLoad = descriptor.cpuLoad;
-    halDescriptor->memoryUsage = descriptor.memoryUsage;
-    memcpy(halDescriptor->name, descriptor.name.data(), descriptor.name.size());
-    memcpy(halDescriptor->implementor,
-            descriptor.implementor.data(), descriptor.implementor.size());
-}
-
-// TODO(mnaganov): These buffer conversion functions should be shared with Effect wrapper
-// via HidlUtils. Move them there when hardware/interfaces will get un-frozen again.
-
-// static
-void EffectHalHidl::effectBufferConfigFromHal(
-        const buffer_config_t& halConfig, EffectBufferConfig* config) {
-    config->samplingRateHz = halConfig.samplingRate;
-    config->channels = AudioChannelMask(halConfig.channels);
-    config->format = AudioFormat(halConfig.format);
-    config->accessMode = EffectBufferAccess(halConfig.accessMode);
-    config->mask = EffectConfigParameters(halConfig.mask);
-}
-
-// static
-void EffectHalHidl::effectBufferConfigToHal(
-        const EffectBufferConfig& config, buffer_config_t* halConfig) {
-    halConfig->buffer.frameCount = 0;
-    halConfig->buffer.raw = NULL;
-    halConfig->samplingRate = config.samplingRateHz;
-    halConfig->channels = static_cast<uint32_t>(config.channels);
-    halConfig->bufferProvider.cookie = NULL;
-    halConfig->bufferProvider.getBuffer = NULL;
-    halConfig->bufferProvider.releaseBuffer = NULL;
-    halConfig->format = static_cast<uint8_t>(config.format);
-    halConfig->accessMode = static_cast<uint8_t>(config.accessMode);
-    halConfig->mask = static_cast<uint8_t>(config.mask);
-}
-
-// static
-void EffectHalHidl::effectConfigFromHal(const effect_config_t& halConfig, EffectConfig* config) {
-    effectBufferConfigFromHal(halConfig.inputCfg, &config->inputCfg);
-    effectBufferConfigFromHal(halConfig.outputCfg, &config->outputCfg);
-}
-
-// static
-void EffectHalHidl::effectConfigToHal(const EffectConfig& config, effect_config_t* halConfig) {
-    effectBufferConfigToHal(config.inputCfg, &halConfig->inputCfg);
-    effectBufferConfigToHal(config.outputCfg, &halConfig->outputCfg);
-}
-
-// static
-status_t EffectHalHidl::analyzeResult(const Result& result) {
-    switch (result) {
-        case Result::OK: return OK;
-        case Result::INVALID_ARGUMENTS: return BAD_VALUE;
-        case Result::INVALID_STATE: return NOT_ENOUGH_DATA;
-        case Result::NOT_INITIALIZED: return NO_INIT;
-        case Result::NOT_SUPPORTED: return INVALID_OPERATION;
-        case Result::RESULT_TOO_BIG: return NO_MEMORY;
-        default: return NO_INIT;
-    }
-}
-
-status_t EffectHalHidl::setInBuffer(const sp<EffectBufferHalInterface>& buffer) {
-    if (mInBuffer == 0 || buffer->audioBuffer() != mInBuffer->audioBuffer()) {
-        mBuffersChanged = true;
-    }
-    mInBuffer = buffer;
-    return OK;
-}
-
-status_t EffectHalHidl::setOutBuffer(const sp<EffectBufferHalInterface>& buffer) {
-    if (mOutBuffer == 0 || buffer->audioBuffer() != mOutBuffer->audioBuffer()) {
-        mBuffersChanged = true;
-    }
-    mOutBuffer = buffer;
-    return OK;
-}
-
-status_t EffectHalHidl::process() {
-    return processImpl(static_cast<uint32_t>(MessageQueueFlagBits::REQUEST_PROCESS));
-}
-
-status_t EffectHalHidl::processReverse() {
-    return processImpl(static_cast<uint32_t>(MessageQueueFlagBits::REQUEST_PROCESS_REVERSE));
-}
-
-status_t EffectHalHidl::prepareForProcessing() {
-    std::unique_ptr<StatusMQ> tempStatusMQ;
-    Result retval;
-    Return<void> ret = mEffect->prepareForProcessing(
-            [&](Result r, const MQDescriptorSync<Result>& statusMQ) {
-                retval = r;
-                if (retval == Result::OK) {
-                    tempStatusMQ.reset(new StatusMQ(statusMQ));
-                    if (tempStatusMQ->isValid() && tempStatusMQ->getEventFlagWord()) {
-                        EventFlag::createEventFlag(tempStatusMQ->getEventFlagWord(), &mEfGroup);
-                    }
-                }
-            });
-    if (!ret.isOk() || retval != Result::OK) {
-        return ret.isOk() ? analyzeResult(retval) : FAILED_TRANSACTION;
-    }
-    if (!tempStatusMQ || !tempStatusMQ->isValid() || !mEfGroup) {
-        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for effects");
-        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
-                "Status message queue for effects is invalid");
-        ALOGE_IF(!mEfGroup, "Event flag creation for effects failed");
-        return NO_INIT;
-    }
-    mStatusMQ = std::move(tempStatusMQ);
-    return OK;
-}
-
-bool EffectHalHidl::needToResetBuffers() {
-    if (mBuffersChanged) return true;
-    bool inBufferFrameCountUpdated = mInBuffer->checkFrameCountChange();
-    bool outBufferFrameCountUpdated = mOutBuffer->checkFrameCountChange();
-    return inBufferFrameCountUpdated || outBufferFrameCountUpdated;
-}
-
-status_t EffectHalHidl::processImpl(uint32_t mqFlag) {
-    if (mEffect == 0 || mInBuffer == 0 || mOutBuffer == 0) return NO_INIT;
-    status_t status;
-    if (!mStatusMQ && (status = prepareForProcessing()) != OK) {
-        return status;
-    }
-    if (needToResetBuffers() && (status = setProcessBuffers()) != OK) {
-        return status;
-    }
-    // The data is already in the buffers, just need to flush it and wake up the server side.
-    std::atomic_thread_fence(std::memory_order_release);
-    mEfGroup->wake(mqFlag);
-    uint32_t efState = 0;
-retry:
-    status_t ret = mEfGroup->wait(
-            static_cast<uint32_t>(MessageQueueFlagBits::DONE_PROCESSING), &efState);
-    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::DONE_PROCESSING)) {
-        Result retval = Result::NOT_INITIALIZED;
-        mStatusMQ->read(&retval);
-        if (retval == Result::OK || retval == Result::INVALID_STATE) {
-            // Sync back the changed contents of the buffer.
-            std::atomic_thread_fence(std::memory_order_acquire);
-        }
-        return analyzeResult(retval);
-    }
-    if (ret == -EAGAIN || ret == -EINTR) {
-        // Spurious wakeup. This normally retries no more than once.
-        goto retry;
-    }
-    return ret;
-}
-
-status_t EffectHalHidl::setProcessBuffers() {
-    Return<Result> ret = mEffect->setProcessBuffers(
-            static_cast<EffectBufferHalHidl*>(mInBuffer.get())->hidlBuffer(),
-            static_cast<EffectBufferHalHidl*>(mOutBuffer.get())->hidlBuffer());
-    if (ret.isOk() && ret == Result::OK) {
-        mBuffersChanged = false;
-        return OK;
-    }
-    return ret.isOk() ? analyzeResult(ret) : FAILED_TRANSACTION;
-}
-
-status_t EffectHalHidl::command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData,
-        uint32_t *replySize, void *pReplyData) {
-    if (mEffect == 0) return NO_INIT;
-
-    // Special cases.
-    if (cmdCode == EFFECT_CMD_SET_CONFIG || cmdCode == EFFECT_CMD_SET_CONFIG_REVERSE) {
-        return setConfigImpl(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
-    } else if (cmdCode == EFFECT_CMD_GET_CONFIG || cmdCode == EFFECT_CMD_GET_CONFIG_REVERSE) {
-        return getConfigImpl(cmdCode, replySize, pReplyData);
-    }
-
-    // Common case.
-    hidl_vec<uint8_t> hidlData;
-    if (pCmdData != nullptr && cmdSize > 0) {
-        hidlData.setToExternal(reinterpret_cast<uint8_t*>(pCmdData), cmdSize);
-    }
-    status_t status;
-    uint32_t replySizeStub = 0;
-    if (replySize == nullptr || pReplyData == nullptr) replySize = &replySizeStub;
-    Return<void> ret = mEffect->command(cmdCode, hidlData, *replySize,
-            [&](int32_t s, const hidl_vec<uint8_t>& result) {
-                status = s;
-                if (status == 0) {
-                    if (*replySize > result.size()) *replySize = result.size();
-                    if (pReplyData != nullptr && *replySize > 0) {
-                        memcpy(pReplyData, &result[0], *replySize);
-                    }
-                }
-            });
-    return ret.isOk() ? status : FAILED_TRANSACTION;
-}
-
-status_t EffectHalHidl::getDescriptor(effect_descriptor_t *pDescriptor) {
-    if (mEffect == 0) return NO_INIT;
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mEffect->getDescriptor(
-            [&](Result r, const EffectDescriptor& result) {
-                retval = r;
-                if (retval == Result::OK) {
-                    effectDescriptorToHal(result, pDescriptor);
-                }
-            });
-    return ret.isOk() ? analyzeResult(retval) : FAILED_TRANSACTION;
-}
-
-status_t EffectHalHidl::close() {
-    if (mEffect == 0) return NO_INIT;
-    Return<Result> ret = mEffect->close();
-    return ret.isOk() ? analyzeResult(ret) : FAILED_TRANSACTION;
-}
-
-status_t EffectHalHidl::getConfigImpl(
-        uint32_t cmdCode, uint32_t *replySize, void *pReplyData) {
-    if (replySize == NULL || *replySize != sizeof(effect_config_t) || pReplyData == NULL) {
-        return BAD_VALUE;
-    }
-    status_t result = FAILED_TRANSACTION;
-    Return<void> ret;
-    if (cmdCode == EFFECT_CMD_GET_CONFIG) {
-        ret = mEffect->getConfig([&] (Result r, const EffectConfig &hidlConfig) {
-            result = analyzeResult(r);
-            if (r == Result::OK) {
-                effectConfigToHal(hidlConfig, static_cast<effect_config_t*>(pReplyData));
-            }
-        });
-    } else {
-        ret = mEffect->getConfigReverse([&] (Result r, const EffectConfig &hidlConfig) {
-            result = analyzeResult(r);
-            if (r == Result::OK) {
-                effectConfigToHal(hidlConfig, static_cast<effect_config_t*>(pReplyData));
-            }
-        });
-    }
-    if (!ret.isOk()) {
-        result = FAILED_TRANSACTION;
-    }
-    return result;
-}
-
-status_t EffectHalHidl::setConfigImpl(
-        uint32_t cmdCode, uint32_t cmdSize, void *pCmdData, uint32_t *replySize, void *pReplyData) {
-    if (pCmdData == NULL || cmdSize != sizeof(effect_config_t) ||
-            replySize == NULL || *replySize != sizeof(int32_t) || pReplyData == NULL) {
-        return BAD_VALUE;
-    }
-    const effect_config_t *halConfig = static_cast<effect_config_t*>(pCmdData);
-    if (halConfig->inputCfg.bufferProvider.getBuffer != NULL ||
-            halConfig->inputCfg.bufferProvider.releaseBuffer != NULL ||
-            halConfig->outputCfg.bufferProvider.getBuffer != NULL ||
-            halConfig->outputCfg.bufferProvider.releaseBuffer != NULL) {
-        ALOGE("Buffer provider callbacks are not supported");
-    }
-    EffectConfig hidlConfig;
-    effectConfigFromHal(*halConfig, &hidlConfig);
-    Return<Result> ret = cmdCode == EFFECT_CMD_SET_CONFIG ?
-            mEffect->setConfig(hidlConfig, nullptr, nullptr) :
-            mEffect->setConfigReverse(hidlConfig, nullptr, nullptr);
-    status_t result = FAILED_TRANSACTION;
-    if (ret.isOk()) {
-        result = analyzeResult(ret);
-        *static_cast<int32_t*>(pReplyData) = result;
-    }
-    return result;
-}
-
-} // namespace android

diff --git a/media/libaudiohal/EffectHalLocal.cpp b/media/libaudiohal/EffectHalLocal.cpp
deleted file mode 100644
index dd465c3..0000000
--- a/media/libaudiohal/EffectHalLocal.cpp
+++ /dev/null

@@ -1,83 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "EffectHalLocal"
-//#define LOG_NDEBUG 0
-
-#include <media/EffectsFactoryApi.h>
-#include <utils/Log.h>
-
-#include "EffectHalLocal.h"
-
-namespace android {
-
-EffectHalLocal::EffectHalLocal(effect_handle_t handle)
-        : mHandle(handle) {
-}
-
-EffectHalLocal::~EffectHalLocal() {
-    int status = EffectRelease(mHandle);
-    ALOGW_IF(status, "Error releasing effect %p: %s", mHandle, strerror(-status));
-    mHandle = 0;
-}
-
-status_t EffectHalLocal::setInBuffer(const sp<EffectBufferHalInterface>& buffer) {
-    mInBuffer = buffer;
-    return OK;
-}
-
-status_t EffectHalLocal::setOutBuffer(const sp<EffectBufferHalInterface>& buffer) {
-    mOutBuffer = buffer;
-    return OK;
-}
-
-status_t EffectHalLocal::process() {
-    if (mInBuffer == nullptr || mOutBuffer == nullptr) {
-        ALOGE_IF(mInBuffer == nullptr, "Input buffer not set");
-        ALOGE_IF(mOutBuffer == nullptr, "Output buffer not set");
-        return NO_INIT;
-    }
-    return (*mHandle)->process(mHandle, mInBuffer->audioBuffer(), mOutBuffer->audioBuffer());
-}
-
-status_t EffectHalLocal::processReverse() {
-    if ((*mHandle)->process_reverse != NULL) {
-        if (mInBuffer == nullptr || mOutBuffer == nullptr) {
-            ALOGE_IF(mInBuffer == nullptr, "Input buffer not set");
-            ALOGE_IF(mOutBuffer == nullptr, "Output buffer not set");
-            return NO_INIT;
-        }
-        return (*mHandle)->process_reverse(
-                mHandle, mInBuffer->audioBuffer(), mOutBuffer->audioBuffer());
-    } else {
-        return INVALID_OPERATION;
-    }
-}
-
-status_t EffectHalLocal::command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData,
-        uint32_t *replySize, void *pReplyData) {
-    return (*mHandle)->command(mHandle, cmdCode, cmdSize, pCmdData, replySize, pReplyData);
-}
-
-status_t EffectHalLocal::getDescriptor(effect_descriptor_t *pDescriptor) {
-    return (*mHandle)->get_descriptor(mHandle, pDescriptor);
-}
-
-status_t EffectHalLocal::close() {
-    return OK;
-}
-
-} // namespace android

diff --git a/media/libaudiohal/EffectHalLocal.h b/media/libaudiohal/EffectHalLocal.h
deleted file mode 100644
index 693fb50..0000000
--- a/media/libaudiohal/EffectHalLocal.h
+++ /dev/null

@@ -1,72 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_EFFECT_HAL_LOCAL_H
-#define ANDROID_HARDWARE_EFFECT_HAL_LOCAL_H
-
-#include <hardware/audio_effect.h>
-#include <media/audiohal/EffectHalInterface.h>
-
-namespace android {
-
-class EffectHalLocal : public EffectHalInterface
-{
-  public:
-    // Set the input buffer.
-    virtual status_t setInBuffer(const sp<EffectBufferHalInterface>& buffer);
-
-    // Set the output buffer.
-    virtual status_t setOutBuffer(const sp<EffectBufferHalInterface>& buffer);
-
-    // Effect process function.
-    virtual status_t process();
-
-    // Process reverse stream function. This function is used to pass
-    // a reference stream to the effect engine.
-    virtual status_t processReverse();
-
-    // Send a command and receive a response to/from effect engine.
-    virtual status_t command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData,
-            uint32_t *replySize, void *pReplyData);
-
-    // Returns the effect descriptor.
-    virtual status_t getDescriptor(effect_descriptor_t *pDescriptor);
-
-    // Free resources on the remote side.
-    virtual status_t close();
-
-    // Whether it's a local implementation.
-    virtual bool isLocal() const { return true; }
-
-    effect_handle_t handle() const { return mHandle; }
-
-  private:
-    effect_handle_t mHandle;
-    sp<EffectBufferHalInterface> mInBuffer;
-    sp<EffectBufferHalInterface> mOutBuffer;
-
-    friend class EffectsFactoryHalLocal;
-
-    // Can not be constructed directly by clients.
-    explicit EffectHalLocal(effect_handle_t handle);
-
-    // The destructor automatically releases the effect.
-    virtual ~EffectHalLocal();
-};
-
-} // namespace android
-
-#endif // ANDROID_HARDWARE_EFFECT_HAL_LOCAL_H

diff --git a/media/libaudiohal/EffectsFactoryHalHidl.cpp b/media/libaudiohal/EffectsFactoryHalHidl.cpp
deleted file mode 100644
index a8081b7..0000000
--- a/media/libaudiohal/EffectsFactoryHalHidl.cpp
+++ /dev/null

@@ -1,148 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "EffectsFactoryHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <cutils/native_handle.h>
-
-#include "ConversionHelperHidl.h"
-#include "EffectHalHidl.h"
-#include "EffectsFactoryHalHidl.h"
-#include "HidlUtils.h"
-
-using ::android::hardware::audio::common::V2_0::Uuid;
-using ::android::hardware::audio::effect::V2_0::IEffect;
-using ::android::hardware::audio::effect::V2_0::Result;
-using ::android::hardware::Return;
-
-namespace android {
-
-// static
-sp<EffectsFactoryHalInterface> EffectsFactoryHalInterface::create() {
-    return new EffectsFactoryHalHidl();
-}
-
-// static
-bool EffectsFactoryHalInterface::isNullUuid(const effect_uuid_t *pEffectUuid) {
-    return memcmp(pEffectUuid, EFFECT_UUID_NULL, sizeof(effect_uuid_t)) == 0;
-}
-
-EffectsFactoryHalHidl::EffectsFactoryHalHidl() : ConversionHelperHidl("EffectsFactory") {
-    mEffectsFactory = IEffectsFactory::getService();
-    if (mEffectsFactory == 0) {
-        ALOGE("Failed to obtain IEffectsFactory service, terminating process.");
-        exit(1);
-    }
-}
-
-EffectsFactoryHalHidl::~EffectsFactoryHalHidl() {
-}
-
-status_t EffectsFactoryHalHidl::queryAllDescriptors() {
-    if (mEffectsFactory == 0) return NO_INIT;
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mEffectsFactory->getAllDescriptors(
-            [&](Result r, const hidl_vec<EffectDescriptor>& result) {
-                retval = r;
-                if (retval == Result::OK) {
-                    mLastDescriptors = result;
-                }
-            });
-    if (ret.isOk()) {
-        return retval == Result::OK ? OK : NO_INIT;
-    }
-    mLastDescriptors.resize(0);
-    return processReturn(__FUNCTION__, ret);
-}
-
-status_t EffectsFactoryHalHidl::queryNumberEffects(uint32_t *pNumEffects) {
-    status_t queryResult = queryAllDescriptors();
-    if (queryResult == OK) {
-        *pNumEffects = mLastDescriptors.size();
-    }
-    return queryResult;
-}
-
-status_t EffectsFactoryHalHidl::getDescriptor(
-        uint32_t index, effect_descriptor_t *pDescriptor) {
-    // TODO: We need somehow to track the changes on the server side
-    // or figure out how to convert everybody to query all the descriptors at once.
-    // TODO: check for nullptr
-    if (mLastDescriptors.size() == 0) {
-        status_t queryResult = queryAllDescriptors();
-        if (queryResult != OK) return queryResult;
-    }
-    if (index >= mLastDescriptors.size()) return NAME_NOT_FOUND;
-    EffectHalHidl::effectDescriptorToHal(mLastDescriptors[index], pDescriptor);
-    return OK;
-}
-
-status_t EffectsFactoryHalHidl::getDescriptor(
-        const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor) {
-    // TODO: check for nullptr
-    if (mEffectsFactory == 0) return NO_INIT;
-    Uuid hidlUuid;
-    HidlUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mEffectsFactory->getDescriptor(hidlUuid,
-            [&](Result r, const EffectDescriptor& result) {
-                retval = r;
-                if (retval == Result::OK) {
-                    EffectHalHidl::effectDescriptorToHal(result, pDescriptor);
-                }
-            });
-    if (ret.isOk()) {
-        if (retval == Result::OK) return OK;
-        else if (retval == Result::INVALID_ARGUMENTS) return NAME_NOT_FOUND;
-        else return NO_INIT;
-    }
-    return processReturn(__FUNCTION__, ret);
-}
-
-status_t EffectsFactoryHalHidl::createEffect(
-        const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId,
-        sp<EffectHalInterface> *effect) {
-    if (mEffectsFactory == 0) return NO_INIT;
-    Uuid hidlUuid;
-    HidlUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mEffectsFactory->createEffect(
-            hidlUuid, sessionId, ioId,
-            [&](Result r, const sp<IEffect>& result, uint64_t effectId) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *effect = new EffectHalHidl(result, effectId);
-                }
-            });
-    if (ret.isOk()) {
-        if (retval == Result::OK) return OK;
-        else if (retval == Result::INVALID_ARGUMENTS) return NAME_NOT_FOUND;
-        else return NO_INIT;
-    }
-    return processReturn(__FUNCTION__, ret);
-}
-
-status_t EffectsFactoryHalHidl::dumpEffects(int fd) {
-    if (mEffectsFactory == 0) return NO_INIT;
-    native_handle_t* hidlHandle = native_handle_create(1, 0);
-    hidlHandle->data[0] = fd;
-    Return<void> ret = mEffectsFactory->debugDump(hidlHandle);
-    native_handle_delete(hidlHandle);
-    return processReturn(__FUNCTION__, ret);
-}
-
-} // namespace android

diff --git a/media/libaudiohal/EffectsFactoryHalHidl.h b/media/libaudiohal/EffectsFactoryHalHidl.h
deleted file mode 100644
index e89f042..0000000
--- a/media/libaudiohal/EffectsFactoryHalHidl.h
+++ /dev/null

@@ -1,67 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H
-#define ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H
-
-#include <android/hardware/audio/effect/2.0/IEffectsFactory.h>
-#include <android/hardware/audio/effect/2.0/types.h>
-#include <media/audiohal/EffectsFactoryHalInterface.h>
-
-namespace android {
-
-using ::android::hardware::audio::effect::V2_0::EffectDescriptor;
-using ::android::hardware::audio::effect::V2_0::IEffectsFactory;
-using ::android::hardware::hidl_vec;
-
-class EffectsFactoryHalHidl : public EffectsFactoryHalInterface, public ConversionHelperHidl
-{
-  public:
-    // Returns the number of different effects in all loaded libraries.
-    virtual status_t queryNumberEffects(uint32_t *pNumEffects);
-
-    // Returns a descriptor of the next available effect.
-    virtual status_t getDescriptor(uint32_t index,
-            effect_descriptor_t *pDescriptor);
-
-    virtual status_t getDescriptor(const effect_uuid_t *pEffectUuid,
-            effect_descriptor_t *pDescriptor);
-
-    // Creates an effect engine of the specified type.
-    // To release the effect engine, it is necessary to release references
-    // to the returned effect object.
-    virtual status_t createEffect(const effect_uuid_t *pEffectUuid,
-            int32_t sessionId, int32_t ioId,
-            sp<EffectHalInterface> *effect);
-
-    virtual status_t dumpEffects(int fd);
-
-  private:
-    friend class EffectsFactoryHalInterface;
-
-    sp<IEffectsFactory> mEffectsFactory;
-    hidl_vec<EffectDescriptor> mLastDescriptors;
-
-    // Can not be constructed directly by clients.
-    EffectsFactoryHalHidl();
-    virtual ~EffectsFactoryHalHidl();
-
-    status_t queryAllDescriptors();
-};
-
-} // namespace android
-
-#endif // ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H

diff --git a/media/libaudiohal/EffectsFactoryHalInterface.cpp b/media/libaudiohal/EffectsFactoryHalInterface.cpp
new file mode 100644
index 0000000..ead1fa2
--- /dev/null
+++ b/media/libaudiohal/EffectsFactoryHalInterface.cpp

@@ -0,0 +1,42 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <android/hardware/audio/effect/2.0/IEffectsFactory.h>
+#include <android/hardware/audio/effect/4.0/IEffectsFactory.h>
+
+#include <EffectsFactoryHalHidl.h>
+#include <libaudiohal/4.0/EffectsFactoryHalHidl.h>
+
+
+namespace android {
+
+// static
+sp<EffectsFactoryHalInterface> EffectsFactoryHalInterface::create() {
+    if (hardware::audio::effect::V4_0::IEffectsFactory::getService() != nullptr) {
+        return new V4_0::EffectsFactoryHalHidl();
+    }
+    if (hardware::audio::effect::V2_0::IEffectsFactory::getService() != nullptr) {
+        return new EffectsFactoryHalHidl();
+    }
+    return nullptr;
+}
+
+// static
+bool EffectsFactoryHalInterface::isNullUuid(const effect_uuid_t *pEffectUuid) {
+    return memcmp(pEffectUuid, EFFECT_UUID_NULL, sizeof(effect_uuid_t)) == 0;
+}
+
+} // namespace android

diff --git a/media/libaudiohal/EffectsFactoryHalLocal.cpp b/media/libaudiohal/EffectsFactoryHalLocal.cpp
deleted file mode 100644
index bbdef5d..0000000
--- a/media/libaudiohal/EffectsFactoryHalLocal.cpp
+++ /dev/null

@@ -1,63 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <media/EffectsFactoryApi.h>
-
-#include "EffectHalLocal.h"
-#include "EffectsFactoryHalLocal.h"
-
-namespace android {
-
-// static
-sp<EffectsFactoryHalInterface> EffectsFactoryHalInterface::create() {
-    return new EffectsFactoryHalLocal();
-}
-
-// static
-bool EffectsFactoryHalInterface::isNullUuid(const effect_uuid_t *pEffectUuid) {
-    return EffectIsNullUuid(pEffectUuid);
-}
-
-status_t EffectsFactoryHalLocal::queryNumberEffects(uint32_t *pNumEffects) {
-    return EffectQueryNumberEffects(pNumEffects);
-}
-
-status_t EffectsFactoryHalLocal::getDescriptor(
-        uint32_t index, effect_descriptor_t *pDescriptor) {
-    return EffectQueryEffect(index, pDescriptor);
-}
-
-status_t EffectsFactoryHalLocal::getDescriptor(
-        const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor) {
-    return EffectGetDescriptor(pEffectUuid, pDescriptor);
-}
-
-status_t EffectsFactoryHalLocal::createEffect(
-        const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId,
-        sp<EffectHalInterface> *effect) {
-    effect_handle_t handle;
-    int result = EffectCreate(pEffectUuid, sessionId, ioId, &handle);
-    if (result == 0) {
-        *effect = new EffectHalLocal(handle);
-    }
-    return result;
-}
-
-status_t EffectsFactoryHalLocal::dumpEffects(int fd) {
-    return EffectDumpEffects(fd);
-}
-
-} // namespace android

diff --git a/media/libaudiohal/EffectsFactoryHalLocal.h b/media/libaudiohal/EffectsFactoryHalLocal.h
deleted file mode 100644
index d5b81be..0000000
--- a/media/libaudiohal/EffectsFactoryHalLocal.h
+++ /dev/null

@@ -1,57 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_LOCAL_H
-#define ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_LOCAL_H
-
-#include <media/audiohal/EffectsFactoryHalInterface.h>
-
-namespace android {
-
-class EffectsFactoryHalLocal : public EffectsFactoryHalInterface
-{
-  public:
-    // Returns the number of different effects in all loaded libraries.
-    virtual status_t queryNumberEffects(uint32_t *pNumEffects);
-
-    // Returns a descriptor of the next available effect.
-    virtual status_t getDescriptor(uint32_t index,
-            effect_descriptor_t *pDescriptor);
-
-    virtual status_t getDescriptor(const effect_uuid_t *pEffectUuid,
-            effect_descriptor_t *pDescriptor);
-
-    // Creates an effect engine of the specified type.
-    // To release the effect engine, it is necessary to release references
-    // to the returned effect object.
-    virtual status_t createEffect(const effect_uuid_t *pEffectUuid,
-            int32_t sessionId, int32_t ioId,
-            sp<EffectHalInterface> *effect);
-
-    virtual status_t dumpEffects(int fd);
-
-  private:
-    friend class EffectsFactoryHalInterface;
-
-    // Can not be constructed directly by clients.
-    EffectsFactoryHalLocal() {}
-
-    virtual ~EffectsFactoryHalLocal() {}
-};
-
-} // namespace android
-
-#endif // ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_LOCAL_H

diff --git a/media/libaudiohal/HalDeathHandlerHidl.cpp b/media/libaudiohal/HalDeathHandlerHidl.cpp
index a742671..6e33523 100644
--- a/media/libaudiohal/HalDeathHandlerHidl.cpp
+++ b/media/libaudiohal/HalDeathHandlerHidl.cpp

@@ -48,12 +48,13 @@
 
 void HalDeathHandler::serviceDied(uint64_t /*cookie*/, const wp<IBase>& /*who*/) {
     // No matter which of the service objects has died,
-    // we need to run all the registered handlers and crash our process.
+    // we need to run all the registered handlers and exit.
     std::lock_guard<std::mutex> guard(mHandlersLock);
     for (const auto& handler : mHandlers) {
         handler.second();
     }
-    LOG_ALWAYS_FATAL("HAL server crashed, need to restart");
+    ALOGE("HAL server crashed, audio server is restarting");
+    _exit(1);  // Avoid calling atexit handlers, as this code runs on a thread from RPC threadpool.
 }
 
 } // namespace android

diff --git a/media/libaudiohal/StreamHalHidl.cpp b/media/libaudiohal/StreamHalHidl.cpp
deleted file mode 100644
index 0cafa36..0000000
--- a/media/libaudiohal/StreamHalHidl.cpp
+++ /dev/null

@@ -1,752 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "StreamHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <android/hardware/audio/2.0/IStreamOutCallback.h>
-#include <hwbinder/IPCThreadState.h>
-#include <mediautils/SchedulingPolicyService.h>
-#include <utils/Log.h>
-
-#include "DeviceHalHidl.h"
-#include "EffectHalHidl.h"
-#include "StreamHalHidl.h"
-
-using ::android::hardware::audio::common::V2_0::AudioChannelMask;
-using ::android::hardware::audio::common::V2_0::AudioFormat;
-using ::android::hardware::audio::common::V2_0::ThreadInfo;
-using ::android::hardware::audio::V2_0::AudioDrain;
-using ::android::hardware::audio::V2_0::IStreamOutCallback;
-using ::android::hardware::audio::V2_0::MessageQueueFlagBits;
-using ::android::hardware::audio::V2_0::MmapBufferInfo;
-using ::android::hardware::audio::V2_0::MmapPosition;
-using ::android::hardware::audio::V2_0::ParameterValue;
-using ::android::hardware::audio::V2_0::Result;
-using ::android::hardware::audio::V2_0::TimeSpec;
-using ::android::hardware::MQDescriptorSync;
-using ::android::hardware::Return;
-using ::android::hardware::Void;
-using ReadCommand = ::android::hardware::audio::V2_0::IStreamIn::ReadCommand;
-
-namespace android {
-
-StreamHalHidl::StreamHalHidl(IStream *stream)
-        : ConversionHelperHidl("Stream"),
-          mStream(stream),
-          mHalThreadPriority(HAL_THREAD_PRIORITY_DEFAULT),
-          mCachedBufferSize(0){
-
-    // Instrument audio signal power logging.
-    // Note: This assumes channel mask, format, and sample rate do not change after creation.
-    if (mStream != nullptr && mStreamPowerLog.isUserDebugOrEngBuild()) {
-        // Obtain audio properties (see StreamHalHidl::getAudioProperties() below).
-        Return<void> ret = mStream->getAudioProperties(
-                [&](uint32_t sr, AudioChannelMask m, AudioFormat f) {
-                mStreamPowerLog.init(sr,
-                        static_cast<audio_channel_mask_t>(m),
-                        static_cast<audio_format_t>(f));
-            });
-    }
-}
-
-StreamHalHidl::~StreamHalHidl() {
-    mStream = nullptr;
-}
-
-status_t StreamHalHidl::getSampleRate(uint32_t *rate) {
-    if (!mStream) return NO_INIT;
-    return processReturn("getSampleRate", mStream->getSampleRate(), rate);
-}
-
-status_t StreamHalHidl::getBufferSize(size_t *size) {
-    if (!mStream) return NO_INIT;
-    status_t status = processReturn("getBufferSize", mStream->getBufferSize(), size);
-    if (status == OK) {
-        mCachedBufferSize = *size;
-    }
-    return status;
-}
-
-status_t StreamHalHidl::getChannelMask(audio_channel_mask_t *mask) {
-    if (!mStream) return NO_INIT;
-    return processReturn("getChannelMask", mStream->getChannelMask(), mask);
-}
-
-status_t StreamHalHidl::getFormat(audio_format_t *format) {
-    if (!mStream) return NO_INIT;
-    return processReturn("getFormat", mStream->getFormat(), format);
-}
-
-status_t StreamHalHidl::getAudioProperties(
-        uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format) {
-    if (!mStream) return NO_INIT;
-    Return<void> ret = mStream->getAudioProperties(
-            [&](uint32_t sr, AudioChannelMask m, AudioFormat f) {
-                *sampleRate = sr;
-                *mask = static_cast<audio_channel_mask_t>(m);
-                *format = static_cast<audio_format_t>(f);
-            });
-    return processReturn("getAudioProperties", ret);
-}
-
-status_t StreamHalHidl::setParameters(const String8& kvPairs) {
-    if (!mStream) return NO_INIT;
-    hidl_vec<ParameterValue> hidlParams;
-    status_t status = parametersFromHal(kvPairs, &hidlParams);
-    if (status != OK) return status;
-    return processReturn("setParameters", mStream->setParameters(hidlParams));
-}
-
-status_t StreamHalHidl::getParameters(const String8& keys, String8 *values) {
-    values->clear();
-    if (!mStream) return NO_INIT;
-    hidl_vec<hidl_string> hidlKeys;
-    status_t status = keysFromHal(keys, &hidlKeys);
-    if (status != OK) return status;
-    Result retval;
-    Return<void> ret = mStream->getParameters(
-            hidlKeys,
-            [&](Result r, const hidl_vec<ParameterValue>& parameters) {
-                retval = r;
-                if (retval == Result::OK) {
-                    parametersToHal(parameters, values);
-                }
-            });
-    return processReturn("getParameters", ret, retval);
-}
-
-status_t StreamHalHidl::addEffect(sp<EffectHalInterface> effect) {
-    if (!mStream) return NO_INIT;
-    return processReturn("addEffect", mStream->addEffect(
-                    static_cast<EffectHalHidl*>(effect.get())->effectId()));
-}
-
-status_t StreamHalHidl::removeEffect(sp<EffectHalInterface> effect) {
-    if (!mStream) return NO_INIT;
-    return processReturn("removeEffect", mStream->removeEffect(
-                    static_cast<EffectHalHidl*>(effect.get())->effectId()));
-}
-
-status_t StreamHalHidl::standby() {
-    if (!mStream) return NO_INIT;
-    return processReturn("standby", mStream->standby());
-}
-
-status_t StreamHalHidl::dump(int fd) {
-    if (!mStream) return NO_INIT;
-    native_handle_t* hidlHandle = native_handle_create(1, 0);
-    hidlHandle->data[0] = fd;
-    Return<void> ret = mStream->debugDump(hidlHandle);
-    native_handle_delete(hidlHandle);
-    mStreamPowerLog.dump(fd);
-    return processReturn("dump", ret);
-}
-
-status_t StreamHalHidl::start() {
-    if (!mStream) return NO_INIT;
-    return processReturn("start", mStream->start());
-}
-
-status_t StreamHalHidl::stop() {
-    if (!mStream) return NO_INIT;
-    return processReturn("stop", mStream->stop());
-}
-
-status_t StreamHalHidl::createMmapBuffer(int32_t minSizeFrames,
-                                  struct audio_mmap_buffer_info *info) {
-    Result retval;
-    Return<void> ret = mStream->createMmapBuffer(
-            minSizeFrames,
-            [&](Result r, const MmapBufferInfo& hidlInfo) {
-                retval = r;
-                if (retval == Result::OK) {
-                    const native_handle *handle = hidlInfo.sharedMemory.handle();
-                    if (handle->numFds > 0) {
-                        info->shared_memory_fd = handle->data[0];
-                        info->buffer_size_frames = hidlInfo.bufferSizeFrames;
-                        info->burst_size_frames = hidlInfo.burstSizeFrames;
-                        // info->shared_memory_address is not needed in HIDL context
-                        info->shared_memory_address = NULL;
-                    } else {
-                        retval = Result::NOT_INITIALIZED;
-                    }
-                }
-            });
-    return processReturn("createMmapBuffer", ret, retval);
-}
-
-status_t StreamHalHidl::getMmapPosition(struct audio_mmap_position *position) {
-    Result retval;
-    Return<void> ret = mStream->getMmapPosition(
-            [&](Result r, const MmapPosition& hidlPosition) {
-                retval = r;
-                if (retval == Result::OK) {
-                    position->time_nanoseconds = hidlPosition.timeNanoseconds;
-                    position->position_frames = hidlPosition.positionFrames;
-                }
-            });
-    return processReturn("getMmapPosition", ret, retval);
-}
-
-status_t StreamHalHidl::setHalThreadPriority(int priority) {
-    mHalThreadPriority = priority;
-    return OK;
-}
-
-status_t StreamHalHidl::getCachedBufferSize(size_t *size) {
-    if (mCachedBufferSize != 0) {
-        *size = mCachedBufferSize;
-        return OK;
-    }
-    return getBufferSize(size);
-}
-
-bool StreamHalHidl::requestHalThreadPriority(pid_t threadPid, pid_t threadId) {
-    if (mHalThreadPriority == HAL_THREAD_PRIORITY_DEFAULT) {
-        return true;
-    }
-    int err = requestPriority(
-            threadPid, threadId,
-            mHalThreadPriority, false /*isForApp*/, true /*asynchronous*/);
-    ALOGE_IF(err, "failed to set priority %d for pid %d tid %d; error %d",
-            mHalThreadPriority, threadPid, threadId, err);
-    // Audio will still work, but latency will be higher and sometimes unacceptable.
-    return err == 0;
-}
-
-namespace {
-
-/* Notes on callback ownership.
-
-This is how (Hw)Binder ownership model looks like. The server implementation
-is owned by Binder framework (via sp<>). Proxies are owned by clients.
-When the last proxy disappears, Binder framework releases the server impl.
-
-Thus, it is not needed to keep any references to StreamOutCallback (this is
-the server impl) -- it will live as long as HAL server holds a strong ref to
-IStreamOutCallback proxy. We clear that reference by calling 'clearCallback'
-from the destructor of StreamOutHalHidl.
-
-The callback only keeps a weak reference to the stream. The stream is owned
-by AudioFlinger.
-
-*/
-
-struct StreamOutCallback : public IStreamOutCallback {
-    StreamOutCallback(const wp<StreamOutHalHidl>& stream) : mStream(stream) {}
-
-    // IStreamOutCallback implementation
-    Return<void> onWriteReady()  override {
-        sp<StreamOutHalHidl> stream = mStream.promote();
-        if (stream != 0) {
-            stream->onWriteReady();
-        }
-        return Void();
-    }
-
-    Return<void> onDrainReady()  override {
-        sp<StreamOutHalHidl> stream = mStream.promote();
-        if (stream != 0) {
-            stream->onDrainReady();
-        }
-        return Void();
-    }
-
-    Return<void> onError()  override {
-        sp<StreamOutHalHidl> stream = mStream.promote();
-        if (stream != 0) {
-            stream->onError();
-        }
-        return Void();
-    }
-
-  private:
-    wp<StreamOutHalHidl> mStream;
-};
-
-}  // namespace
-
-StreamOutHalHidl::StreamOutHalHidl(const sp<IStreamOut>& stream)
-        : StreamHalHidl(stream.get()), mStream(stream), mWriterClient(0), mEfGroup(nullptr) {
-}
-
-StreamOutHalHidl::~StreamOutHalHidl() {
-    if (mStream != 0) {
-        if (mCallback.unsafe_get()) {
-            processReturn("clearCallback", mStream->clearCallback());
-        }
-        processReturn("close", mStream->close());
-        mStream.clear();
-    }
-    mCallback.clear();
-    hardware::IPCThreadState::self()->flushCommands();
-    if (mEfGroup) {
-        EventFlag::deleteEventFlag(&mEfGroup);
-    }
-}
-
-status_t StreamOutHalHidl::getFrameSize(size_t *size) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("getFrameSize", mStream->getFrameSize(), size);
-}
-
-status_t StreamOutHalHidl::getLatency(uint32_t *latency) {
-    if (mStream == 0) return NO_INIT;
-    if (mWriterClient == gettid() && mCommandMQ) {
-        return callWriterThread(
-                WriteCommand::GET_LATENCY, "getLatency", nullptr, 0,
-                [&](const WriteStatus& writeStatus) {
-                    *latency = writeStatus.reply.latencyMs;
-                });
-    } else {
-        return processReturn("getLatency", mStream->getLatency(), latency);
-    }
-}
-
-status_t StreamOutHalHidl::setVolume(float left, float right) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("setVolume", mStream->setVolume(left, right));
-}
-
-status_t StreamOutHalHidl::write(const void *buffer, size_t bytes, size_t *written) {
-    if (mStream == 0) return NO_INIT;
-    *written = 0;
-
-    if (bytes == 0 && !mDataMQ) {
-        // Can't determine the size for the MQ buffer. Wait for a non-empty write request.
-        ALOGW_IF(mCallback.unsafe_get(), "First call to async write with 0 bytes");
-        return OK;
-    }
-
-    status_t status;
-    if (!mDataMQ) {
-        // In case if playback starts close to the end of a compressed track, the bytes
-        // that need to be written is less than the actual buffer size. Need to use
-        // full buffer size for the MQ since otherwise after seeking back to the middle
-        // data will be truncated.
-        size_t bufferSize;
-        if ((status = getCachedBufferSize(&bufferSize)) != OK) {
-            return status;
-        }
-        if (bytes > bufferSize) bufferSize = bytes;
-        if ((status = prepareForWriting(bufferSize)) != OK) {
-            return status;
-        }
-    }
-
-    status = callWriterThread(
-            WriteCommand::WRITE, "write", static_cast<const uint8_t*>(buffer), bytes,
-            [&] (const WriteStatus& writeStatus) {
-                *written = writeStatus.reply.written;
-                // Diagnostics of the cause of b/35813113.
-                ALOGE_IF(*written > bytes,
-                        "hal reports more bytes written than asked for: %lld > %lld",
-                        (long long)*written, (long long)bytes);
-            });
-    mStreamPowerLog.log(buffer, *written);
-    return status;
-}
-
-status_t StreamOutHalHidl::callWriterThread(
-        WriteCommand cmd, const char* cmdName,
-        const uint8_t* data, size_t dataSize, StreamOutHalHidl::WriterCallback callback) {
-    if (!mCommandMQ->write(&cmd)) {
-        ALOGE("command message queue write failed for \"%s\"", cmdName);
-        return -EAGAIN;
-    }
-    if (data != nullptr) {
-        size_t availableToWrite = mDataMQ->availableToWrite();
-        if (dataSize > availableToWrite) {
-            ALOGW("truncating write data from %lld to %lld due to insufficient data queue space",
-                    (long long)dataSize, (long long)availableToWrite);
-            dataSize = availableToWrite;
-        }
-        if (!mDataMQ->write(data, dataSize)) {
-            ALOGE("data message queue write failed for \"%s\"", cmdName);
-        }
-    }
-    mEfGroup->wake(static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY));
-
-    // TODO: Remove manual event flag handling once blocking MQ is implemented. b/33815422
-    uint32_t efState = 0;
-retry:
-    status_t ret = mEfGroup->wait(static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL), &efState);
-    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL)) {
-        WriteStatus writeStatus;
-        writeStatus.retval = Result::NOT_INITIALIZED;
-        if (!mStatusMQ->read(&writeStatus)) {
-            ALOGE("status message read failed for \"%s\"", cmdName);
-        }
-        if (writeStatus.retval == Result::OK) {
-            ret = OK;
-            callback(writeStatus);
-        } else {
-            ret = processReturn(cmdName, writeStatus.retval);
-        }
-        return ret;
-    }
-    if (ret == -EAGAIN || ret == -EINTR) {
-        // Spurious wakeup. This normally retries no more than once.
-        goto retry;
-    }
-    return ret;
-}
-
-status_t StreamOutHalHidl::prepareForWriting(size_t bufferSize) {
-    std::unique_ptr<CommandMQ> tempCommandMQ;
-    std::unique_ptr<DataMQ> tempDataMQ;
-    std::unique_ptr<StatusMQ> tempStatusMQ;
-    Result retval;
-    pid_t halThreadPid, halThreadTid;
-    Return<void> ret = mStream->prepareForWriting(
-            1, bufferSize,
-            [&](Result r,
-                    const CommandMQ::Descriptor& commandMQ,
-                    const DataMQ::Descriptor& dataMQ,
-                    const StatusMQ::Descriptor& statusMQ,
-                    const ThreadInfo& halThreadInfo) {
-                retval = r;
-                if (retval == Result::OK) {
-                    tempCommandMQ.reset(new CommandMQ(commandMQ));
-                    tempDataMQ.reset(new DataMQ(dataMQ));
-                    tempStatusMQ.reset(new StatusMQ(statusMQ));
-                    if (tempDataMQ->isValid() && tempDataMQ->getEventFlagWord()) {
-                        EventFlag::createEventFlag(tempDataMQ->getEventFlagWord(), &mEfGroup);
-                    }
-                    halThreadPid = halThreadInfo.pid;
-                    halThreadTid = halThreadInfo.tid;
-                }
-            });
-    if (!ret.isOk() || retval != Result::OK) {
-        return processReturn("prepareForWriting", ret, retval);
-    }
-    if (!tempCommandMQ || !tempCommandMQ->isValid() ||
-            !tempDataMQ || !tempDataMQ->isValid() ||
-            !tempStatusMQ || !tempStatusMQ->isValid() ||
-            !mEfGroup) {
-        ALOGE_IF(!tempCommandMQ, "Failed to obtain command message queue for writing");
-        ALOGE_IF(tempCommandMQ && !tempCommandMQ->isValid(),
-                "Command message queue for writing is invalid");
-        ALOGE_IF(!tempDataMQ, "Failed to obtain data message queue for writing");
-        ALOGE_IF(tempDataMQ && !tempDataMQ->isValid(), "Data message queue for writing is invalid");
-        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for writing");
-        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
-                "Status message queue for writing is invalid");
-        ALOGE_IF(!mEfGroup, "Event flag creation for writing failed");
-        return NO_INIT;
-    }
-    requestHalThreadPriority(halThreadPid, halThreadTid);
-
-    mCommandMQ = std::move(tempCommandMQ);
-    mDataMQ = std::move(tempDataMQ);
-    mStatusMQ = std::move(tempStatusMQ);
-    mWriterClient = gettid();
-    return OK;
-}
-
-status_t StreamOutHalHidl::getRenderPosition(uint32_t *dspFrames) {
-    if (mStream == 0) return NO_INIT;
-    Result retval;
-    Return<void> ret = mStream->getRenderPosition(
-            [&](Result r, uint32_t d) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *dspFrames = d;
-                }
-            });
-    return processReturn("getRenderPosition", ret, retval);
-}
-
-status_t StreamOutHalHidl::getNextWriteTimestamp(int64_t *timestamp) {
-    if (mStream == 0) return NO_INIT;
-    Result retval;
-    Return<void> ret = mStream->getNextWriteTimestamp(
-            [&](Result r, int64_t t) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *timestamp = t;
-                }
-            });
-    return processReturn("getRenderPosition", ret, retval);
-}
-
-status_t StreamOutHalHidl::setCallback(wp<StreamOutHalInterfaceCallback> callback) {
-    if (mStream == 0) return NO_INIT;
-    status_t status = processReturn(
-            "setCallback", mStream->setCallback(new StreamOutCallback(this)));
-    if (status == OK) {
-        mCallback = callback;
-    }
-    return status;
-}
-
-status_t StreamOutHalHidl::supportsPauseAndResume(bool *supportsPause, bool *supportsResume) {
-    if (mStream == 0) return NO_INIT;
-    Return<void> ret = mStream->supportsPauseAndResume(
-            [&](bool p, bool r) {
-                *supportsPause = p;
-                *supportsResume = r;
-            });
-    return processReturn("supportsPauseAndResume", ret);
-}
-
-status_t StreamOutHalHidl::pause() {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("pause", mStream->pause());
-}
-
-status_t StreamOutHalHidl::resume() {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("pause", mStream->resume());
-}
-
-status_t StreamOutHalHidl::supportsDrain(bool *supportsDrain) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("supportsDrain", mStream->supportsDrain(), supportsDrain);
-}
-
-status_t StreamOutHalHidl::drain(bool earlyNotify) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn(
-            "drain", mStream->drain(earlyNotify ? AudioDrain::EARLY_NOTIFY : AudioDrain::ALL));
-}
-
-status_t StreamOutHalHidl::flush() {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("pause", mStream->flush());
-}
-
-status_t StreamOutHalHidl::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) {
-    if (mStream == 0) return NO_INIT;
-    if (mWriterClient == gettid() && mCommandMQ) {
-        return callWriterThread(
-                WriteCommand::GET_PRESENTATION_POSITION, "getPresentationPosition", nullptr, 0,
-                [&](const WriteStatus& writeStatus) {
-                    *frames = writeStatus.reply.presentationPosition.frames;
-                    timestamp->tv_sec = writeStatus.reply.presentationPosition.timeStamp.tvSec;
-                    timestamp->tv_nsec = writeStatus.reply.presentationPosition.timeStamp.tvNSec;
-                });
-    } else {
-        Result retval;
-        Return<void> ret = mStream->getPresentationPosition(
-                [&](Result r, uint64_t hidlFrames, const TimeSpec& hidlTimeStamp) {
-                    retval = r;
-                    if (retval == Result::OK) {
-                        *frames = hidlFrames;
-                        timestamp->tv_sec = hidlTimeStamp.tvSec;
-                        timestamp->tv_nsec = hidlTimeStamp.tvNSec;
-                    }
-                });
-        return processReturn("getPresentationPosition", ret, retval);
-    }
-}
-
-void StreamOutHalHidl::onWriteReady() {
-    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
-    if (callback == 0) return;
-    ALOGV("asyncCallback onWriteReady");
-    callback->onWriteReady();
-}
-
-void StreamOutHalHidl::onDrainReady() {
-    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
-    if (callback == 0) return;
-    ALOGV("asyncCallback onDrainReady");
-    callback->onDrainReady();
-}
-
-void StreamOutHalHidl::onError() {
-    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
-    if (callback == 0) return;
-    ALOGV("asyncCallback onError");
-    callback->onError();
-}
-
-
-StreamInHalHidl::StreamInHalHidl(const sp<IStreamIn>& stream)
-        : StreamHalHidl(stream.get()), mStream(stream), mReaderClient(0), mEfGroup(nullptr) {
-}
-
-StreamInHalHidl::~StreamInHalHidl() {
-    if (mStream != 0) {
-        processReturn("close", mStream->close());
-        mStream.clear();
-        hardware::IPCThreadState::self()->flushCommands();
-    }
-    if (mEfGroup) {
-        EventFlag::deleteEventFlag(&mEfGroup);
-    }
-}
-
-status_t StreamInHalHidl::getFrameSize(size_t *size) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("getFrameSize", mStream->getFrameSize(), size);
-}
-
-status_t StreamInHalHidl::setGain(float gain) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("setGain", mStream->setGain(gain));
-}
-
-status_t StreamInHalHidl::read(void *buffer, size_t bytes, size_t *read) {
-    if (mStream == 0) return NO_INIT;
-    *read = 0;
-
-    if (bytes == 0 && !mDataMQ) {
-        // Can't determine the size for the MQ buffer. Wait for a non-empty read request.
-        return OK;
-    }
-
-    status_t status;
-    if (!mDataMQ && (status = prepareForReading(bytes)) != OK) {
-        return status;
-    }
-
-    ReadParameters params;
-    params.command = ReadCommand::READ;
-    params.params.read = bytes;
-    status = callReaderThread(params, "read",
-            [&](const ReadStatus& readStatus) {
-                const size_t availToRead = mDataMQ->availableToRead();
-                if (!mDataMQ->read(static_cast<uint8_t*>(buffer), std::min(bytes, availToRead))) {
-                    ALOGE("data message queue read failed for \"read\"");
-                }
-                ALOGW_IF(availToRead != readStatus.reply.read,
-                        "HAL read report inconsistent: mq = %d, status = %d",
-                        (int32_t)availToRead, (int32_t)readStatus.reply.read);
-                *read = readStatus.reply.read;
-            });
-    mStreamPowerLog.log(buffer, *read);
-    return status;
-}
-
-status_t StreamInHalHidl::callReaderThread(
-        const ReadParameters& params, const char* cmdName,
-        StreamInHalHidl::ReaderCallback callback) {
-    if (!mCommandMQ->write(&params)) {
-        ALOGW("command message queue write failed");
-        return -EAGAIN;
-    }
-    mEfGroup->wake(static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL));
-
-    // TODO: Remove manual event flag handling once blocking MQ is implemented. b/33815422
-    uint32_t efState = 0;
-retry:
-    status_t ret = mEfGroup->wait(static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY), &efState);
-    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY)) {
-        ReadStatus readStatus;
-        readStatus.retval = Result::NOT_INITIALIZED;
-        if (!mStatusMQ->read(&readStatus)) {
-            ALOGE("status message read failed for \"%s\"", cmdName);
-        }
-         if (readStatus.retval == Result::OK) {
-            ret = OK;
-            callback(readStatus);
-        } else {
-            ret = processReturn(cmdName, readStatus.retval);
-        }
-        return ret;
-    }
-    if (ret == -EAGAIN || ret == -EINTR) {
-        // Spurious wakeup. This normally retries no more than once.
-        goto retry;
-    }
-    return ret;
-}
-
-status_t StreamInHalHidl::prepareForReading(size_t bufferSize) {
-    std::unique_ptr<CommandMQ> tempCommandMQ;
-    std::unique_ptr<DataMQ> tempDataMQ;
-    std::unique_ptr<StatusMQ> tempStatusMQ;
-    Result retval;
-    pid_t halThreadPid, halThreadTid;
-    Return<void> ret = mStream->prepareForReading(
-            1, bufferSize,
-            [&](Result r,
-                    const CommandMQ::Descriptor& commandMQ,
-                    const DataMQ::Descriptor& dataMQ,
-                    const StatusMQ::Descriptor& statusMQ,
-                    const ThreadInfo& halThreadInfo) {
-                retval = r;
-                if (retval == Result::OK) {
-                    tempCommandMQ.reset(new CommandMQ(commandMQ));
-                    tempDataMQ.reset(new DataMQ(dataMQ));
-                    tempStatusMQ.reset(new StatusMQ(statusMQ));
-                    if (tempDataMQ->isValid() && tempDataMQ->getEventFlagWord()) {
-                        EventFlag::createEventFlag(tempDataMQ->getEventFlagWord(), &mEfGroup);
-                    }
-                    halThreadPid = halThreadInfo.pid;
-                    halThreadTid = halThreadInfo.tid;
-                }
-            });
-    if (!ret.isOk() || retval != Result::OK) {
-        return processReturn("prepareForReading", ret, retval);
-    }
-    if (!tempCommandMQ || !tempCommandMQ->isValid() ||
-            !tempDataMQ || !tempDataMQ->isValid() ||
-            !tempStatusMQ || !tempStatusMQ->isValid() ||
-            !mEfGroup) {
-        ALOGE_IF(!tempCommandMQ, "Failed to obtain command message queue for writing");
-        ALOGE_IF(tempCommandMQ && !tempCommandMQ->isValid(),
-                "Command message queue for writing is invalid");
-        ALOGE_IF(!tempDataMQ, "Failed to obtain data message queue for reading");
-        ALOGE_IF(tempDataMQ && !tempDataMQ->isValid(), "Data message queue for reading is invalid");
-        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for reading");
-        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
-                "Status message queue for reading is invalid");
-        ALOGE_IF(!mEfGroup, "Event flag creation for reading failed");
-        return NO_INIT;
-    }
-    requestHalThreadPriority(halThreadPid, halThreadTid);
-
-    mCommandMQ = std::move(tempCommandMQ);
-    mDataMQ = std::move(tempDataMQ);
-    mStatusMQ = std::move(tempStatusMQ);
-    mReaderClient = gettid();
-    return OK;
-}
-
-status_t StreamInHalHidl::getInputFramesLost(uint32_t *framesLost) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("getInputFramesLost", mStream->getInputFramesLost(), framesLost);
-}
-
-status_t StreamInHalHidl::getCapturePosition(int64_t *frames, int64_t *time) {
-    if (mStream == 0) return NO_INIT;
-    if (mReaderClient == gettid() && mCommandMQ) {
-        ReadParameters params;
-        params.command = ReadCommand::GET_CAPTURE_POSITION;
-        return callReaderThread(params, "getCapturePosition",
-                [&](const ReadStatus& readStatus) {
-                    *frames = readStatus.reply.capturePosition.frames;
-                    *time = readStatus.reply.capturePosition.time;
-                });
-    } else {
-        Result retval;
-        Return<void> ret = mStream->getCapturePosition(
-                [&](Result r, uint64_t hidlFrames, uint64_t hidlTime) {
-                    retval = r;
-                    if (retval == Result::OK) {
-                        *frames = hidlFrames;
-                        *time = hidlTime;
-                    }
-                });
-        return processReturn("getCapturePosition", ret, retval);
-    }
-}
-
-} // namespace android

diff --git a/media/libaudiohal/StreamHalHidl.h b/media/libaudiohal/StreamHalHidl.h
deleted file mode 100644
index d4ab943..0000000
--- a/media/libaudiohal/StreamHalHidl.h
+++ /dev/null

@@ -1,239 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_STREAM_HAL_HIDL_H
-#define ANDROID_HARDWARE_STREAM_HAL_HIDL_H
-
-#include <atomic>
-
-#include <android/hardware/audio/2.0/IStream.h>
-#include <android/hardware/audio/2.0/IStreamIn.h>
-#include <android/hardware/audio/2.0/IStreamOut.h>
-#include <fmq/EventFlag.h>
-#include <fmq/MessageQueue.h>
-#include <media/audiohal/StreamHalInterface.h>
-
-#include "ConversionHelperHidl.h"
-#include "StreamPowerLog.h"
-
-using ::android::hardware::audio::V2_0::IStream;
-using ::android::hardware::audio::V2_0::IStreamIn;
-using ::android::hardware::audio::V2_0::IStreamOut;
-using ::android::hardware::EventFlag;
-using ::android::hardware::MessageQueue;
-using ::android::hardware::Return;
-using ReadParameters = ::android::hardware::audio::V2_0::IStreamIn::ReadParameters;
-using ReadStatus = ::android::hardware::audio::V2_0::IStreamIn::ReadStatus;
-using WriteCommand = ::android::hardware::audio::V2_0::IStreamOut::WriteCommand;
-using WriteStatus = ::android::hardware::audio::V2_0::IStreamOut::WriteStatus;
-
-namespace android {
-
-class DeviceHalHidl;
-
-class StreamHalHidl : public virtual StreamHalInterface, public ConversionHelperHidl
-{
-  public:
-    // Return the sampling rate in Hz - eg. 44100.
-    virtual status_t getSampleRate(uint32_t *rate);
-
-    // Return size of input/output buffer in bytes for this stream - eg. 4800.
-    virtual status_t getBufferSize(size_t *size);
-
-    // Return the channel mask.
-    virtual status_t getChannelMask(audio_channel_mask_t *mask);
-
-    // Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT.
-    virtual status_t getFormat(audio_format_t *format);
-
-    // Convenience method.
-    virtual status_t getAudioProperties(
-            uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format);
-
-    // Set audio stream parameters.
-    virtual status_t setParameters(const String8& kvPairs);
-
-    // Get audio stream parameters.
-    virtual status_t getParameters(const String8& keys, String8 *values);
-
-    // Add or remove the effect on the stream.
-    virtual status_t addEffect(sp<EffectHalInterface> effect);
-    virtual status_t removeEffect(sp<EffectHalInterface> effect);
-
-    // Put the audio hardware input/output into standby mode.
-    virtual status_t standby();
-
-    virtual status_t dump(int fd);
-
-    // Start a stream operating in mmap mode.
-    virtual status_t start();
-
-    // Stop a stream operating in mmap mode.
-    virtual status_t stop();
-
-    // Retrieve information on the data buffer in mmap mode.
-    virtual status_t createMmapBuffer(int32_t minSizeFrames,
-                                      struct audio_mmap_buffer_info *info);
-
-    // Get current read/write position in the mmap buffer
-    virtual status_t getMmapPosition(struct audio_mmap_position *position);
-
-    // Set the priority of the thread that interacts with the HAL
-    // (must match the priority of the audioflinger's thread that calls 'read' / 'write')
-    virtual status_t setHalThreadPriority(int priority);
-
-  protected:
-    // Subclasses can not be constructed directly by clients.
-    explicit StreamHalHidl(IStream *stream);
-
-    // The destructor automatically closes the stream.
-    virtual ~StreamHalHidl();
-
-    status_t getCachedBufferSize(size_t *size);
-
-    bool requestHalThreadPriority(pid_t threadPid, pid_t threadId);
-
-    // mStreamPowerLog is used for audio signal power logging.
-    StreamPowerLog mStreamPowerLog;
-
-  private:
-    const int HAL_THREAD_PRIORITY_DEFAULT = -1;
-    IStream *mStream;
-    int mHalThreadPriority;
-    size_t mCachedBufferSize;
-};
-
-class StreamOutHalHidl : public StreamOutHalInterface, public StreamHalHidl {
-  public:
-    // Return the frame size (number of bytes per sample) of a stream.
-    virtual status_t getFrameSize(size_t *size);
-
-    // Return the audio hardware driver estimated latency in milliseconds.
-    virtual status_t getLatency(uint32_t *latency);
-
-    // Use this method in situations where audio mixing is done in the hardware.
-    virtual status_t setVolume(float left, float right);
-
-    // Write audio buffer to driver.
-    virtual status_t write(const void *buffer, size_t bytes, size_t *written);
-
-    // Return the number of audio frames written by the audio dsp to DAC since
-    // the output has exited standby.
-    virtual status_t getRenderPosition(uint32_t *dspFrames);
-
-    // Get the local time at which the next write to the audio driver will be presented.
-    virtual status_t getNextWriteTimestamp(int64_t *timestamp);
-
-    // Set the callback for notifying completion of non-blocking write and drain.
-    virtual status_t setCallback(wp<StreamOutHalInterfaceCallback> callback);
-
-    // Returns whether pause and resume operations are supported.
-    virtual status_t supportsPauseAndResume(bool *supportsPause, bool *supportsResume);
-
-    // Notifies to the audio driver to resume playback following a pause.
-    virtual status_t pause();
-
-    // Notifies to the audio driver to resume playback following a pause.
-    virtual status_t resume();
-
-    // Returns whether drain operation is supported.
-    virtual status_t supportsDrain(bool *supportsDrain);
-
-    // Requests notification when data buffered by the driver/hardware has been played.
-    virtual status_t drain(bool earlyNotify);
-
-    // Notifies to the audio driver to flush the queued data.
-    virtual status_t flush();
-
-    // Return a recent count of the number of audio frames presented to an external observer.
-    virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
-
-    // Methods used by StreamOutCallback (HIDL).
-    void onWriteReady();
-    void onDrainReady();
-    void onError();
-
-  private:
-    friend class DeviceHalHidl;
-    typedef MessageQueue<WriteCommand, hardware::kSynchronizedReadWrite> CommandMQ;
-    typedef MessageQueue<uint8_t, hardware::kSynchronizedReadWrite> DataMQ;
-    typedef MessageQueue<WriteStatus, hardware::kSynchronizedReadWrite> StatusMQ;
-
-    wp<StreamOutHalInterfaceCallback> mCallback;
-    sp<IStreamOut> mStream;
-    std::unique_ptr<CommandMQ> mCommandMQ;
-    std::unique_ptr<DataMQ> mDataMQ;
-    std::unique_ptr<StatusMQ> mStatusMQ;
-    std::atomic<pid_t> mWriterClient;
-    EventFlag* mEfGroup;
-
-    // Can not be constructed directly by clients.
-    StreamOutHalHidl(const sp<IStreamOut>& stream);
-
-    virtual ~StreamOutHalHidl();
-
-    using WriterCallback = std::function<void(const WriteStatus& writeStatus)>;
-    status_t callWriterThread(
-            WriteCommand cmd, const char* cmdName,
-            const uint8_t* data, size_t dataSize, WriterCallback callback);
-    status_t prepareForWriting(size_t bufferSize);
-};
-
-class StreamInHalHidl : public StreamInHalInterface, public StreamHalHidl {
-  public:
-    // Return the frame size (number of bytes per sample) of a stream.
-    virtual status_t getFrameSize(size_t *size);
-
-    // Set the input gain for the audio driver.
-    virtual status_t setGain(float gain);
-
-    // Read audio buffer in from driver.
-    virtual status_t read(void *buffer, size_t bytes, size_t *read);
-
-    // Return the amount of input frames lost in the audio driver.
-    virtual status_t getInputFramesLost(uint32_t *framesLost);
-
-    // Return a recent count of the number of audio frames received and
-    // the clock time associated with that frame count.
-    virtual status_t getCapturePosition(int64_t *frames, int64_t *time);
-
-  private:
-    friend class DeviceHalHidl;
-    typedef MessageQueue<ReadParameters, hardware::kSynchronizedReadWrite> CommandMQ;
-    typedef MessageQueue<uint8_t, hardware::kSynchronizedReadWrite> DataMQ;
-    typedef MessageQueue<ReadStatus, hardware::kSynchronizedReadWrite> StatusMQ;
-
-    sp<IStreamIn> mStream;
-    std::unique_ptr<CommandMQ> mCommandMQ;
-    std::unique_ptr<DataMQ> mDataMQ;
-    std::unique_ptr<StatusMQ> mStatusMQ;
-    std::atomic<pid_t> mReaderClient;
-    EventFlag* mEfGroup;
-
-    // Can not be constructed directly by clients.
-    StreamInHalHidl(const sp<IStreamIn>& stream);
-
-    virtual ~StreamInHalHidl();
-
-    using ReaderCallback = std::function<void(const ReadStatus& readStatus)>;
-    status_t callReaderThread(
-            const ReadParameters& params, const char* cmdName, ReaderCallback callback);
-    status_t prepareForReading(size_t bufferSize);
-};
-
-} // namespace android
-
-#endif // ANDROID_HARDWARE_STREAM_HAL_HIDL_H

diff --git a/media/libaudiohal/StreamHalLocal.cpp b/media/libaudiohal/StreamHalLocal.cpp
deleted file mode 100644
index dc17f5c..0000000
--- a/media/libaudiohal/StreamHalLocal.cpp
+++ /dev/null

@@ -1,319 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "StreamHalLocal"
-//#define LOG_NDEBUG 0
-
-#include <hardware/audio.h>
-#include <utils/Log.h>
-
-#include "DeviceHalLocal.h"
-#include "EffectHalLocal.h"
-#include "StreamHalLocal.h"
-
-namespace android {
-
-StreamHalLocal::StreamHalLocal(audio_stream_t *stream, sp<DeviceHalLocal> device)
-        : mDevice(device),
-          mStream(stream) {
-    // Instrument audio signal power logging.
-    // Note: This assumes channel mask, format, and sample rate do not change after creation.
-    if (mStream != nullptr && mStreamPowerLog.isUserDebugOrEngBuild()) {
-        mStreamPowerLog.init(mStream->get_sample_rate(mStream),
-                mStream->get_channels(mStream),
-                mStream->get_format(mStream));
-    }
-}
-
-StreamHalLocal::~StreamHalLocal() {
-    mStream = 0;
-    mDevice.clear();
-}
-
-status_t StreamHalLocal::getSampleRate(uint32_t *rate) {
-    *rate = mStream->get_sample_rate(mStream);
-    return OK;
-}
-
-status_t StreamHalLocal::getBufferSize(size_t *size) {
-    *size = mStream->get_buffer_size(mStream);
-    return OK;
-}
-
-status_t StreamHalLocal::getChannelMask(audio_channel_mask_t *mask) {
-    *mask = mStream->get_channels(mStream);
-    return OK;
-}
-
-status_t StreamHalLocal::getFormat(audio_format_t *format) {
-    *format = mStream->get_format(mStream);
-    return OK;
-}
-
-status_t StreamHalLocal::getAudioProperties(
-        uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format) {
-    *sampleRate = mStream->get_sample_rate(mStream);
-    *mask = mStream->get_channels(mStream);
-    *format = mStream->get_format(mStream);
-    return OK;
-}
-
-status_t StreamHalLocal::setParameters(const String8& kvPairs) {
-    return mStream->set_parameters(mStream, kvPairs.string());
-}
-
-status_t StreamHalLocal::getParameters(const String8& keys, String8 *values) {
-    char *halValues = mStream->get_parameters(mStream, keys.string());
-    if (halValues != NULL) {
-        values->setTo(halValues);
-        free(halValues);
-    } else {
-        values->clear();
-    }
-    return OK;
-}
-
-status_t StreamHalLocal::addEffect(sp<EffectHalInterface> effect) {
-    LOG_ALWAYS_FATAL_IF(!effect->isLocal(), "Only local effects can be added for a local stream");
-    return mStream->add_audio_effect(mStream,
-            static_cast<EffectHalLocal*>(effect.get())->handle());
-}
-
-status_t StreamHalLocal::removeEffect(sp<EffectHalInterface> effect) {
-    LOG_ALWAYS_FATAL_IF(!effect->isLocal(), "Only local effects can be removed for a local stream");
-    return mStream->remove_audio_effect(mStream,
-            static_cast<EffectHalLocal*>(effect.get())->handle());
-}
-
-status_t StreamHalLocal::standby() {
-    return mStream->standby(mStream);
-}
-
-status_t StreamHalLocal::dump(int fd) {
-    status_t status = mStream->dump(mStream, fd);
-    mStreamPowerLog.dump(fd);
-    return status;
-}
-
-status_t StreamHalLocal::setHalThreadPriority(int) {
-    // Don't need to do anything as local hal is executed by audioflinger directly
-    // on the same thread.
-    return OK;
-}
-
-StreamOutHalLocal::StreamOutHalLocal(audio_stream_out_t *stream, sp<DeviceHalLocal> device)
-        : StreamHalLocal(&stream->common, device), mStream(stream) {
-}
-
-StreamOutHalLocal::~StreamOutHalLocal() {
-    mCallback.clear();
-    mDevice->closeOutputStream(mStream);
-    mStream = 0;
-}
-
-status_t StreamOutHalLocal::getFrameSize(size_t *size) {
-    *size = audio_stream_out_frame_size(mStream);
-    return OK;
-}
-
-status_t StreamOutHalLocal::getLatency(uint32_t *latency) {
-    *latency = mStream->get_latency(mStream);
-    return OK;
-}
-
-status_t StreamOutHalLocal::setVolume(float left, float right) {
-    if (mStream->set_volume == NULL) return INVALID_OPERATION;
-    return mStream->set_volume(mStream, left, right);
-}
-
-status_t StreamOutHalLocal::write(const void *buffer, size_t bytes, size_t *written) {
-    ssize_t writeResult = mStream->write(mStream, buffer, bytes);
-    if (writeResult > 0) {
-        *written = writeResult;
-        mStreamPowerLog.log(buffer, *written);
-        return OK;
-    } else {
-        *written = 0;
-        return writeResult;
-    }
-}
-
-status_t StreamOutHalLocal::getRenderPosition(uint32_t *dspFrames) {
-    return mStream->get_render_position(mStream, dspFrames);
-}
-
-status_t StreamOutHalLocal::getNextWriteTimestamp(int64_t *timestamp) {
-    if (mStream->get_next_write_timestamp == NULL) return INVALID_OPERATION;
-    return mStream->get_next_write_timestamp(mStream, timestamp);
-}
-
-status_t StreamOutHalLocal::setCallback(wp<StreamOutHalInterfaceCallback> callback) {
-    if (mStream->set_callback == NULL) return INVALID_OPERATION;
-    status_t result = mStream->set_callback(mStream, StreamOutHalLocal::asyncCallback, this);
-    if (result == OK) {
-        mCallback = callback;
-    }
-    return result;
-}
-
-// static
-int StreamOutHalLocal::asyncCallback(stream_callback_event_t event, void*, void *cookie) {
-    // We act as if we gave a wp<StreamOutHalLocal> to HAL. This way we should handle
-    // correctly the case when the callback is invoked while StreamOutHalLocal's destructor is
-    // already running, because the destructor is invoked after the refcount has been atomically
-    // decremented.
-    wp<StreamOutHalLocal> weakSelf(static_cast<StreamOutHalLocal*>(cookie));
-    sp<StreamOutHalLocal> self = weakSelf.promote();
-    if (self == 0) return 0;
-    sp<StreamOutHalInterfaceCallback> callback = self->mCallback.promote();
-    if (callback == 0) return 0;
-    ALOGV("asyncCallback() event %d", event);
-    switch (event) {
-        case STREAM_CBK_EVENT_WRITE_READY:
-            callback->onWriteReady();
-            break;
-        case STREAM_CBK_EVENT_DRAIN_READY:
-            callback->onDrainReady();
-            break;
-        case STREAM_CBK_EVENT_ERROR:
-            callback->onError();
-            break;
-        default:
-            ALOGW("asyncCallback() unknown event %d", event);
-            break;
-    }
-    return 0;
-}
-
-status_t StreamOutHalLocal::supportsPauseAndResume(bool *supportsPause, bool *supportsResume) {
-    *supportsPause = mStream->pause != NULL;
-    *supportsResume = mStream->resume != NULL;
-    return OK;
-}
-
-status_t StreamOutHalLocal::pause() {
-    if (mStream->pause == NULL) return INVALID_OPERATION;
-    return mStream->pause(mStream);
-}
-
-status_t StreamOutHalLocal::resume() {
-    if (mStream->resume == NULL) return INVALID_OPERATION;
-    return mStream->resume(mStream);
-}
-
-status_t StreamOutHalLocal::supportsDrain(bool *supportsDrain) {
-    *supportsDrain = mStream->drain != NULL;
-    return OK;
-}
-
-status_t StreamOutHalLocal::drain(bool earlyNotify) {
-    if (mStream->drain == NULL) return INVALID_OPERATION;
-    return mStream->drain(mStream, earlyNotify ? AUDIO_DRAIN_EARLY_NOTIFY : AUDIO_DRAIN_ALL);
-}
-
-status_t StreamOutHalLocal::flush() {
-    if (mStream->flush == NULL) return INVALID_OPERATION;
-    return mStream->flush(mStream);
-}
-
-status_t StreamOutHalLocal::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) {
-    if (mStream->get_presentation_position == NULL) return INVALID_OPERATION;
-    return mStream->get_presentation_position(mStream, frames, timestamp);
-}
-
-status_t StreamOutHalLocal::start() {
-    if (mStream->start == NULL) return INVALID_OPERATION;
-    return mStream->start(mStream);
-}
-
-status_t StreamOutHalLocal::stop() {
-    if (mStream->stop == NULL) return INVALID_OPERATION;
-    return mStream->stop(mStream);
-}
-
-status_t StreamOutHalLocal::createMmapBuffer(int32_t minSizeFrames,
-                                  struct audio_mmap_buffer_info *info) {
-    if (mStream->create_mmap_buffer == NULL) return INVALID_OPERATION;
-    return mStream->create_mmap_buffer(mStream, minSizeFrames, info);
-}
-
-status_t StreamOutHalLocal::getMmapPosition(struct audio_mmap_position *position) {
-    if (mStream->get_mmap_position == NULL) return INVALID_OPERATION;
-    return mStream->get_mmap_position(mStream, position);
-}
-
-StreamInHalLocal::StreamInHalLocal(audio_stream_in_t *stream, sp<DeviceHalLocal> device)
-        : StreamHalLocal(&stream->common, device), mStream(stream) {
-}
-
-StreamInHalLocal::~StreamInHalLocal() {
-    mDevice->closeInputStream(mStream);
-    mStream = 0;
-}
-
-status_t StreamInHalLocal::getFrameSize(size_t *size) {
-    *size = audio_stream_in_frame_size(mStream);
-    return OK;
-}
-
-status_t StreamInHalLocal::setGain(float gain) {
-    return mStream->set_gain(mStream, gain);
-}
-
-status_t StreamInHalLocal::read(void *buffer, size_t bytes, size_t *read) {
-    ssize_t readResult = mStream->read(mStream, buffer, bytes);
-    if (readResult > 0) {
-        *read = readResult;
-        mStreamPowerLog.log( buffer, *read);
-        return OK;
-    } else {
-        *read = 0;
-        return readResult;
-    }
-}
-
-status_t StreamInHalLocal::getInputFramesLost(uint32_t *framesLost) {
-    *framesLost = mStream->get_input_frames_lost(mStream);
-    return OK;
-}
-
-status_t StreamInHalLocal::getCapturePosition(int64_t *frames, int64_t *time) {
-    if (mStream->get_capture_position == NULL) return INVALID_OPERATION;
-    return mStream->get_capture_position(mStream, frames, time);
-}
-
-status_t StreamInHalLocal::start() {
-    if (mStream->start == NULL) return INVALID_OPERATION;
-    return mStream->start(mStream);
-}
-
-status_t StreamInHalLocal::stop() {
-    if (mStream->stop == NULL) return INVALID_OPERATION;
-    return mStream->stop(mStream);
-}
-
-status_t StreamInHalLocal::createMmapBuffer(int32_t minSizeFrames,
-                                  struct audio_mmap_buffer_info *info) {
-    if (mStream->create_mmap_buffer == NULL) return INVALID_OPERATION;
-    return mStream->create_mmap_buffer(mStream, minSizeFrames, info);
-}
-
-status_t StreamInHalLocal::getMmapPosition(struct audio_mmap_position *position) {
-    if (mStream->get_mmap_position == NULL) return INVALID_OPERATION;
-    return mStream->get_mmap_position(mStream, position);
-}
-
-} // namespace android

diff --git a/media/libaudiohal/StreamHalLocal.h b/media/libaudiohal/StreamHalLocal.h
deleted file mode 100644
index c7136df..0000000
--- a/media/libaudiohal/StreamHalLocal.h
+++ /dev/null

@@ -1,210 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_STREAM_HAL_LOCAL_H
-#define ANDROID_HARDWARE_STREAM_HAL_LOCAL_H
-
-#include <media/audiohal/StreamHalInterface.h>
-#include "StreamPowerLog.h"
-
-namespace android {
-
-class DeviceHalLocal;
-
-class StreamHalLocal : public virtual StreamHalInterface
-{
-  public:
-    // Return the sampling rate in Hz - eg. 44100.
-    virtual status_t getSampleRate(uint32_t *rate);
-
-    // Return size of input/output buffer in bytes for this stream - eg. 4800.
-    virtual status_t getBufferSize(size_t *size);
-
-    // Return the channel mask.
-    virtual status_t getChannelMask(audio_channel_mask_t *mask);
-
-    // Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT.
-    virtual status_t getFormat(audio_format_t *format);
-
-    // Convenience method.
-    virtual status_t getAudioProperties(
-            uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format);
-
-    // Set audio stream parameters.
-    virtual status_t setParameters(const String8& kvPairs);
-
-    // Get audio stream parameters.
-    virtual status_t getParameters(const String8& keys, String8 *values);
-
-    // Add or remove the effect on the stream.
-    virtual status_t addEffect(sp<EffectHalInterface> effect);
-    virtual status_t removeEffect(sp<EffectHalInterface> effect);
-
-    // Put the audio hardware input/output into standby mode.
-    virtual status_t standby();
-
-    virtual status_t dump(int fd);
-
-    // Start a stream operating in mmap mode.
-    virtual status_t start() = 0;
-
-    // Stop a stream operating in mmap mode.
-    virtual status_t stop() = 0;
-
-    // Retrieve information on the data buffer in mmap mode.
-    virtual status_t createMmapBuffer(int32_t minSizeFrames,
-                                      struct audio_mmap_buffer_info *info) = 0;
-
-    // Get current read/write position in the mmap buffer
-    virtual status_t getMmapPosition(struct audio_mmap_position *position) = 0;
-
-    // Set the priority of the thread that interacts with the HAL
-    // (must match the priority of the audioflinger's thread that calls 'read' / 'write')
-    virtual status_t setHalThreadPriority(int priority);
-
-  protected:
-    // Subclasses can not be constructed directly by clients.
-    StreamHalLocal(audio_stream_t *stream, sp<DeviceHalLocal> device);
-
-    // The destructor automatically closes the stream.
-    virtual ~StreamHalLocal();
-
-    sp<DeviceHalLocal> mDevice;
-
-    // mStreamPowerLog is used for audio signal power logging.
-    StreamPowerLog mStreamPowerLog;
-
-  private:
-    audio_stream_t *mStream;
-};
-
-class StreamOutHalLocal : public StreamOutHalInterface, public StreamHalLocal {
-  public:
-    // Return the frame size (number of bytes per sample) of a stream.
-    virtual status_t getFrameSize(size_t *size);
-
-    // Return the audio hardware driver estimated latency in milliseconds.
-    virtual status_t getLatency(uint32_t *latency);
-
-    // Use this method in situations where audio mixing is done in the hardware.
-    virtual status_t setVolume(float left, float right);
-
-    // Write audio buffer to driver.
-    virtual status_t write(const void *buffer, size_t bytes, size_t *written);
-
-    // Return the number of audio frames written by the audio dsp to DAC since
-    // the output has exited standby.
-    virtual status_t getRenderPosition(uint32_t *dspFrames);
-
-    // Get the local time at which the next write to the audio driver will be presented.
-    virtual status_t getNextWriteTimestamp(int64_t *timestamp);
-
-    // Set the callback for notifying completion of non-blocking write and drain.
-    virtual status_t setCallback(wp<StreamOutHalInterfaceCallback> callback);
-
-    // Returns whether pause and resume operations are supported.
-    virtual status_t supportsPauseAndResume(bool *supportsPause, bool *supportsResume);
-
-    // Notifies to the audio driver to resume playback following a pause.
-    virtual status_t pause();
-
-    // Notifies to the audio driver to resume playback following a pause.
-    virtual status_t resume();
-
-    // Returns whether drain operation is supported.
-    virtual status_t supportsDrain(bool *supportsDrain);
-
-    // Requests notification when data buffered by the driver/hardware has been played.
-    virtual status_t drain(bool earlyNotify);
-
-    // Notifies to the audio driver to flush the queued data.
-    virtual status_t flush();
-
-    // Return a recent count of the number of audio frames presented to an external observer.
-    virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
-
-    // Start a stream operating in mmap mode.
-    virtual status_t start();
-
-    // Stop a stream operating in mmap mode.
-    virtual status_t stop();
-
-    // Retrieve information on the data buffer in mmap mode.
-    virtual status_t createMmapBuffer(int32_t minSizeFrames,
-                                      struct audio_mmap_buffer_info *info);
-
-    // Get current read/write position in the mmap buffer
-    virtual status_t getMmapPosition(struct audio_mmap_position *position);
-
-  private:
-    audio_stream_out_t *mStream;
-    wp<StreamOutHalInterfaceCallback> mCallback;
-
-    friend class DeviceHalLocal;
-
-    // Can not be constructed directly by clients.
-    StreamOutHalLocal(audio_stream_out_t *stream, sp<DeviceHalLocal> device);
-
-    virtual ~StreamOutHalLocal();
-
-    static int asyncCallback(stream_callback_event_t event, void *param, void *cookie);
-};
-
-class StreamInHalLocal : public StreamInHalInterface, public StreamHalLocal {
-  public:
-    // Return the frame size (number of bytes per sample) of a stream.
-    virtual status_t getFrameSize(size_t *size);
-
-    // Set the input gain for the audio driver.
-    virtual status_t setGain(float gain);
-
-    // Read audio buffer in from driver.
-    virtual status_t read(void *buffer, size_t bytes, size_t *read);
-
-    // Return the amount of input frames lost in the audio driver.
-    virtual status_t getInputFramesLost(uint32_t *framesLost);
-
-    // Return a recent count of the number of audio frames received and
-    // the clock time associated with that frame count.
-    virtual status_t getCapturePosition(int64_t *frames, int64_t *time);
-
-    // Start a stream operating in mmap mode.
-    virtual status_t start();
-
-    // Stop a stream operating in mmap mode.
-    virtual status_t stop();
-
-    // Retrieve information on the data buffer in mmap mode.
-    virtual status_t createMmapBuffer(int32_t minSizeFrames,
-                                      struct audio_mmap_buffer_info *info);
-
-    // Get current read/write position in the mmap buffer
-    virtual status_t getMmapPosition(struct audio_mmap_position *position);
-
-  private:
-    audio_stream_in_t *mStream;
-
-    friend class DeviceHalLocal;
-
-    // Can not be constructed directly by clients.
-    StreamInHalLocal(audio_stream_in_t *stream, sp<DeviceHalLocal> device);
-
-    virtual ~StreamInHalLocal();
-};
-
-} // namespace android
-
-#endif // ANDROID_HARDWARE_STREAM_HAL_LOCAL_H

diff --git a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
index caf01be..7de8eb3 100644
--- a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h

@@ -17,6 +17,7 @@
 #ifndef ANDROID_HARDWARE_DEVICE_HAL_INTERFACE_H
 #define ANDROID_HARDWARE_DEVICE_HAL_INTERFACE_H
 
+#include <media/MicrophoneInfo.h>
 #include <system/audio.h>
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
@@ -105,6 +106,9 @@
     // Set audio port configuration.
     virtual status_t setAudioPortConfig(const struct audio_port_config *config) = 0;
 
+    // List microphones
+    virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones) = 0;
+
     virtual status_t dump(int fd) = 0;
 
   protected:

diff --git a/media/libaudiohal/include/media/audiohal/EffectBufferHalInterface.h b/media/libaudiohal/include/media/audiohal/EffectBufferHalInterface.h
index e862f6e..d0603cd 100644
--- a/media/libaudiohal/include/media/audiohal/EffectBufferHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/EffectBufferHalInterface.h

@@ -26,6 +26,7 @@
 // Abstraction for an audio buffer. It may be a "mirror" for
 // a buffer that the effect chain doesn't own, or a buffer owned by
 // the effect chain.
+// Buffers are created from EffectsFactoryHalInterface
 class EffectBufferHalInterface : public RefBase
 {
   public:
@@ -37,6 +38,8 @@
         return externalData() != nullptr ? externalData() : audioBuffer()->raw;
     }
 
+    virtual size_t getSize() const = 0;
+
     virtual void setExternalData(void* external) = 0;
     virtual void setFrameCount(size_t frameCount) = 0;
     virtual bool checkFrameCountChange() = 0;  // returns whether frame count has been updated
@@ -47,9 +50,6 @@
     virtual void update(size_t size) = 0;  // copies partial data from external buffer
     virtual void commit(size_t size) = 0;  // copies partial data to external buffer
 
-    static status_t allocate(size_t size, sp<EffectBufferHalInterface>* buffer);
-    static status_t mirror(void* external, size_t size, sp<EffectBufferHalInterface>* buffer);
-
   protected:
     // Subclasses can not be constructed directly by clients.
     EffectBufferHalInterface() {}

diff --git a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
index a616e86..316a46c 100644
--- a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h

@@ -48,6 +48,10 @@
 
     static sp<EffectsFactoryHalInterface> create();
 
+    virtual status_t allocateBuffer(size_t size, sp<EffectBufferHalInterface>* buffer) = 0;
+    virtual status_t mirrorBuffer(void* external, size_t size,
+                                  sp<EffectBufferHalInterface>* buffer) = 0;
+
     // Helper function to compare effect uuid to EFFECT_UUID_NULL.
     static bool isNullUuid(const effect_uuid_t *pEffectUuid);
 

diff --git a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
index 7419c34..c969e28 100644
--- a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h

@@ -17,7 +17,10 @@
 #ifndef ANDROID_HARDWARE_STREAM_HAL_INTERFACE_H
 #define ANDROID_HARDWARE_STREAM_HAL_INTERFACE_H
 
+#include <vector>
+
 #include <media/audiohal/EffectHalInterface.h>
+#include <media/MicrophoneInfo.h>
 #include <system/audio.h>
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
@@ -142,6 +145,15 @@
     // Return a recent count of the number of audio frames presented to an external observer.
     virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp) = 0;
 
+    struct SourceMetadata {
+        std::vector<playback_track_metadata_t> tracks;
+    };
+    /**
+     * Called when the metadata of the stream's source has been changed.
+     * @param sourceMetadata Description of the audio that is played by the clients.
+     */
+    virtual status_t updateSourceMetadata(const SourceMetadata& sourceMetadata) = 0;
+
   protected:
     virtual ~StreamOutHalInterface() {}
 };
@@ -161,6 +173,18 @@
     // the clock time associated with that frame count.
     virtual status_t getCapturePosition(int64_t *frames, int64_t *time) = 0;
 
+    // Get active microphones
+    virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones) = 0;
+
+    struct SinkMetadata {
+        std::vector<record_track_metadata_t> tracks;
+    };
+    /**
+     * Called when the metadata of the stream's sink has been changed.
+     * @param sinkMetadata Description of the audio that is suggested by the clients.
+     */
+    virtual status_t updateSinkMetadata(const SinkMetadata& sinkMetadata) = 0;
+
   protected:
     virtual ~StreamInHalInterface() {}
 };

diff --git a/media/libaudioprocessing/Android.mk b/media/libaudioprocessing/Android.mk
index c850984..da1ecc2 100644
--- a/media/libaudioprocessing/Android.mk
+++ b/media/libaudioprocessing/Android.mk

@@ -24,6 +24,7 @@
     libcutils \
     liblog \
     libnbaio \
+    libnblog \
     libsonic \
     libutils \
 

diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index 238925d..f6f817a 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp

@@ -62,21 +62,22 @@
 #define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
 #endif
 
-// TODO: Move these macro/inlines to a header file.
-template <typename T>
-static inline
-T max(const T& x, const T& y) {
-    return x > y ? x : y;
-}
-
 // Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
 // original code will be used for stereo sinks, the new mixer for multichannel.
-static const bool kUseNewMixer = true;
+static constexpr bool kUseNewMixer = true;
 
 // Set kUseFloat to true to allow floating input into the mixer engine.
 // If kUseNewMixer is false, this is ignored or may be overridden internally
 // because of downmix/upmix support.
-static const bool kUseFloat = true;
+static constexpr bool kUseFloat = true;
+
+#ifdef FLOAT_AUX
+using TYPE_AUX = float;
+static_assert(kUseNewMixer && kUseFloat,
+        "kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
+#else
+using TYPE_AUX = int32_t; // q4.27
+#endif
 
 // Set to default copy buffer size in frames for input processing.
 static const size_t kCopyBufferFrameCount = 256;
@@ -85,88 +86,28 @@
 
 // ----------------------------------------------------------------------------
 
-template <typename T>
-T min(const T& a, const T& b)
-{
-    return a < b ? a : b;
-}
-
-// ----------------------------------------------------------------------------
-
-// Ensure mConfiguredNames bitmask is initialized properly on all architectures.
-// The value of 1 << x is undefined in C when x >= 32.
-
-AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks)
-    :   mTrackNames(0), mConfiguredNames((maxNumTracks >= 32 ? 0 : 1 << maxNumTracks) - 1),
-        mSampleRate(sampleRate)
-{
-    ALOG_ASSERT(maxNumTracks <= MAX_NUM_TRACKS, "maxNumTracks %u > MAX_NUM_TRACKS %u",
-            maxNumTracks, MAX_NUM_TRACKS);
-
-    // AudioMixer is not yet capable of more than 32 active track inputs
-    ALOG_ASSERT(32 >= MAX_NUM_TRACKS, "bad MAX_NUM_TRACKS %d", MAX_NUM_TRACKS);
-
-    pthread_once(&sOnceControl, &sInitRoutine);
-
-    mState.enabledTracks= 0;
-    mState.needsChanged = 0;
-    mState.frameCount   = frameCount;
-    mState.hook         = process__nop;
-    mState.outputTemp   = NULL;
-    mState.resampleTemp = NULL;
-    mState.mNBLogWriter = &mDummyLogWriter;
-    // mState.reserved
-
-    // FIXME Most of the following initialization is probably redundant since
-    // tracks[i] should only be referenced if (mTrackNames & (1 << i)) != 0
-    // and mTrackNames is initially 0.  However, leave it here until that's verified.
-    track_t* t = mState.tracks;
-    for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
-        t->resampler = NULL;
-        t->downmixerBufferProvider = NULL;
-        t->mReformatBufferProvider = NULL;
-        t->mTimestretchBufferProvider = NULL;
-        t++;
-    }
-
-}
-
-AudioMixer::~AudioMixer()
-{
-    track_t* t = mState.tracks;
-    for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
-        delete t->resampler;
-        delete t->downmixerBufferProvider;
-        delete t->mReformatBufferProvider;
-        delete t->mTimestretchBufferProvider;
-        t++;
-    }
-    delete [] mState.outputTemp;
-    delete [] mState.resampleTemp;
-}
-
-void AudioMixer::setNBLogWriter(NBLog::Writer *logWriter)
-{
-    mState.mNBLogWriter = logWriter;
-}
-
 static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
     return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
 }
 
-int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
-        audio_format_t format, int sessionId)
+status_t AudioMixer::create(
+        int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
 {
-    if (!isValidPcmTrackFormat(format)) {
-        ALOGE("AudioMixer::getTrackName invalid format (%#x)", format);
-        return -1;
+    LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);
+
+    if (!isValidChannelMask(channelMask)) {
+        ALOGE("%s invalid channelMask: %#x", __func__, channelMask);
+        return BAD_VALUE;
     }
-    uint32_t names = (~mTrackNames) & mConfiguredNames;
-    if (names != 0) {
-        int n = __builtin_ctz(names);
-        ALOGV("add track (%d)", n);
+    if (!isValidFormat(format)) {
+        ALOGE("%s invalid format: %#x", __func__, format);
+        return BAD_VALUE;
+    }
+
+    auto t = std::make_shared<Track>();
+    {
+        // TODO: move initialization to the Track constructor.
         // assume default parameters for the track, except where noted below
-        track_t* t = &mState.tracks[n];
         t->needs = 0;
 
         // Integer volume.
@@ -207,17 +148,12 @@
         // no initialization needed
         // t->buffer.frameCount
         t->hook = NULL;
-        t->in = NULL;
-        t->resampler = NULL;
+        t->mIn = NULL;
         t->sampleRate = mSampleRate;
         // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
         t->mainBuffer = NULL;
         t->auxBuffer = NULL;
         t->mInputBufferProvider = NULL;
-        t->mReformatBufferProvider = NULL;
-        t->downmixerBufferProvider = NULL;
-        t->mPostDownmixReformatBufferProvider = NULL;
-        t->mTimestretchBufferProvider = NULL;
         t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
         t->mFormat = format;
         t->mMixerInFormat = selectMixerInFormat(format);
@@ -230,96 +166,83 @@
         status_t status = t->prepareForDownmix();
         if (status != OK) {
             ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
-            return -1;
+            return BAD_VALUE;
         }
         // prepareForDownmix() may change mDownmixRequiresFormat
         ALOGVV("mMixerFormat:%#x  mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
         t->prepareForReformat();
-        mTrackNames |= 1 << n;
-        return TRACK0 + n;
-    }
-    ALOGE("AudioMixer::getTrackName out of available tracks");
-    return -1;
-}
 
-void AudioMixer::invalidateState(uint32_t mask)
-{
-    if (mask != 0) {
-        mState.needsChanged |= mask;
-        mState.hook = process__validate;
+        mTracks[name] = t;
+        return OK;
     }
- }
+}
 
 // Called when channel masks have changed for a track name
 // TODO: Fix DownmixerBufferProvider not to (possibly) change mixer input format,
 // which will simplify this logic.
 bool AudioMixer::setChannelMasks(int name,
         audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
-    track_t &track = mState.tracks[name];
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<Track> &track = mTracks[name];
 
-    if (trackChannelMask == track.channelMask
-            && mixerChannelMask == track.mMixerChannelMask) {
+    if (trackChannelMask == track->channelMask
+            && mixerChannelMask == track->mMixerChannelMask) {
         return false;  // no need to change
     }
     // always recompute for both channel masks even if only one has changed.
     const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
     const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
-    const bool mixerChannelCountChanged = track.mMixerChannelCount != mixerChannelCount;
 
     ALOG_ASSERT((trackChannelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX)
             && trackChannelCount
             && mixerChannelCount);
-    track.channelMask = trackChannelMask;
-    track.channelCount = trackChannelCount;
-    track.mMixerChannelMask = mixerChannelMask;
-    track.mMixerChannelCount = mixerChannelCount;
+    track->channelMask = trackChannelMask;
+    track->channelCount = trackChannelCount;
+    track->mMixerChannelMask = mixerChannelMask;
+    track->mMixerChannelCount = mixerChannelCount;
 
     // channel masks have changed, does this track need a downmixer?
     // update to try using our desired format (if we aren't already using it)
-    const audio_format_t prevDownmixerFormat = track.mDownmixRequiresFormat;
-    const status_t status = mState.tracks[name].prepareForDownmix();
+    const status_t status = track->prepareForDownmix();
     ALOGE_IF(status != OK,
             "prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x",
-            status, track.channelMask, track.mMixerChannelMask);
+            status, track->channelMask, track->mMixerChannelMask);
 
-    if (prevDownmixerFormat != track.mDownmixRequiresFormat) {
-        track.prepareForReformat(); // because of downmixer, track format may change!
-    }
+    // always do reformat since channel mask changed,
+    // do it after downmix since track format may change!
+    track->prepareForReformat();
 
-    if (track.resampler && mixerChannelCountChanged) {
+    if (track->mResampler.get() != nullptr) {
         // resampler channels may have changed.
-        const uint32_t resetToSampleRate = track.sampleRate;
-        delete track.resampler;
-        track.resampler = NULL;
-        track.sampleRate = mSampleRate; // without resampler, track rate is device sample rate.
+        const uint32_t resetToSampleRate = track->sampleRate;
+        track->mResampler.reset(nullptr);
+        track->sampleRate = mSampleRate; // without resampler, track rate is device sample rate.
         // recreate the resampler with updated format, channels, saved sampleRate.
-        track.setResampler(resetToSampleRate /*trackSampleRate*/, mSampleRate /*devSampleRate*/);
+        track->setResampler(resetToSampleRate /*trackSampleRate*/, mSampleRate /*devSampleRate*/);
     }
     return true;
 }
 
-void AudioMixer::track_t::unprepareForDownmix() {
+void AudioMixer::Track::unprepareForDownmix() {
     ALOGV("AudioMixer::unprepareForDownmix(%p)", this);
 
-    if (mPostDownmixReformatBufferProvider != nullptr) {
+    if (mPostDownmixReformatBufferProvider.get() != nullptr) {
         // release any buffers held by the mPostDownmixReformatBufferProvider
-        // before deallocating the downmixerBufferProvider.
+        // before deallocating the mDownmixerBufferProvider.
         mPostDownmixReformatBufferProvider->reset();
     }
 
     mDownmixRequiresFormat = AUDIO_FORMAT_INVALID;
-    if (downmixerBufferProvider != NULL) {
+    if (mDownmixerBufferProvider.get() != nullptr) {
         // this track had previously been configured with a downmixer, delete it
-        ALOGV(" deleting old downmixer");
-        delete downmixerBufferProvider;
-        downmixerBufferProvider = NULL;
+        mDownmixerBufferProvider.reset(nullptr);
         reconfigureBufferProviders();
     } else {
         ALOGV(" nothing to do, no downmixer to delete");
     }
 }
 
-status_t AudioMixer::track_t::prepareForDownmix()
+status_t AudioMixer::Track::prepareForDownmix()
 {
     ALOGV("AudioMixer::prepareForDownmix(%p) with mask 0x%x",
             this, channelMask);
@@ -337,40 +260,35 @@
     if (audio_channel_mask_get_representation(channelMask)
                 == AUDIO_CHANNEL_REPRESENTATION_POSITION
             && DownmixerBufferProvider::isMultichannelCapable()) {
-        DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(channelMask,
+        mDownmixerBufferProvider.reset(new DownmixerBufferProvider(channelMask,
                 mMixerChannelMask,
                 AUDIO_FORMAT_PCM_16_BIT /* TODO: use mMixerInFormat, now only PCM 16 */,
-                sampleRate, sessionId, kCopyBufferFrameCount);
-
-        if (pDbp->isValid()) { // if constructor completed properly
+                sampleRate, sessionId, kCopyBufferFrameCount));
+        if (static_cast<DownmixerBufferProvider *>(mDownmixerBufferProvider.get())->isValid()) {
             mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
-            downmixerBufferProvider = pDbp;
             reconfigureBufferProviders();
             return NO_ERROR;
         }
-        delete pDbp;
+        // mDownmixerBufferProvider reset below.
     }
 
     // Effect downmixer does not accept the channel conversion.  Let's use our remixer.
-    RemixBufferProvider* pRbp = new RemixBufferProvider(channelMask,
-            mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount);
+    mDownmixerBufferProvider.reset(new RemixBufferProvider(channelMask,
+            mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount));
     // Remix always finds a conversion whereas Downmixer effect above may fail.
-    downmixerBufferProvider = pRbp;
     reconfigureBufferProviders();
     return NO_ERROR;
 }
 
-void AudioMixer::track_t::unprepareForReformat() {
+void AudioMixer::Track::unprepareForReformat() {
     ALOGV("AudioMixer::unprepareForReformat(%p)", this);
     bool requiresReconfigure = false;
-    if (mReformatBufferProvider != NULL) {
-        delete mReformatBufferProvider;
-        mReformatBufferProvider = NULL;
+    if (mReformatBufferProvider.get() != nullptr) {
+        mReformatBufferProvider.reset(nullptr);
         requiresReconfigure = true;
     }
-    if (mPostDownmixReformatBufferProvider != NULL) {
-        delete mPostDownmixReformatBufferProvider;
-        mPostDownmixReformatBufferProvider = NULL;
+    if (mPostDownmixReformatBufferProvider.get() != nullptr) {
+        mPostDownmixReformatBufferProvider.reset(nullptr);
         requiresReconfigure = true;
     }
     if (requiresReconfigure) {
@@ -378,7 +296,7 @@
     }
 }
 
-status_t AudioMixer::track_t::prepareForReformat()
+status_t AudioMixer::Track::prepareForReformat()
 {
     ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat);
     // discard previous reformatters
@@ -388,19 +306,27 @@
             ? mDownmixRequiresFormat : mMixerInFormat;
     bool requiresReconfigure = false;
     if (mFormat != targetFormat) {
-        mReformatBufferProvider = new ReformatBufferProvider(
+        mReformatBufferProvider.reset(new ReformatBufferProvider(
                 audio_channel_count_from_out_mask(channelMask),
                 mFormat,
                 targetFormat,
-                kCopyBufferFrameCount);
+                kCopyBufferFrameCount));
+        requiresReconfigure = true;
+    } else if (mFormat == AUDIO_FORMAT_PCM_FLOAT) {
+        // Input and output are floats, make sure application did not provide > 3db samples
+        // that would break volume application (b/68099072)
+        // TODO: add a trusted source flag to avoid the overhead
+        mReformatBufferProvider.reset(new ClampFloatBufferProvider(
+                audio_channel_count_from_out_mask(channelMask),
+                kCopyBufferFrameCount));
         requiresReconfigure = true;
     }
     if (targetFormat != mMixerInFormat) {
-        mPostDownmixReformatBufferProvider = new ReformatBufferProvider(
+        mPostDownmixReformatBufferProvider.reset(new ReformatBufferProvider(
                 audio_channel_count_from_out_mask(mMixerChannelMask),
                 targetFormat,
                 mMixerInFormat,
-                kCopyBufferFrameCount);
+                kCopyBufferFrameCount));
         requiresReconfigure = true;
     }
     if (requiresReconfigure) {
@@ -409,74 +335,59 @@
     return NO_ERROR;
 }
 
-void AudioMixer::track_t::reconfigureBufferProviders()
+void AudioMixer::Track::reconfigureBufferProviders()
 {
     bufferProvider = mInputBufferProvider;
-    if (mReformatBufferProvider) {
+    if (mReformatBufferProvider.get() != nullptr) {
         mReformatBufferProvider->setBufferProvider(bufferProvider);
-        bufferProvider = mReformatBufferProvider;
+        bufferProvider = mReformatBufferProvider.get();
     }
-    if (downmixerBufferProvider) {
-        downmixerBufferProvider->setBufferProvider(bufferProvider);
-        bufferProvider = downmixerBufferProvider;
+    if (mDownmixerBufferProvider.get() != nullptr) {
+        mDownmixerBufferProvider->setBufferProvider(bufferProvider);
+        bufferProvider = mDownmixerBufferProvider.get();
     }
-    if (mPostDownmixReformatBufferProvider) {
+    if (mPostDownmixReformatBufferProvider.get() != nullptr) {
         mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider);
-        bufferProvider = mPostDownmixReformatBufferProvider;
+        bufferProvider = mPostDownmixReformatBufferProvider.get();
     }
-    if (mTimestretchBufferProvider) {
+    if (mTimestretchBufferProvider.get() != nullptr) {
         mTimestretchBufferProvider->setBufferProvider(bufferProvider);
-        bufferProvider = mTimestretchBufferProvider;
+        bufferProvider = mTimestretchBufferProvider.get();
     }
 }
 
-void AudioMixer::deleteTrackName(int name)
+void AudioMixer::destroy(int name)
 {
-    ALOGV("AudioMixer::deleteTrackName(%d)", name);
-    name -= TRACK0;
-    LOG_ALWAYS_FATAL_IF(name < 0 || name >= (int)MAX_NUM_TRACKS, "bad track name %d", name);
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
     ALOGV("deleteTrackName(%d)", name);
-    track_t& track(mState.tracks[ name ]);
-    if (track.enabled) {
-        track.enabled = false;
-        invalidateState(1<<name);
+
+    if (mTracks[name]->enabled) {
+        invalidate();
     }
-    // delete the resampler
-    delete track.resampler;
-    track.resampler = NULL;
-    // delete the downmixer
-    mState.tracks[name].unprepareForDownmix();
-    // delete the reformatter
-    mState.tracks[name].unprepareForReformat();
-    // delete the timestretch provider
-    delete track.mTimestretchBufferProvider;
-    track.mTimestretchBufferProvider = NULL;
-    mTrackNames &= ~(1<<name);
+    mTracks.erase(name); // deallocate track
 }
 
 void AudioMixer::enable(int name)
 {
-    name -= TRACK0;
-    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
-    track_t& track = mState.tracks[name];
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<Track> &track = mTracks[name];
 
-    if (!track.enabled) {
-        track.enabled = true;
+    if (!track->enabled) {
+        track->enabled = true;
         ALOGV("enable(%d)", name);
-        invalidateState(1 << name);
+        invalidate();
     }
 }
 
 void AudioMixer::disable(int name)
 {
-    name -= TRACK0;
-    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
-    track_t& track = mState.tracks[name];
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<Track> &track = mTracks[name];
 
-    if (track.enabled) {
-        track.enabled = false;
+    if (track->enabled) {
+        track->enabled = false;
         ALOGV("disable(%d)", name);
-        invalidateState(1 << name);
+        invalidate();
     }
 }
 
@@ -554,7 +465,8 @@
         ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
                 " prev:%f  set_to:%f", *pPrevVolume, *pSetVolume);
         const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
-        const float maxv = max(newVolume, *pPrevVolume); // could be inf, cannot be nan, subnormal
+        // could be inf, cannot be nan, subnormal
+        const float maxv = std::max(newVolume, *pPrevVolume);
 
         if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
                 && maxv + inc != maxv) { // inc must make forward progress
@@ -607,9 +519,8 @@
 
 void AudioMixer::setParameter(int name, int target, int param, void *value)
 {
-    name -= TRACK0;
-    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
-    track_t& track = mState.tracks[name];
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<Track> &track = mTracks[name];
 
     int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
     int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
@@ -621,33 +532,33 @@
         case CHANNEL_MASK: {
             const audio_channel_mask_t trackChannelMask =
                 static_cast<audio_channel_mask_t>(valueInt);
-            if (setChannelMasks(name, trackChannelMask, track.mMixerChannelMask)) {
+            if (setChannelMasks(name, trackChannelMask, track->mMixerChannelMask)) {
                 ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
-                invalidateState(1 << name);
+                invalidate();
             }
             } break;
         case MAIN_BUFFER:
-            if (track.mainBuffer != valueBuf) {
-                track.mainBuffer = valueBuf;
+            if (track->mainBuffer != valueBuf) {
+                track->mainBuffer = valueBuf;
                 ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
-                invalidateState(1 << name);
+                invalidate();
             }
             break;
         case AUX_BUFFER:
-            if (track.auxBuffer != valueBuf) {
-                track.auxBuffer = valueBuf;
+            if (track->auxBuffer != valueBuf) {
+                track->auxBuffer = valueBuf;
                 ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
-                invalidateState(1 << name);
+                invalidate();
             }
             break;
         case FORMAT: {
             audio_format_t format = static_cast<audio_format_t>(valueInt);
-            if (track.mFormat != format) {
+            if (track->mFormat != format) {
                 ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
-                track.mFormat = format;
+                track->mFormat = format;
                 ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
-                track.prepareForReformat();
-                invalidateState(1 << name);
+                track->prepareForReformat();
+                invalidate();
             }
             } break;
         // FIXME do we want to support setting the downmix type from AudioFlinger?
@@ -656,17 +567,17 @@
             break          */
         case MIXER_FORMAT: {
             audio_format_t format = static_cast<audio_format_t>(valueInt);
-            if (track.mMixerFormat != format) {
-                track.mMixerFormat = format;
+            if (track->mMixerFormat != format) {
+                track->mMixerFormat = format;
                 ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
             }
             } break;
         case MIXER_CHANNEL_MASK: {
             const audio_channel_mask_t mixerChannelMask =
                     static_cast<audio_channel_mask_t>(valueInt);
-            if (setChannelMasks(name, track.channelMask, mixerChannelMask)) {
+            if (setChannelMasks(name, track->channelMask, mixerChannelMask)) {
                 ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
-                invalidateState(1 << name);
+                invalidate();
             }
             } break;
         default:
@@ -678,21 +589,20 @@
         switch (param) {
         case SAMPLE_RATE:
             ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
-            if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
+            if (track->setResampler(uint32_t(valueInt), mSampleRate)) {
                 ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
                         uint32_t(valueInt));
-                invalidateState(1 << name);
+                invalidate();
             }
             break;
         case RESET:
-            track.resetResampler();
-            invalidateState(1 << name);
+            track->resetResampler();
+            invalidate();
             break;
         case REMOVE:
-            delete track.resampler;
-            track.resampler = NULL;
-            track.sampleRate = mSampleRate;
-            invalidateState(1 << name);
+            track->mResampler.reset(nullptr);
+            track->sampleRate = mSampleRate;
+            invalidate();
             break;
         default:
             LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
@@ -704,26 +614,28 @@
         switch (param) {
         case AUXLEVEL:
             if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
-                    target == RAMP_VOLUME ? mState.frameCount : 0,
-                    &track.auxLevel, &track.prevAuxLevel, &track.auxInc,
-                    &track.mAuxLevel, &track.mPrevAuxLevel, &track.mAuxInc)) {
+                    target == RAMP_VOLUME ? mFrameCount : 0,
+                    &track->auxLevel, &track->prevAuxLevel, &track->auxInc,
+                    &track->mAuxLevel, &track->mPrevAuxLevel, &track->mAuxInc)) {
                 ALOGV("setParameter(%s, AUXLEVEL: %04x)",
-                        target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track.auxLevel);
-                invalidateState(1 << name);
+                        target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track->auxLevel);
+                invalidate();
             }
             break;
         default:
             if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
                 if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
-                        target == RAMP_VOLUME ? mState.frameCount : 0,
-                        &track.volume[param - VOLUME0], &track.prevVolume[param - VOLUME0],
-                        &track.volumeInc[param - VOLUME0],
-                        &track.mVolume[param - VOLUME0], &track.mPrevVolume[param - VOLUME0],
-                        &track.mVolumeInc[param - VOLUME0])) {
+                        target == RAMP_VOLUME ? mFrameCount : 0,
+                        &track->volume[param - VOLUME0],
+                        &track->prevVolume[param - VOLUME0],
+                        &track->volumeInc[param - VOLUME0],
+                        &track->mVolume[param - VOLUME0],
+                        &track->mPrevVolume[param - VOLUME0],
+                        &track->mVolumeInc[param - VOLUME0])) {
                     ALOGV("setParameter(%s, VOLUME%d: %04x)",
                             target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
-                                    track.volume[param - VOLUME0]);
-                    invalidateState(1 << name);
+                                    track->volume[param - VOLUME0]);
+                    invalidate();
                 }
             } else {
                 LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
@@ -736,16 +648,16 @@
                 const AudioPlaybackRate *playbackRate =
                         reinterpret_cast<AudioPlaybackRate*>(value);
                 ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
-                        "bad parameters speed %f, pitch %f",playbackRate->mSpeed,
-                        playbackRate->mPitch);
-                if (track.setPlaybackRate(*playbackRate)) {
+                        "bad parameters speed %f, pitch %f",
+                        playbackRate->mSpeed, playbackRate->mPitch);
+                if (track->setPlaybackRate(*playbackRate)) {
                     ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
                             "%f %f %d %d",
                             playbackRate->mSpeed,
                             playbackRate->mPitch,
                             playbackRate->mStretchMode,
                             playbackRate->mFallbackMode);
-                    // invalidateState(1 << name);
+                    // invalidate();  (should not require reconfigure)
                 }
             } break;
             default:
@@ -758,12 +670,12 @@
     }
 }
 
-bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
+bool AudioMixer::Track::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
 {
-    if (trackSampleRate != devSampleRate || resampler != NULL) {
+    if (trackSampleRate != devSampleRate || mResampler.get() != nullptr) {
         if (sampleRate != trackSampleRate) {
             sampleRate = trackSampleRate;
-            if (resampler == NULL) {
+            if (mResampler.get() == nullptr) {
                 ALOGV("Creating resampler from track %d Hz to device %d Hz",
                         trackSampleRate, devSampleRate);
                 AudioResampler::src_quality quality;
@@ -779,15 +691,15 @@
 
                 // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
                 // but if none exists, it is the channel count (1 for mono).
-                const int resamplerChannelCount = downmixerBufferProvider != NULL
+                const int resamplerChannelCount = mDownmixerBufferProvider.get() != nullptr
                         ? mMixerChannelCount : channelCount;
                 ALOGVV("Creating resampler:"
                         " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
                         mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
-                resampler = AudioResampler::create(
+                mResampler.reset(AudioResampler::create(
                         mMixerInFormat,
                         resamplerChannelCount,
-                        devSampleRate, quality);
+                        devSampleRate, quality));
             }
             return true;
         }
@@ -795,25 +707,25 @@
     return false;
 }
 
-bool AudioMixer::track_t::setPlaybackRate(const AudioPlaybackRate &playbackRate)
+bool AudioMixer::Track::setPlaybackRate(const AudioPlaybackRate &playbackRate)
 {
-    if ((mTimestretchBufferProvider == NULL &&
+    if ((mTimestretchBufferProvider.get() == nullptr &&
             fabs(playbackRate.mSpeed - mPlaybackRate.mSpeed) < AUDIO_TIMESTRETCH_SPEED_MIN_DELTA &&
             fabs(playbackRate.mPitch - mPlaybackRate.mPitch) < AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) ||
             isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
         return false;
     }
     mPlaybackRate = playbackRate;
-    if (mTimestretchBufferProvider == NULL) {
+    if (mTimestretchBufferProvider.get() == nullptr) {
         // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
         // but if none exists, it is the channel count (1 for mono).
-        const int timestretchChannelCount = downmixerBufferProvider != NULL
+        const int timestretchChannelCount = mDownmixerBufferProvider.get() != nullptr
                 ? mMixerChannelCount : channelCount;
-        mTimestretchBufferProvider = new TimestretchBufferProvider(timestretchChannelCount,
-                mMixerInFormat, sampleRate, playbackRate);
+        mTimestretchBufferProvider.reset(new TimestretchBufferProvider(timestretchChannelCount,
+                mMixerInFormat, sampleRate, playbackRate));
         reconfigureBufferProviders();
     } else {
-        reinterpret_cast<TimestretchBufferProvider*>(mTimestretchBufferProvider)
+        static_cast<TimestretchBufferProvider*>(mTimestretchBufferProvider.get())
                 ->setPlaybackRate(playbackRate);
     }
     return true;
@@ -832,7 +744,7 @@
  *
  * There is a bit of duplicated code here, but it keeps backward compatibility.
  */
-inline void AudioMixer::track_t::adjustVolumeRamp(bool aux, bool useFloat)
+inline void AudioMixer::Track::adjustVolumeRamp(bool aux, bool useFloat)
 {
     if (useFloat) {
         for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
@@ -861,113 +773,98 @@
             }
         }
     }
-    /* TODO: aux is always integer regardless of output buffer type */
+
     if (aux) {
-        if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) ||
-                ((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {
+#ifdef FLOAT_AUX
+        if (useFloat) {
+            if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||
+                    (mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
+                auxInc = 0;
+                prevAuxLevel = auxLevel << 16;
+                mAuxInc = 0.f;
+                mPrevAuxLevel = mAuxLevel;
+            }
+        } else
+#endif
+        if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||
+                (auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
             auxInc = 0;
             prevAuxLevel = auxLevel << 16;
-            mAuxInc = 0.;
+            mAuxInc = 0.f;
             mPrevAuxLevel = mAuxLevel;
-        } else {
-            //ALOGV("aux ramp: %d %d %d", auxLevel << 16, prevAuxLevel, auxInc);
         }
     }
 }
 
 size_t AudioMixer::getUnreleasedFrames(int name) const
 {
-    name -= TRACK0;
-    if (uint32_t(name) < MAX_NUM_TRACKS) {
-        return mState.tracks[name].getUnreleasedFrames();
+    const auto it = mTracks.find(name);
+    if (it != mTracks.end()) {
+        return it->second->getUnreleasedFrames();
     }
     return 0;
 }
 
 void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
 {
-    name -= TRACK0;
-    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<Track> &track = mTracks[name];
 
-    if (mState.tracks[name].mInputBufferProvider == bufferProvider) {
+    if (track->mInputBufferProvider == bufferProvider) {
         return; // don't reset any buffer providers if identical.
     }
-    if (mState.tracks[name].mReformatBufferProvider != NULL) {
-        mState.tracks[name].mReformatBufferProvider->reset();
-    } else if (mState.tracks[name].downmixerBufferProvider != NULL) {
-        mState.tracks[name].downmixerBufferProvider->reset();
-    } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) {
-        mState.tracks[name].mPostDownmixReformatBufferProvider->reset();
-    } else if (mState.tracks[name].mTimestretchBufferProvider != NULL) {
-        mState.tracks[name].mTimestretchBufferProvider->reset();
+    if (track->mReformatBufferProvider.get() != nullptr) {
+        track->mReformatBufferProvider->reset();
+    } else if (track->mDownmixerBufferProvider != nullptr) {
+        track->mDownmixerBufferProvider->reset();
+    } else if (track->mPostDownmixReformatBufferProvider.get() != nullptr) {
+        track->mPostDownmixReformatBufferProvider->reset();
+    } else if (track->mTimestretchBufferProvider.get() != nullptr) {
+        track->mTimestretchBufferProvider->reset();
     }
 
-    mState.tracks[name].mInputBufferProvider = bufferProvider;
-    mState.tracks[name].reconfigureBufferProviders();
+    track->mInputBufferProvider = bufferProvider;
+    track->reconfigureBufferProviders();
 }
 
-
-void AudioMixer::process()
+void AudioMixer::process__validate()
 {
-    mState.hook(&mState);
-}
-
-
-void AudioMixer::process__validate(state_t* state)
-{
-    ALOGW_IF(!state->needsChanged,
-        "in process__validate() but nothing's invalid");
-
-    uint32_t changed = state->needsChanged;
-    state->needsChanged = 0; // clear the validation flag
-
-    // recompute which tracks are enabled / disabled
-    uint32_t enabled = 0;
-    uint32_t disabled = 0;
-    while (changed) {
-        const int i = 31 - __builtin_clz(changed);
-        const uint32_t mask = 1<<i;
-        changed &= ~mask;
-        track_t& t = state->tracks[i];
-        (t.enabled ? enabled : disabled) |= mask;
-    }
-    state->enabledTracks &= ~disabled;
-    state->enabledTracks |=  enabled;
-
-    // compute everything we need...
-    int countActiveTracks = 0;
     // TODO: fix all16BitsStereNoResample logic to
     // either properly handle muted tracks (it should ignore them)
     // or remove altogether as an obsolete optimization.
     bool all16BitsStereoNoResample = true;
     bool resampling = false;
     bool volumeRamp = false;
-    uint32_t en = state->enabledTracks;
-    while (en) {
-        const int i = 31 - __builtin_clz(en);
-        en &= ~(1<<i);
 
-        countActiveTracks++;
-        track_t& t = state->tracks[i];
+    mEnabled.clear();
+    mGroups.clear();
+    for (const auto &pair : mTracks) {
+        const int name = pair.first;
+        const std::shared_ptr<Track> &t = pair.second;
+        if (!t->enabled) continue;
+
+        mEnabled.emplace_back(name);  // we add to mEnabled in order of name.
+        mGroups[t->mainBuffer].emplace_back(name); // mGroups also in order of name.
+
         uint32_t n = 0;
         // FIXME can overflow (mask is only 3 bits)
-        n |= NEEDS_CHANNEL_1 + t.channelCount - 1;
-        if (t.doesResample()) {
+        n |= NEEDS_CHANNEL_1 + t->channelCount - 1;
+        if (t->doesResample()) {
             n |= NEEDS_RESAMPLE;
         }
-        if (t.auxLevel != 0 && t.auxBuffer != NULL) {
+        if (t->auxLevel != 0 && t->auxBuffer != NULL) {
             n |= NEEDS_AUX;
         }
 
-        if (t.volumeInc[0]|t.volumeInc[1]) {
+        if (t->volumeInc[0]|t->volumeInc[1]) {
             volumeRamp = true;
-        } else if (!t.doesResample() && t.volumeRL == 0) {
+        } else if (!t->doesResample() && t->volumeRL == 0) {
             n |= NEEDS_MUTE;
         }
-        t.needs = n;
+        t->needs = n;
 
         if (n & NEEDS_MUTE) {
-            t.hook = track__nop;
+            t->hook = &Track::track__nop;
         } else {
             if (n & NEEDS_AUX) {
                 all16BitsStereoNoResample = false;
@@ -975,23 +872,23 @@
             if (n & NEEDS_RESAMPLE) {
                 all16BitsStereoNoResample = false;
                 resampling = true;
-                t.hook = getTrackHook(TRACKTYPE_RESAMPLE, t.mMixerChannelCount,
-                        t.mMixerInFormat, t.mMixerFormat);
+                t->hook = Track::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
+                        t->mMixerInFormat, t->mMixerFormat);
                 ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
                         "Track %d needs downmix + resample", i);
             } else {
                 if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
-                    t.hook = getTrackHook(
-                            (t.mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO  // TODO: MONO_HACK
-                                    && t.channelMask == AUDIO_CHANNEL_OUT_MONO)
+                    t->hook = Track::getTrackHook(
+                            (t->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO  // TODO: MONO_HACK
+                                    && t->channelMask == AUDIO_CHANNEL_OUT_MONO)
                                 ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
-                            t.mMixerChannelCount,
-                            t.mMixerInFormat, t.mMixerFormat);
+                            t->mMixerChannelCount,
+                            t->mMixerInFormat, t->mMixerFormat);
                     all16BitsStereoNoResample = false;
                 }
                 if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
-                    t.hook = getTrackHook(TRACKTYPE_NORESAMPLE, t.mMixerChannelCount,
-                            t.mMixerInFormat, t.mMixerFormat);
+                    t->hook = Track::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
+                            t->mMixerInFormat, t->mMixerFormat);
                     ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
                             "Track %d needs downmix", i);
                 }
@@ -1000,137 +897,125 @@
     }
 
     // select the processing hooks
-    state->hook = process__nop;
-    if (countActiveTracks > 0) {
+    mHook = &AudioMixer::process__nop;
+    if (mEnabled.size() > 0) {
         if (resampling) {
-            if (!state->outputTemp) {
-                state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
+            if (mOutputTemp.get() == nullptr) {
+                mOutputTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
             }
-            if (!state->resampleTemp) {
-                state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount];
+            if (mResampleTemp.get() == nullptr) {
+                mResampleTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
             }
-            state->hook = process__genericResampling;
+            mHook = &AudioMixer::process__genericResampling;
         } else {
-            if (state->outputTemp) {
-                delete [] state->outputTemp;
-                state->outputTemp = NULL;
-            }
-            if (state->resampleTemp) {
-                delete [] state->resampleTemp;
-                state->resampleTemp = NULL;
-            }
-            state->hook = process__genericNoResampling;
+            // we keep temp arrays around.
+            mHook = &AudioMixer::process__genericNoResampling;
             if (all16BitsStereoNoResample && !volumeRamp) {
-                if (countActiveTracks == 1) {
-                    const int i = 31 - __builtin_clz(state->enabledTracks);
-                    track_t& t = state->tracks[i];
-                    if ((t.needs & NEEDS_MUTE) == 0) {
+                if (mEnabled.size() == 1) {
+                    const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
+                    if ((t->needs & NEEDS_MUTE) == 0) {
                         // The check prevents a muted track from acquiring a process hook.
                         //
                         // This is dangerous if the track is MONO as that requires
                         // special case handling due to implicit channel duplication.
                         // Stereo or Multichannel should actually be fine here.
-                        state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
-                                t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
+                        mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+                                t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
                     }
                 }
             }
         }
     }
 
-    ALOGV("mixer configuration change: %d activeTracks (%08x) "
+    ALOGV("mixer configuration change: %zu "
         "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
-        countActiveTracks, state->enabledTracks,
-        all16BitsStereoNoResample, resampling, volumeRamp);
+        mEnabled.size(), all16BitsStereoNoResample, resampling, volumeRamp);
 
-   state->hook(state);
+   process();
 
     // Now that the volume ramp has been done, set optimal state and
     // track hooks for subsequent mixer process
-    if (countActiveTracks > 0) {
+    if (mEnabled.size() > 0) {
         bool allMuted = true;
-        uint32_t en = state->enabledTracks;
-        while (en) {
-            const int i = 31 - __builtin_clz(en);
-            en &= ~(1<<i);
-            track_t& t = state->tracks[i];
-            if (!t.doesResample() && t.volumeRL == 0) {
-                t.needs |= NEEDS_MUTE;
-                t.hook = track__nop;
+
+        for (const int name : mEnabled) {
+            const std::shared_ptr<Track> &t = mTracks[name];
+            if (!t->doesResample() && t->volumeRL == 0) {
+                t->needs |= NEEDS_MUTE;
+                t->hook = &Track::track__nop;
             } else {
                 allMuted = false;
             }
         }
         if (allMuted) {
-            state->hook = process__nop;
+            mHook = &AudioMixer::process__nop;
         } else if (all16BitsStereoNoResample) {
-            if (countActiveTracks == 1) {
-                const int i = 31 - __builtin_clz(state->enabledTracks);
-                track_t& t = state->tracks[i];
+            if (mEnabled.size() == 1) {
+                //const int i = 31 - __builtin_clz(enabledTracks);
+                const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
                 // Muted single tracks handled by allMuted above.
-                state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
-                        t.mMixerChannelCount, t.mMixerInFormat, t.mMixerFormat);
+                mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+                        t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
             }
         }
     }
 }
 
-
-void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount,
-        int32_t* temp, int32_t* aux)
+void AudioMixer::Track::track__genericResample(
+        int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
 {
     ALOGVV("track__genericResample\n");
-    t->resampler->setSampleRate(t->sampleRate);
+    mResampler->setSampleRate(sampleRate);
 
     // ramp gain - resample to temp buffer and scale/mix in 2nd step
     if (aux != NULL) {
         // always resample with unity gain when sending to auxiliary buffer to be able
         // to apply send level after resampling
-        t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
-        memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(int32_t));
-        t->resampler->resample(temp, outFrameCount, t->bufferProvider);
-        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
-            volumeRampStereo(t, out, outFrameCount, temp, aux);
+        mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+        memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(int32_t));
+        mResampler->resample(temp, outFrameCount, bufferProvider);
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+            volumeRampStereo(out, outFrameCount, temp, aux);
         } else {
-            volumeStereo(t, out, outFrameCount, temp, aux);
+            volumeStereo(out, outFrameCount, temp, aux);
         }
     } else {
-        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
-            t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+            mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
             memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
-            t->resampler->resample(temp, outFrameCount, t->bufferProvider);
-            volumeRampStereo(t, out, outFrameCount, temp, aux);
+            mResampler->resample(temp, outFrameCount, bufferProvider);
+            volumeRampStereo(out, outFrameCount, temp, aux);
         }
 
         // constant gain
         else {
-            t->resampler->setVolume(t->mVolume[0], t->mVolume[1]);
-            t->resampler->resample(out, outFrameCount, t->bufferProvider);
+            mResampler->setVolume(mVolume[0], mVolume[1]);
+            mResampler->resample(out, outFrameCount, bufferProvider);
         }
     }
 }
 
-void AudioMixer::track__nop(track_t* t __unused, int32_t* out __unused,
+void AudioMixer::Track::track__nop(int32_t* out __unused,
         size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
 {
 }
 
-void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
-        int32_t* aux)
+void AudioMixer::Track::volumeRampStereo(
+        int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
 {
-    int32_t vl = t->prevVolume[0];
-    int32_t vr = t->prevVolume[1];
-    const int32_t vlInc = t->volumeInc[0];
-    const int32_t vrInc = t->volumeInc[1];
+    int32_t vl = prevVolume[0];
+    int32_t vr = prevVolume[1];
+    const int32_t vlInc = volumeInc[0];
+    const int32_t vrInc = volumeInc[1];
 
     //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-    //        t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+    //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
     //       (vl + vlInc*frameCount)/65536.0f, frameCount);
 
     // ramp volume
     if (CC_UNLIKELY(aux != NULL)) {
-        int32_t va = t->prevAuxLevel;
-        const int32_t vaInc = t->auxInc;
+        int32_t va = prevAuxLevel;
+        const int32_t vaInc = auxInc;
         int32_t l;
         int32_t r;
 
@@ -1144,7 +1029,7 @@
             vr += vrInc;
             va += vaInc;
         } while (--frameCount);
-        t->prevAuxLevel = va;
+        prevAuxLevel = va;
     } else {
         do {
             *out++ += (vl >> 16) * (*temp++ >> 12);
@@ -1153,19 +1038,19 @@
             vr += vrInc;
         } while (--frameCount);
     }
-    t->prevVolume[0] = vl;
-    t->prevVolume[1] = vr;
-    t->adjustVolumeRamp(aux != NULL);
+    prevVolume[0] = vl;
+    prevVolume[1] = vr;
+    adjustVolumeRamp(aux != NULL);
 }
 
-void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp,
-        int32_t* aux)
+void AudioMixer::Track::volumeStereo(
+        int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
 {
-    const int16_t vl = t->volume[0];
-    const int16_t vr = t->volume[1];
+    const int16_t vl = volume[0];
+    const int16_t vr = volume[1];
 
     if (CC_UNLIKELY(aux != NULL)) {
-        const int16_t va = t->auxLevel;
+        const int16_t va = auxLevel;
         do {
             int16_t l = (int16_t)(*temp++ >> 12);
             int16_t r = (int16_t)(*temp++ >> 12);
@@ -1187,25 +1072,25 @@
     }
 }
 
-void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount,
-        int32_t* temp __unused, int32_t* aux)
+void AudioMixer::Track::track__16BitsStereo(
+        int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
 {
     ALOGVV("track__16BitsStereo\n");
-    const int16_t *in = static_cast<const int16_t *>(t->in);
+    const int16_t *in = static_cast<const int16_t *>(mIn);
 
     if (CC_UNLIKELY(aux != NULL)) {
         int32_t l;
         int32_t r;
         // ramp gain
-        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
-            int32_t vl = t->prevVolume[0];
-            int32_t vr = t->prevVolume[1];
-            int32_t va = t->prevAuxLevel;
-            const int32_t vlInc = t->volumeInc[0];
-            const int32_t vrInc = t->volumeInc[1];
-            const int32_t vaInc = t->auxInc;
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            int32_t va = prevAuxLevel;
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
+            const int32_t vaInc = auxInc;
             // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //        t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+            //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
             //        (vl + vlInc*frameCount)/65536.0f, frameCount);
 
             do {
@@ -1219,16 +1104,16 @@
                 va += vaInc;
             } while (--frameCount);
 
-            t->prevVolume[0] = vl;
-            t->prevVolume[1] = vr;
-            t->prevAuxLevel = va;
-            t->adjustVolumeRamp(true);
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            prevAuxLevel = va;
+            adjustVolumeRamp(true);
         }
 
         // constant gain
         else {
-            const uint32_t vrl = t->volumeRL;
-            const int16_t va = (int16_t)t->auxLevel;
+            const uint32_t vrl = volumeRL;
+            const int16_t va = (int16_t)auxLevel;
             do {
                 uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
                 int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
@@ -1242,14 +1127,14 @@
         }
     } else {
         // ramp gain
-        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
-            int32_t vl = t->prevVolume[0];
-            int32_t vr = t->prevVolume[1];
-            const int32_t vlInc = t->volumeInc[0];
-            const int32_t vrInc = t->volumeInc[1];
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
 
             // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //        t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+            //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
             //        (vl + vlInc*frameCount)/65536.0f, frameCount);
 
             do {
@@ -1259,14 +1144,14 @@
                 vr += vrInc;
             } while (--frameCount);
 
-            t->prevVolume[0] = vl;
-            t->prevVolume[1] = vr;
-            t->adjustVolumeRamp(false);
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            adjustVolumeRamp(false);
         }
 
         // constant gain
         else {
-            const uint32_t vrl = t->volumeRL;
+            const uint32_t vrl = volumeRL;
             do {
                 uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
                 in += 2;
@@ -1276,27 +1161,27 @@
             } while (--frameCount);
         }
     }
-    t->in = in;
+    mIn = in;
 }
 
-void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount,
-        int32_t* temp __unused, int32_t* aux)
+void AudioMixer::Track::track__16BitsMono(
+        int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
 {
     ALOGVV("track__16BitsMono\n");
-    const int16_t *in = static_cast<int16_t const *>(t->in);
+    const int16_t *in = static_cast<int16_t const *>(mIn);
 
     if (CC_UNLIKELY(aux != NULL)) {
         // ramp gain
-        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
-            int32_t vl = t->prevVolume[0];
-            int32_t vr = t->prevVolume[1];
-            int32_t va = t->prevAuxLevel;
-            const int32_t vlInc = t->volumeInc[0];
-            const int32_t vrInc = t->volumeInc[1];
-            const int32_t vaInc = t->auxInc;
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            int32_t va = prevAuxLevel;
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
+            const int32_t vaInc = auxInc;
 
             // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //         t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+            //         t, vlInc/65536.0f, vl/65536.0f, volume[0],
             //         (vl + vlInc*frameCount)/65536.0f, frameCount);
 
             do {
@@ -1309,16 +1194,16 @@
                 va += vaInc;
             } while (--frameCount);
 
-            t->prevVolume[0] = vl;
-            t->prevVolume[1] = vr;
-            t->prevAuxLevel = va;
-            t->adjustVolumeRamp(true);
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            prevAuxLevel = va;
+            adjustVolumeRamp(true);
         }
         // constant gain
         else {
-            const int16_t vl = t->volume[0];
-            const int16_t vr = t->volume[1];
-            const int16_t va = (int16_t)t->auxLevel;
+            const int16_t vl = volume[0];
+            const int16_t vr = volume[1];
+            const int16_t va = (int16_t)auxLevel;
             do {
                 int16_t l = *in++;
                 out[0] = mulAdd(l, vl, out[0]);
@@ -1330,14 +1215,14 @@
         }
     } else {
         // ramp gain
-        if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
-            int32_t vl = t->prevVolume[0];
-            int32_t vr = t->prevVolume[1];
-            const int32_t vlInc = t->volumeInc[0];
-            const int32_t vrInc = t->volumeInc[1];
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
 
             // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //         t, vlInc/65536.0f, vl/65536.0f, t->volume[0],
+            //         t, vlInc/65536.0f, vl/65536.0f, volume[0],
             //         (vl + vlInc*frameCount)/65536.0f, frameCount);
 
             do {
@@ -1348,14 +1233,14 @@
                 vr += vrInc;
             } while (--frameCount);
 
-            t->prevVolume[0] = vl;
-            t->prevVolume[1] = vr;
-            t->adjustVolumeRamp(false);
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            adjustVolumeRamp(false);
         }
         // constant gain
         else {
-            const int16_t vl = t->volume[0];
-            const int16_t vr = t->volume[1];
+            const int16_t vl = volume[0];
+            const int16_t vr = volume[1];
             do {
                 int16_t l = *in++;
                 out[0] = mulAdd(l, vl, out[0]);
@@ -1364,273 +1249,213 @@
             } while (--frameCount);
         }
     }
-    t->in = in;
+    mIn = in;
 }
 
 // no-op case
-void AudioMixer::process__nop(state_t* state)
+void AudioMixer::process__nop()
 {
     ALOGVV("process__nop\n");
-    uint32_t e0 = state->enabledTracks;
-    while (e0) {
+
+    for (const auto &pair : mGroups) {
         // process by group of tracks with same output buffer to
         // avoid multiple memset() on same buffer
-        uint32_t e1 = e0, e2 = e0;
-        int i = 31 - __builtin_clz(e1);
-        {
-            track_t& t1 = state->tracks[i];
-            e2 &= ~(1<<i);
-            while (e2) {
-                i = 31 - __builtin_clz(e2);
-                e2 &= ~(1<<i);
-                track_t& t2 = state->tracks[i];
-                if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
-                    e1 &= ~(1<<i);
-                }
-            }
-            e0 &= ~(e1);
+        const auto &group = pair.second;
 
-            memset(t1.mainBuffer, 0, state->frameCount * t1.mMixerChannelCount
-                    * audio_bytes_per_sample(t1.mMixerFormat));
-        }
+        const std::shared_ptr<Track> &t = mTracks[group[0]];
+        memset(t->mainBuffer, 0,
+                mFrameCount * t->mMixerChannelCount
+                * audio_bytes_per_sample(t->mMixerFormat));
 
-        while (e1) {
-            i = 31 - __builtin_clz(e1);
-            e1 &= ~(1<<i);
-            {
-                track_t& t3 = state->tracks[i];
-                size_t outFrames = state->frameCount;
-                while (outFrames) {
-                    t3.buffer.frameCount = outFrames;
-                    t3.bufferProvider->getNextBuffer(&t3.buffer);
-                    if (t3.buffer.raw == NULL) break;
-                    outFrames -= t3.buffer.frameCount;
-                    t3.bufferProvider->releaseBuffer(&t3.buffer);
-                }
+        // now consume data
+        for (const int name : group) {
+            const std::shared_ptr<Track> &t = mTracks[name];
+            size_t outFrames = mFrameCount;
+            while (outFrames) {
+                t->buffer.frameCount = outFrames;
+                t->bufferProvider->getNextBuffer(&t->buffer);
+                if (t->buffer.raw == NULL) break;
+                outFrames -= t->buffer.frameCount;
+                t->bufferProvider->releaseBuffer(&t->buffer);
             }
         }
     }
 }
 
 // generic code without resampling
-void AudioMixer::process__genericNoResampling(state_t* state)
+void AudioMixer::process__genericNoResampling()
 {
     ALOGVV("process__genericNoResampling\n");
     int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
 
-    // acquire each track's buffer
-    uint32_t enabledTracks = state->enabledTracks;
-    uint32_t e0 = enabledTracks;
-    while (e0) {
-        const int i = 31 - __builtin_clz(e0);
-        e0 &= ~(1<<i);
-        track_t& t = state->tracks[i];
-        t.buffer.frameCount = state->frameCount;
-        t.bufferProvider->getNextBuffer(&t.buffer);
-        t.frameCount = t.buffer.frameCount;
-        t.in = t.buffer.raw;
-    }
+    for (const auto &pair : mGroups) {
+        // process by group of tracks with same output main buffer to
+        // avoid multiple memset() on same buffer
+        const auto &group = pair.second;
 
-    e0 = enabledTracks;
-    while (e0) {
-        // process by group of tracks with same output buffer to
-        // optimize cache use
-        uint32_t e1 = e0, e2 = e0;
-        int j = 31 - __builtin_clz(e1);
-        track_t& t1 = state->tracks[j];
-        e2 &= ~(1<<j);
-        while (e2) {
-            j = 31 - __builtin_clz(e2);
-            e2 &= ~(1<<j);
-            track_t& t2 = state->tracks[j];
-            if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
-                e1 &= ~(1<<j);
-            }
+        // acquire buffer
+        for (const int name : group) {
+            const std::shared_ptr<Track> &t = mTracks[name];
+            t->buffer.frameCount = mFrameCount;
+            t->bufferProvider->getNextBuffer(&t->buffer);
+            t->frameCount = t->buffer.frameCount;
+            t->mIn = t->buffer.raw;
         }
-        e0 &= ~(e1);
-        // this assumes output 16 bits stereo, no resampling
-        int32_t *out = t1.mainBuffer;
+
+        int32_t *out = (int *)pair.first;
         size_t numFrames = 0;
         do {
+            const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames);
             memset(outTemp, 0, sizeof(outTemp));
-            e2 = e1;
-            while (e2) {
-                const int i = 31 - __builtin_clz(e2);
-                e2 &= ~(1<<i);
-                track_t& t = state->tracks[i];
-                size_t outFrames = BLOCKSIZE;
+            for (const int name : group) {
+                const std::shared_ptr<Track> &t = mTracks[name];
                 int32_t *aux = NULL;
-                if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
-                    aux = t.auxBuffer + numFrames;
+                if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
+                    aux = t->auxBuffer + numFrames;
                 }
-                while (outFrames) {
-                    // t.in == NULL can happen if the track was flushed just after having
+                for (int outFrames = frameCount; outFrames > 0; ) {
+                    // t->in == nullptr can happen if the track was flushed just after having
                     // been enabled for mixing.
-                   if (t.in == NULL) {
-                        enabledTracks &= ~(1<<i);
-                        e1 &= ~(1<<i);
+                    if (t->mIn == nullptr) {
                         break;
                     }
-                    size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
+                    size_t inFrames = (t->frameCount > outFrames)?outFrames:t->frameCount;
                     if (inFrames > 0) {
-                        t.hook(&t, outTemp + (BLOCKSIZE - outFrames) * t.mMixerChannelCount,
-                                inFrames, state->resampleTemp, aux);
-                        t.frameCount -= inFrames;
+                        (t.get()->*t->hook)(
+                                outTemp + (frameCount - outFrames) * t->mMixerChannelCount,
+                                inFrames, mResampleTemp.get() /* naked ptr */, aux);
+                        t->frameCount -= inFrames;
                         outFrames -= inFrames;
                         if (CC_UNLIKELY(aux != NULL)) {
                             aux += inFrames;
                         }
                     }
-                    if (t.frameCount == 0 && outFrames) {
-                        t.bufferProvider->releaseBuffer(&t.buffer);
-                        t.buffer.frameCount = (state->frameCount - numFrames) -
-                                (BLOCKSIZE - outFrames);
-                        t.bufferProvider->getNextBuffer(&t.buffer);
-                        t.in = t.buffer.raw;
-                        if (t.in == NULL) {
-                            enabledTracks &= ~(1<<i);
-                            e1 &= ~(1<<i);
+                    if (t->frameCount == 0 && outFrames) {
+                        t->bufferProvider->releaseBuffer(&t->buffer);
+                        t->buffer.frameCount = (mFrameCount - numFrames) -
+                                (frameCount - outFrames);
+                        t->bufferProvider->getNextBuffer(&t->buffer);
+                        t->mIn = t->buffer.raw;
+                        if (t->mIn == nullptr) {
                             break;
                         }
-                        t.frameCount = t.buffer.frameCount;
+                        t->frameCount = t->buffer.frameCount;
                     }
                 }
             }
 
-            convertMixerFormat(out, t1.mMixerFormat, outTemp, t1.mMixerInFormat,
-                    BLOCKSIZE * t1.mMixerChannelCount);
+            const std::shared_ptr<Track> &t1 = mTracks[group[0]];
+            convertMixerFormat(out, t1->mMixerFormat, outTemp, t1->mMixerInFormat,
+                    frameCount * t1->mMixerChannelCount);
             // TODO: fix ugly casting due to choice of out pointer type
             out = reinterpret_cast<int32_t*>((uint8_t*)out
-                    + BLOCKSIZE * t1.mMixerChannelCount
-                        * audio_bytes_per_sample(t1.mMixerFormat));
-            numFrames += BLOCKSIZE;
-        } while (numFrames < state->frameCount);
-    }
+                    + frameCount * t1->mMixerChannelCount
+                    * audio_bytes_per_sample(t1->mMixerFormat));
+            numFrames += frameCount;
+        } while (numFrames < mFrameCount);
 
-    // release each track's buffer
-    e0 = enabledTracks;
-    while (e0) {
-        const int i = 31 - __builtin_clz(e0);
-        e0 &= ~(1<<i);
-        track_t& t = state->tracks[i];
-        t.bufferProvider->releaseBuffer(&t.buffer);
+        // release each track's buffer
+        for (const int name : group) {
+            const std::shared_ptr<Track> &t = mTracks[name];
+            t->bufferProvider->releaseBuffer(&t->buffer);
+        }
     }
 }
 
-
 // generic code with resampling
-void AudioMixer::process__genericResampling(state_t* state)
+void AudioMixer::process__genericResampling()
 {
     ALOGVV("process__genericResampling\n");
-    // this const just means that local variable outTemp doesn't change
-    int32_t* const outTemp = state->outputTemp;
-    size_t numFrames = state->frameCount;
+    int32_t * const outTemp = mOutputTemp.get(); // naked ptr
+    size_t numFrames = mFrameCount;
 
-    uint32_t e0 = state->enabledTracks;
-    while (e0) {
-        // process by group of tracks with same output buffer
-        // to optimize cache use
-        uint32_t e1 = e0, e2 = e0;
-        int j = 31 - __builtin_clz(e1);
-        track_t& t1 = state->tracks[j];
-        e2 &= ~(1<<j);
-        while (e2) {
-            j = 31 - __builtin_clz(e2);
-            e2 &= ~(1<<j);
-            track_t& t2 = state->tracks[j];
-            if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
-                e1 &= ~(1<<j);
-            }
-        }
-        e0 &= ~(e1);
-        int32_t *out = t1.mainBuffer;
-        memset(outTemp, 0, sizeof(*outTemp) * t1.mMixerChannelCount * state->frameCount);
-        while (e1) {
-            const int i = 31 - __builtin_clz(e1);
-            e1 &= ~(1<<i);
-            track_t& t = state->tracks[i];
+    for (const auto &pair : mGroups) {
+        const auto &group = pair.second;
+        const std::shared_ptr<Track> &t1 = mTracks[group[0]];
+
+        // clear temp buffer
+        memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount);
+        for (const int name : group) {
+            const std::shared_ptr<Track> &t = mTracks[name];
             int32_t *aux = NULL;
-            if (CC_UNLIKELY(t.needs & NEEDS_AUX)) {
-                aux = t.auxBuffer;
+            if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
+                aux = t->auxBuffer;
             }
 
             // this is a little goofy, on the resampling case we don't
             // acquire/release the buffers because it's done by
             // the resampler.
-            if (t.needs & NEEDS_RESAMPLE) {
-                t.hook(&t, outTemp, numFrames, state->resampleTemp, aux);
+            if (t->needs & NEEDS_RESAMPLE) {
+                (t.get()->*t->hook)(outTemp, numFrames, mResampleTemp.get() /* naked ptr */, aux);
             } else {
 
                 size_t outFrames = 0;
 
                 while (outFrames < numFrames) {
-                    t.buffer.frameCount = numFrames - outFrames;
-                    t.bufferProvider->getNextBuffer(&t.buffer);
-                    t.in = t.buffer.raw;
-                    // t.in == NULL can happen if the track was flushed just after having
+                    t->buffer.frameCount = numFrames - outFrames;
+                    t->bufferProvider->getNextBuffer(&t->buffer);
+                    t->mIn = t->buffer.raw;
+                    // t->mIn == nullptr can happen if the track was flushed just after having
                     // been enabled for mixing.
-                    if (t.in == NULL) break;
+                    if (t->mIn == nullptr) break;
 
-                    if (CC_UNLIKELY(aux != NULL)) {
-                        aux += outFrames;
-                    }
-                    t.hook(&t, outTemp + outFrames * t.mMixerChannelCount, t.buffer.frameCount,
-                            state->resampleTemp, aux);
-                    outFrames += t.buffer.frameCount;
-                    t.bufferProvider->releaseBuffer(&t.buffer);
+                    (t.get()->*t->hook)(
+                            outTemp + outFrames * t->mMixerChannelCount, t->buffer.frameCount,
+                            mResampleTemp.get() /* naked ptr */,
+                            aux != nullptr ? aux + outFrames : nullptr);
+                    outFrames += t->buffer.frameCount;
+
+                    t->bufferProvider->releaseBuffer(&t->buffer);
                 }
             }
         }
-        convertMixerFormat(out, t1.mMixerFormat,
-                outTemp, t1.mMixerInFormat, numFrames * t1.mMixerChannelCount);
+        convertMixerFormat(t1->mainBuffer, t1->mMixerFormat,
+                outTemp, t1->mMixerInFormat, numFrames * t1->mMixerChannelCount);
     }
 }
 
 // one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
+void AudioMixer::process__oneTrack16BitsStereoNoResampling()
 {
-    ALOGVV("process__OneTrack16BitsStereoNoResampling\n");
-    // This method is only called when state->enabledTracks has exactly
-    // one bit set.  The asserts below would verify this, but are commented out
-    // since the whole point of this method is to optimize performance.
-    //ALOG_ASSERT(0 != state->enabledTracks, "no tracks enabled");
-    const int i = 31 - __builtin_clz(state->enabledTracks);
-    //ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
-    const track_t& t = state->tracks[i];
+    ALOGVV("process__oneTrack16BitsStereoNoResampling\n");
+    LOG_ALWAYS_FATAL_IF(mEnabled.size() != 0,
+            "%zu != 1 tracks enabled", mEnabled.size());
+    const int name = mEnabled[0];
+    const std::shared_ptr<Track> &t = mTracks[name];
 
-    AudioBufferProvider::Buffer& b(t.buffer);
+    AudioBufferProvider::Buffer& b(t->buffer);
 
-    int32_t* out = t.mainBuffer;
+    int32_t* out = t->mainBuffer;
     float *fout = reinterpret_cast<float*>(out);
-    size_t numFrames = state->frameCount;
+    size_t numFrames = mFrameCount;
 
-    const int16_t vl = t.volume[0];
-    const int16_t vr = t.volume[1];
-    const uint32_t vrl = t.volumeRL;
+    const int16_t vl = t->volume[0];
+    const int16_t vr = t->volume[1];
+    const uint32_t vrl = t->volumeRL;
     while (numFrames) {
         b.frameCount = numFrames;
-        t.bufferProvider->getNextBuffer(&b);
+        t->bufferProvider->getNextBuffer(&b);
         const int16_t *in = b.i16;
 
         // in == NULL can happen if the track was flushed just after having
         // been enabled for mixing.
         if (in == NULL || (((uintptr_t)in) & 3)) {
-            if ( AUDIO_FORMAT_PCM_FLOAT == t.mMixerFormat ) {
+            if ( AUDIO_FORMAT_PCM_FLOAT == t->mMixerFormat ) {
                  memset((char*)fout, 0, numFrames
-                         * t.mMixerChannelCount * audio_bytes_per_sample(t.mMixerFormat));
+                         * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
             } else {
                  memset((char*)out, 0, numFrames
-                         * t.mMixerChannelCount * audio_bytes_per_sample(t.mMixerFormat));
+                         * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
             }
             ALOGE_IF((((uintptr_t)in) & 3),
-                    "process__OneTrack16BitsStereoNoResampling: misaligned buffer"
+                    "process__oneTrack16BitsStereoNoResampling: misaligned buffer"
                     " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
-                    in, i, t.channelCount, t.needs, vrl, t.mVolume[0], t.mVolume[1]);
+                    in, name, t->channelCount, t->needs, vrl, t->mVolume[0], t->mVolume[1]);
             return;
         }
         size_t outFrames = b.frameCount;
 
-        switch (t.mMixerFormat) {
+        switch (t->mMixerFormat) {
         case AUDIO_FORMAT_PCM_FLOAT:
             do {
                 uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
@@ -1668,10 +1493,10 @@
             }
             break;
         default:
-            LOG_ALWAYS_FATAL("bad mixer format: %d", t.mMixerFormat);
+            LOG_ALWAYS_FATAL("bad mixer format: %d", t->mMixerFormat);
         }
         numFrames -= b.frameCount;
-        t.bufferProvider->releaseBuffer(&b);
+        t->bufferProvider->releaseBuffer(&b);
     }
 }
 
@@ -1694,7 +1519,7 @@
 /* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
  * TO: int32_t (Q4.27) or float
  * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27)
+ * TA: int32_t (Q4.27) or float
  */
 template <int MIXTYPE,
         typename TO, typename TI, typename TV, typename TA, typename TAV>
@@ -1738,7 +1563,7 @@
 /* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
  * TO: int32_t (Q4.27) or float
  * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27)
+ * TA: int32_t (Q4.27) or float
  */
 template <int MIXTYPE,
         typename TO, typename TI, typename TV, typename TA, typename TAV>
@@ -1778,34 +1603,46 @@
  * ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
  * TO: int32_t (Q4.27) or float
  * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27)
+ * TA: int32_t (Q4.27) or float
  */
 template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
     typename TO, typename TI, typename TA>
-void AudioMixer::volumeMix(TO *out, size_t outFrames,
-        const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t)
+void AudioMixer::Track::volumeMix(TO *out, size_t outFrames,
+        const TI *in, TA *aux, bool ramp)
 {
     if (USEFLOATVOL) {
         if (ramp) {
-            volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
-                    t->mPrevVolume, t->mVolumeInc, &t->prevAuxLevel, t->auxInc);
+            volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    mPrevVolume, mVolumeInc,
+#ifdef FLOAT_AUX
+                    &mPrevAuxLevel, mAuxInc
+#else
+                    &prevAuxLevel, auxInc
+#endif
+                );
             if (ADJUSTVOL) {
-                t->adjustVolumeRamp(aux != NULL, true);
+                adjustVolumeRamp(aux != NULL, true);
             }
         } else {
-            volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
-                    t->mVolume, t->auxLevel);
+            volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    mVolume,
+#ifdef FLOAT_AUX
+                    mAuxLevel
+#else
+                    auxLevel
+#endif
+            );
         }
     } else {
         if (ramp) {
-            volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
-                    t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc);
+            volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    prevVolume, volumeInc, &prevAuxLevel, auxInc);
             if (ADJUSTVOL) {
-                t->adjustVolumeRamp(aux != NULL);
+                adjustVolumeRamp(aux != NULL);
             }
         } else {
-            volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
-                    t->volume, t->auxLevel);
+            volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    volume, auxLevel);
         }
     }
 }
@@ -1820,19 +1657,18 @@
  * TA: int32_t (Q4.27)
  */
 template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::process_NoResampleOneTrack(state_t* state)
+void AudioMixer::process__noResampleOneTrack()
 {
-    ALOGVV("process_NoResampleOneTrack\n");
-    // CLZ is faster than CTZ on ARM, though really not sure if true after 31 - clz.
-    const int i = 31 - __builtin_clz(state->enabledTracks);
-    ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
-    track_t *t = &state->tracks[i];
+    ALOGVV("process__noResampleOneTrack\n");
+    LOG_ALWAYS_FATAL_IF(mEnabled.size() != 1,
+            "%zu != 1 tracks enabled", mEnabled.size());
+    const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
     const uint32_t channels = t->mMixerChannelCount;
     TO* out = reinterpret_cast<TO*>(t->mainBuffer);
     TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
     const bool ramp = t->needsRamp();
 
-    for (size_t numFrames = state->frameCount; numFrames; ) {
+    for (size_t numFrames = mFrameCount; numFrames > 0; ) {
         AudioBufferProvider::Buffer& b(t->buffer);
         // get input buffer
         b.frameCount = numFrames;
@@ -1844,19 +1680,19 @@
         if (in == NULL || (((uintptr_t)in) & 3)) {
             memset(out, 0, numFrames
                     * channels * audio_bytes_per_sample(t->mMixerFormat));
-            ALOGE_IF((((uintptr_t)in) & 3), "process_NoResampleOneTrack: bus error: "
+            ALOGE_IF((((uintptr_t)in) & 3), "process__noResampleOneTrack: bus error: "
                     "buffer %p track %p, channels %d, needs %#x",
-                    in, t, t->channelCount, t->needs);
+                    in, &t, t->channelCount, t->needs);
             return;
         }
 
         const size_t outFrames = b.frameCount;
-        volumeMix<MIXTYPE, is_same<TI, float>::value, false> (
-                out, outFrames, in, aux, ramp, t);
+        t->volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (
+                out, outFrames, in, aux, ramp);
 
         out += outFrames * channels;
         if (aux != NULL) {
-            aux += channels;
+            aux += outFrames;
         }
         numFrames -= b.frameCount;
 
@@ -1874,59 +1710,59 @@
  * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
  * TO: int32_t (Q4.27) or float
  * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27)
+ * TA: int32_t (Q4.27) or float
  */
 template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux)
+void AudioMixer::Track::track__Resample(TO* out, size_t outFrameCount, TO* temp, TA* aux)
 {
     ALOGVV("track__Resample\n");
-    t->resampler->setSampleRate(t->sampleRate);
-    const bool ramp = t->needsRamp();
+    mResampler->setSampleRate(sampleRate);
+    const bool ramp = needsRamp();
     if (ramp || aux != NULL) {
         // if ramp:        resample with unity gain to temp buffer and scale/mix in 2nd step.
         // if aux != NULL: resample with unity gain to temp buffer then apply send level.
 
-        t->resampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
-        memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(TO));
-        t->resampler->resample((int32_t*)temp, outFrameCount, t->bufferProvider);
+        mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+        memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(TO));
+        mResampler->resample((int32_t*)temp, outFrameCount, bufferProvider);
 
-        volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
-                out, outFrameCount, temp, aux, ramp, t);
+        volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
+                out, outFrameCount, temp, aux, ramp);
 
     } else { // constant volume gain
-        t->resampler->setVolume(t->mVolume[0], t->mVolume[1]);
-        t->resampler->resample((int32_t*)out, outFrameCount, t->bufferProvider);
+        mResampler->setVolume(mVolume[0], mVolume[1]);
+        mResampler->resample((int32_t*)out, outFrameCount, bufferProvider);
     }
 }
 
 /* This track hook is called to mix a track, when no resampling is required.
- * The input buffer should be present in t->in.
+ * The input buffer should be present in in.
  *
  * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
  * TO: int32_t (Q4.27) or float
  * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27)
+ * TA: int32_t (Q4.27) or float
  */
 template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
-        TO* temp __unused, TA* aux)
+void AudioMixer::Track::track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux)
 {
     ALOGVV("track__NoResample\n");
-    const TI *in = static_cast<const TI *>(t->in);
+    const TI *in = static_cast<const TI *>(mIn);
 
-    volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
-            out, frameCount, in, aux, t->needsRamp(), t);
+    volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
+            out, frameCount, in, aux, needsRamp());
 
     // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
     // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
-    in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * t->mMixerChannelCount;
-    t->in = in;
+    in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * mMixerChannelCount;
+    mIn = in;
 }
 
 /* The Mixer engine generates either int32_t (Q4_27) or float data.
  * We use this function to convert the engine buffers
  * to the desired mixer output format, either int16_t (Q.15) or float.
  */
+/* static */
 void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
         void *in, audio_format_t mixerInFormat, size_t sampleCount)
 {
@@ -1947,11 +1783,10 @@
     case AUDIO_FORMAT_PCM_16_BIT:
         switch (mixerOutFormat) {
         case AUDIO_FORMAT_PCM_FLOAT:
-            memcpy_to_float_from_q4_27((float*)out, (int32_t*)in, sampleCount);
+            memcpy_to_float_from_q4_27((float*)out, (const int32_t*)in, sampleCount);
             break;
         case AUDIO_FORMAT_PCM_16_BIT:
-            // two int16_t are produced per iteration
-            ditherAndClamp((int32_t*)out, (int32_t*)in, sampleCount >> 1);
+            memcpy_to_i16_from_q4_27((int16_t*)out, (const int32_t*)in, sampleCount);
             break;
         default:
             LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
@@ -1966,19 +1801,20 @@
 
 /* Returns the proper track hook to use for mixing the track into the output buffer.
  */
-AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount,
+/* static */
+AudioMixer::hook_t AudioMixer::Track::getTrackHook(int trackType, uint32_t channelCount,
         audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
 {
     if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
         switch (trackType) {
         case TRACKTYPE_NOP:
-            return track__nop;
+            return &Track::track__nop;
         case TRACKTYPE_RESAMPLE:
-            return track__genericResample;
+            return &Track::track__genericResample;
         case TRACKTYPE_NORESAMPLEMONO:
-            return track__16BitsMono;
+            return &Track::track__16BitsMono;
         case TRACKTYPE_NORESAMPLE:
-            return track__16BitsStereo;
+            return &Track::track__16BitsStereo;
         default:
             LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
             break;
@@ -1987,15 +1823,15 @@
     LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
     switch (trackType) {
     case TRACKTYPE_NOP:
-        return track__nop;
+        return &Track::track__nop;
     case TRACKTYPE_RESAMPLE:
         switch (mixerInFormat) {
         case AUDIO_FORMAT_PCM_FLOAT:
-            return (AudioMixer::hook_t)
-                    track__Resample<MIXTYPE_MULTI, float /*TO*/, float /*TI*/, int32_t /*TA*/>;
+            return (AudioMixer::hook_t) &Track::track__Resample<
+                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
         case AUDIO_FORMAT_PCM_16_BIT:
-            return (AudioMixer::hook_t)\
-                    track__Resample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
+            return (AudioMixer::hook_t) &Track::track__Resample<
+                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
         default:
             LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
             break;
@@ -2004,11 +1840,11 @@
     case TRACKTYPE_NORESAMPLEMONO:
         switch (mixerInFormat) {
         case AUDIO_FORMAT_PCM_FLOAT:
-            return (AudioMixer::hook_t)
-                    track__NoResample<MIXTYPE_MONOEXPAND, float, float, int32_t>;
+            return (AudioMixer::hook_t) &Track::track__NoResample<
+                            MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;
         case AUDIO_FORMAT_PCM_16_BIT:
-            return (AudioMixer::hook_t)
-                    track__NoResample<MIXTYPE_MONOEXPAND, int32_t, int16_t, int32_t>;
+            return (AudioMixer::hook_t) &Track::track__NoResample<
+                            MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
         default:
             LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
             break;
@@ -2017,11 +1853,11 @@
     case TRACKTYPE_NORESAMPLE:
         switch (mixerInFormat) {
         case AUDIO_FORMAT_PCM_FLOAT:
-            return (AudioMixer::hook_t)
-                    track__NoResample<MIXTYPE_MULTI, float, float, int32_t>;
+            return (AudioMixer::hook_t) &Track::track__NoResample<
+                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
         case AUDIO_FORMAT_PCM_16_BIT:
-            return (AudioMixer::hook_t)
-                    track__NoResample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
+            return (AudioMixer::hook_t) &Track::track__NoResample<
+                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
         default:
             LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
             break;
@@ -2041,7 +1877,9 @@
  * a stereo output track, the input track cannot be MONO.  This should be
  * prevented by the caller.
  */
-AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, uint32_t channelCount,
+/* static */
+AudioMixer::process_hook_t AudioMixer::getProcessHook(
+        int processType, uint32_t channelCount,
         audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
 {
     if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
@@ -2049,18 +1887,18 @@
         return NULL;
     }
     if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
-        return process__OneTrack16BitsStereoNoResampling;
+        return &AudioMixer::process__oneTrack16BitsStereoNoResampling;
     }
     LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
     switch (mixerInFormat) {
     case AUDIO_FORMAT_PCM_FLOAT:
         switch (mixerOutFormat) {
         case AUDIO_FORMAT_PCM_FLOAT:
-            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
-                    float /*TO*/, float /*TI*/, int32_t /*TA*/>;
+            return &AudioMixer::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;
         case AUDIO_FORMAT_PCM_16_BIT:
-            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
-                    int16_t, float, int32_t>;
+            return &AudioMixer::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;
         default:
             LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
             break;
@@ -2069,11 +1907,11 @@
     case AUDIO_FORMAT_PCM_16_BIT:
         switch (mixerOutFormat) {
         case AUDIO_FORMAT_PCM_FLOAT:
-            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
-                    float, int16_t, int32_t>;
+            return &AudioMixer::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;
         case AUDIO_FORMAT_PCM_16_BIT:
-            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
-                    int16_t, int16_t, int32_t>;
+            return &AudioMixer::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
         default:
             LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
             break;

diff --git a/media/libaudioprocessing/AudioMixerOps.h b/media/libaudioprocessing/AudioMixerOps.h
index 8d74024..f33e361 100644
--- a/media/libaudioprocessing/AudioMixerOps.h
+++ b/media/libaudioprocessing/AudioMixerOps.h

@@ -188,13 +188,13 @@
 
 template<>
 inline void MixAccum<float, int16_t>(float *auxaccum, int16_t value) {
-    static const float norm = 1. / (1 << 15);
+    static constexpr float norm = 1. / (1 << 15);
     *auxaccum += norm * value;
 }
 
 template<>
 inline void MixAccum<float, int32_t>(float *auxaccum, int32_t value) {
-    static const float norm = 1. / (1 << 27);
+    static constexpr float norm = 1. / (1 << 27);
     *auxaccum += norm * value;
 }
 
@@ -238,6 +238,7 @@
  *   NCHAN represents number of input and output channels.
  *   TO: int32_t (Q4.27) or float
  *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ *   TA: int32_t (Q4.27) or float
  *   TV: int32_t (U4.28) or int16_t (U4.12) or float
  *   vol: represents a volume array.
  *
@@ -247,7 +248,8 @@
  *   Single input channel. NCHAN represents number of output channels.
  *   TO: int32_t (Q4.27) or float
  *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- *   TV: int32_t (U4.28) or int16_t (U4.12) or float
+ *   TA: int32_t (Q4.27) or float
+ *   TV/TAV: int32_t (U4.28) or int16_t (U4.12) or float
  *   Input channel count is 1.
  *   vol: represents volume array.
  *
@@ -257,7 +259,8 @@
  *   NCHAN represents number of input and output channels.
  *   TO: int16_t (Q.15) or float
  *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- *   TV: int32_t (U4.28) or int16_t (U4.12) or float
+ *   TA: int32_t (Q4.27) or float
+ *   TV/TAV: int32_t (U4.28) or int16_t (U4.12) or float
  *   vol: represents a volume array.
  *
  *   MIXTYPE_MULTI_SAVEONLY does not accumulate into the out pointer.

diff --git a/media/libaudioprocessing/AudioResamplerDyn.cpp b/media/libaudioprocessing/AudioResamplerDyn.cpp
index 8f7b982..eeeecce 100644
--- a/media/libaudioprocessing/AudioResamplerDyn.cpp
+++ b/media/libaudioprocessing/AudioResamplerDyn.cpp

@@ -38,6 +38,9 @@
 
 //#define DEBUG_RESAMPLER
 
+// use this for our buffer alignment.  Should be at least 32 bytes.
+constexpr size_t CACHE_LINE_SIZE = 64;
+
 namespace android {
 
 /*
@@ -94,7 +97,10 @@
 
     // create new buffer
     TI* state = NULL;
-    (void)posix_memalign(reinterpret_cast<void**>(&state), 32, stateCount*sizeof(*state));
+    (void)posix_memalign(
+            reinterpret_cast<void **>(&state),
+            CACHE_LINE_SIZE /* alignment */,
+            stateCount * sizeof(*state));
     memset(state, 0, stateCount*sizeof(*state));
 
     // attempt to preserve state
@@ -185,6 +191,16 @@
     // setSampleRate() for 1:1. (May be removed if precalculated filters are used.)
     mInSampleRate = 0;
     mConstants.set(128, 8, mSampleRate, mSampleRate); // TODO: set better
+
+    // fetch property based resampling parameters
+    mPropertyEnableAtSampleRate = property_get_int32(
+            "ro.audio.resampler.psd.enable_at_samplerate", mPropertyEnableAtSampleRate);
+    mPropertyHalfFilterLength = property_get_int32(
+            "ro.audio.resampler.psd.halflength", mPropertyHalfFilterLength);
+    mPropertyStopbandAttenuation = property_get_int32(
+            "ro.audio.resampler.psd.stopband", mPropertyStopbandAttenuation);
+    mPropertyCutoffPercent = property_get_int32(
+            "ro.audio.resampler.psd.cutoff_percent", mPropertyCutoffPercent);
 }
 
 template<typename TC, typename TI, typename TO>
@@ -215,6 +231,8 @@
     }
 }
 
+// TODO: update to C++11
+
 template<typename T> T max(T a, T b) {return a > b ? a : b;}
 
 template<typename T> T absdiff(T a, T b) {return a > b ? a - b : b - a;}
@@ -223,37 +241,74 @@
 void AudioResamplerDyn<TC, TI, TO>::createKaiserFir(Constants &c,
         double stopBandAtten, int inSampleRate, int outSampleRate, double tbwCheat)
 {
-    TC* buf = NULL;
-    static const double atten = 0.9998;   // to avoid ripple overflow
-    double fcr;
-    double tbw = firKaiserTbw(c.mHalfNumCoefs, stopBandAtten);
+    // compute the normalized transition bandwidth
+    const double tbw = firKaiserTbw(c.mHalfNumCoefs, stopBandAtten);
+    const double halfbw = tbw / 2.;
 
-    (void)posix_memalign(reinterpret_cast<void**>(&buf), 32, (c.mL+1)*c.mHalfNumCoefs*sizeof(TC));
+    double fcr; // compute fcr, the 3 dB amplitude cut-off.
     if (inSampleRate < outSampleRate) { // upsample
-        fcr = max(0.5*tbwCheat - tbw/2, tbw/2);
+        fcr = max(0.5 * tbwCheat - halfbw, halfbw);
     } else { // downsample
-        fcr = max(0.5*tbwCheat*outSampleRate/inSampleRate - tbw/2, tbw/2);
+        fcr = max(0.5 * tbwCheat * outSampleRate / inSampleRate - halfbw, halfbw);
     }
-    // create and set filter
-    firKaiserGen(buf, c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten);
-    c.mFirCoefs = buf;
-    if (mCoefBuffer) {
-        free(mCoefBuffer);
-    }
-    mCoefBuffer = buf;
-#ifdef DEBUG_RESAMPLER
+    createKaiserFir(c, stopBandAtten, fcr);
+}
+
+template<typename TC, typename TI, typename TO>
+void AudioResamplerDyn<TC, TI, TO>::createKaiserFir(Constants &c,
+        double stopBandAtten, double fcr) {
+    // compute the normalized transition bandwidth
+    const double tbw = firKaiserTbw(c.mHalfNumCoefs, stopBandAtten);
+    const int phases = c.mL;
+    const int halfLength = c.mHalfNumCoefs;
+
+    // create buffer
+    TC *coefs = nullptr;
+    int ret = posix_memalign(
+            reinterpret_cast<void **>(&coefs),
+            CACHE_LINE_SIZE /* alignment */,
+            (phases + 1) * halfLength * sizeof(TC));
+    LOG_ALWAYS_FATAL_IF(ret != 0, "Cannot allocate buffer memory, ret %d", ret);
+    c.mFirCoefs = coefs;
+    free(mCoefBuffer);
+    mCoefBuffer = coefs;
+
+    // square the computed minimum passband value (extra safety).
+    double attenuation =
+            computeWindowedSincMinimumPassbandValue(stopBandAtten);
+    attenuation *= attenuation;
+
+    // design filter
+    firKaiserGen(coefs, phases, halfLength, stopBandAtten, fcr, attenuation);
+
+    // update the design criteria
+    mNormalizedCutoffFrequency = fcr;
+    mNormalizedTransitionBandwidth = tbw;
+    mFilterAttenuation = attenuation;
+    mStopbandAttenuationDb = stopBandAtten;
+    mPassbandRippleDb = computeWindowedSincPassbandRippleDb(stopBandAtten);
+
+#if 0
+    // Keep this debug code in case an app causes resampler design issues.
+    const double halfbw = tbw / 2.;
     // print basic filter stats
-    printf("L:%d  hnc:%d  stopBandAtten:%lf  fcr:%lf  atten:%lf  tbw:%lf\n",
-            c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten, tbw);
-    // test the filter and report results
-    double fp = (fcr - tbw/2)/c.mL;
-    double fs = (fcr + tbw/2)/c.mL;
+    ALOGD("L:%d  hnc:%d  stopBandAtten:%lf  fcr:%lf  atten:%lf  tbw:%lf\n",
+            c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, attenuation, tbw);
+
+    // test the filter and report results.
+    // Since this is a polyphase filter, normalized fp and fs must be scaled.
+    const double fp = (fcr - halfbw) / phases;
+    const double fs = (fcr + halfbw) / phases;
+
     double passMin, passMax, passRipple;
     double stopMax, stopRipple;
-    testFir(buf, c.mL, c.mHalfNumCoefs, fp, fs, /*passSteps*/ 1000, /*stopSteps*/ 100000,
+
+    const int32_t passSteps = 1000;
+
+    testFir(coefs, c.mL, c.mHalfNumCoefs, fp, fs, passSteps, passSteps * c.ML /*stopSteps*/,
             passMin, passMax, passRipple, stopMax, stopRipple);
-    printf("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, passMin, passMax, passRipple);
-    printf("stopband(%lf, %lf): %.8lf %.3lf\n", fs, 0.5, stopMax, stopRipple);
+    ALOGD("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, passMin, passMax, passRipple);
+    ALOGD("stopband(%lf, %lf): %.8lf %.3lf\n", fs, 0.5, stopMax, stopRipple);
 #endif
 }
 
@@ -304,6 +359,11 @@
         mFilterSampleRate = inSampleRate;
         mFilterQuality = getQuality();
 
+        double stopBandAtten;
+        double tbwCheat = 1.; // how much we "cheat" into aliasing
+        int halfLength;
+        double fcr = 0.;
+
         // Begin Kaiser Filter computation
         //
         // The quantization floor for S16 is about 96db - 10*log_10(#length) + 3dB.
@@ -313,52 +373,60 @@
         // 96-98dB
         //
 
-        double stopBandAtten;
-        double tbwCheat = 1.; // how much we "cheat" into aliasing
-        int halfLength;
-        if (mFilterQuality == DYN_HIGH_QUALITY) {
-            // 32b coefficients, 64 length
+        if (mPropertyEnableAtSampleRate >= 0 && mSampleRate >= mPropertyEnableAtSampleRate) {
+            // An alternative method which allows allows a greater fcr
+            // at the expense of potential aliasing.
+            halfLength = mPropertyHalfFilterLength;
+            stopBandAtten = mPropertyStopbandAttenuation;
             useS32 = true;
-            stopBandAtten = 98.;
-            if (inSampleRate >= mSampleRate * 4) {
-                halfLength = 48;
-            } else if (inSampleRate >= mSampleRate * 2) {
-                halfLength = 40;
-            } else {
-                halfLength = 32;
-            }
-        } else if (mFilterQuality == DYN_LOW_QUALITY) {
-            // 16b coefficients, 16-32 length
-            useS32 = false;
-            stopBandAtten = 80.;
-            if (inSampleRate >= mSampleRate * 4) {
-                halfLength = 24;
-            } else if (inSampleRate >= mSampleRate * 2) {
-                halfLength = 16;
-            } else {
-                halfLength = 8;
-            }
-            if (inSampleRate <= mSampleRate) {
-                tbwCheat = 1.05;
-            } else {
-                tbwCheat = 1.03;
-            }
-        } else { // DYN_MED_QUALITY
-            // 16b coefficients, 32-64 length
-            // note: > 64 length filters with 16b coefs can have quantization noise problems
-            useS32 = false;
-            stopBandAtten = 84.;
-            if (inSampleRate >= mSampleRate * 4) {
-                halfLength = 32;
-            } else if (inSampleRate >= mSampleRate * 2) {
-                halfLength = 24;
-            } else {
-                halfLength = 16;
-            }
-            if (inSampleRate <= mSampleRate) {
-                tbwCheat = 1.03;
-            } else {
-                tbwCheat = 1.01;
+            fcr = mInSampleRate <= mSampleRate
+                    ? 0.5 : 0.5 * mSampleRate / mInSampleRate;
+            fcr *= mPropertyCutoffPercent / 100.;
+        } else {
+            if (mFilterQuality == DYN_HIGH_QUALITY) {
+                // 32b coefficients, 64 length
+                useS32 = true;
+                stopBandAtten = 98.;
+                if (inSampleRate >= mSampleRate * 4) {
+                    halfLength = 48;
+                } else if (inSampleRate >= mSampleRate * 2) {
+                    halfLength = 40;
+                } else {
+                    halfLength = 32;
+                }
+            } else if (mFilterQuality == DYN_LOW_QUALITY) {
+                // 16b coefficients, 16-32 length
+                useS32 = false;
+                stopBandAtten = 80.;
+                if (inSampleRate >= mSampleRate * 4) {
+                    halfLength = 24;
+                } else if (inSampleRate >= mSampleRate * 2) {
+                    halfLength = 16;
+                } else {
+                    halfLength = 8;
+                }
+                if (inSampleRate <= mSampleRate) {
+                    tbwCheat = 1.05;
+                } else {
+                    tbwCheat = 1.03;
+                }
+            } else { // DYN_MED_QUALITY
+                // 16b coefficients, 32-64 length
+                // note: > 64 length filters with 16b coefs can have quantization noise problems
+                useS32 = false;
+                stopBandAtten = 84.;
+                if (inSampleRate >= mSampleRate * 4) {
+                    halfLength = 32;
+                } else if (inSampleRate >= mSampleRate * 2) {
+                    halfLength = 24;
+                } else {
+                    halfLength = 16;
+                }
+                if (inSampleRate <= mSampleRate) {
+                    tbwCheat = 1.03;
+                } else {
+                    tbwCheat = 1.01;
+                }
             }
         }
 
@@ -390,8 +458,12 @@
 
         // create the filter
         mConstants.set(phases, halfLength, inSampleRate, mSampleRate);
-        createKaiserFir(mConstants, stopBandAtten,
-                inSampleRate, mSampleRate, tbwCheat);
+        if (fcr > 0.) {
+            createKaiserFir(mConstants, stopBandAtten, fcr);
+        } else {
+            createKaiserFir(mConstants, stopBandAtten,
+                    inSampleRate, mSampleRate, tbwCheat);
+        }
     } // End Kaiser filter
 
     // update phase and state based on the new filter.

diff --git a/media/libaudioprocessing/AudioResamplerDyn.h b/media/libaudioprocessing/AudioResamplerDyn.h
index 1840fc7..92144d0 100644
--- a/media/libaudioprocessing/AudioResamplerDyn.h
+++ b/media/libaudioprocessing/AudioResamplerDyn.h

@@ -55,6 +55,39 @@
     virtual size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 
+    // Make available key design criteria for testing
+    int getHalfLength() const {
+        return mConstants.mHalfNumCoefs;
+    }
+
+    const TC *getFilterCoefs() const {
+        return mConstants.mFirCoefs;
+    }
+
+    int getPhases() const {
+        return mConstants.mL;
+    }
+
+    double getStopbandAttenuationDb() const {
+        return mStopbandAttenuationDb;
+    }
+
+    double getPassbandRippleDb() const {
+        return mPassbandRippleDb;
+    }
+
+    double getNormalizedTransitionBandwidth() const {
+        return mNormalizedTransitionBandwidth;
+    }
+
+    double getFilterAttenuation() const {
+        return mFilterAttenuation;
+    }
+
+    double getNormalizedCutoffFrequency() const {
+        return mNormalizedCutoffFrequency;
+    }
+
 private:
 
     class Constants { // stores the filter constants.
@@ -112,6 +145,8 @@
     void createKaiserFir(Constants &c, double stopBandAtten,
             int inSampleRate, int outSampleRate, double tbwCheat);
 
+    void createKaiserFir(Constants &c, double stopBandAtten, double fcr);
+
     template<int CHANNELS, bool LOCKED, int STRIDE>
     size_t resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
 
@@ -127,6 +162,38 @@
             int32_t mFilterSampleRate; // designed filter sample rate.
         src_quality mFilterQuality;    // designed filter quality.
               void* mCoefBuffer;       // if a filter is created, this is not null
+
+    // Property selected design parameters.
+              // This will enable fixed high quality resampling.
+
+              // 32 char PROP_NAME_MAX limit enforced before Android O
+
+              // Use for sample rates greater than or equal to this value.
+              // Set to non-negative to enable, negative to disable.
+              int32_t mPropertyEnableAtSampleRate = 48000;
+                      // "ro.audio.resampler.psd.enable_at_samplerate"
+
+              // Specify HALF the resampling filter length.
+              // Set to a value which is a multiple of 4.
+              int32_t mPropertyHalfFilterLength = 32;
+                      // "ro.audio.resampler.psd.halflength"
+
+              // Specify the stopband attenuation in positive dB.
+              // Set to a value greater or equal to 20.
+              int32_t mPropertyStopbandAttenuation = 90;
+                      // "ro.audio.resampler.psd.stopband"
+
+              // Specify the cutoff frequency as a percentage of Nyquist.
+              // Set to a value between 50 and 100.
+              int32_t mPropertyCutoffPercent = 100;
+                      // "ro.audio.resampler.psd.cutoff_percent"
+
+    // Filter creation design parameters, see setSampleRate()
+             double mStopbandAttenuationDb = 0.;
+             double mPassbandRippleDb = 0.;
+             double mNormalizedTransitionBandwidth = 0.;
+             double mFilterAttenuation = 0.;
+             double mNormalizedCutoffFrequency = 0.;
 };
 
 } // namespace android

diff --git a/media/libaudioprocessing/AudioResamplerFirGen.h b/media/libaudioprocessing/AudioResamplerFirGen.h
index ad18965..39cafeb 100644
--- a/media/libaudioprocessing/AudioResamplerFirGen.h
+++ b/media/libaudioprocessing/AudioResamplerFirGen.h

@@ -546,8 +546,9 @@
         }
         wstart += wstep;
     }
-    // renormalize - this is only needed for integer filter types
-    double norm = 1./((1ULL<<(sizeof(T)*8-1))*L);
+    // renormalize - this is needed for integer filter types, use 1 for float or double.
+    constexpr int64_t integralShift = std::is_integral<T>::value ? (sizeof(T) * 8 - 1) : 0;
+    const double norm = 1. / (L << integralShift);
 
     firMin = fmin * norm;
     firMax = fmax * norm;
@@ -557,9 +558,12 @@
  * evaluates the |H(f)| lowpass band characteristics.
  *
  * This function tests the lowpass characteristics for the overall polyphase filter,
- * and is used to verify the design.  For this case, fp should be set to the
+ * and is used to verify the design.
+ *
+ * For a polyphase filter (L > 1), typically fp should be set to the
  * passband normalized frequency from 0 to 0.5 for the overall filter (thus it
  * is the designed polyphase bank value / L).  Likewise for fs.
+ * Similarly the stopSteps should be L * passSteps for equivalent accuracy.
  *
  * @param coef is the designed polyphase filter banks
  *
@@ -610,6 +614,74 @@
 }
 
 /*
+ * Estimate the windowed sinc minimum passband value.
+ *
+ * This is the minimum value for a windowed sinc filter in its passband,
+ * which is identical to the scaling required not to cause overflow of a 0dBFS signal.
+ * The actual value used to attenuate the filter amplitude should be slightly
+ * smaller than this (suggest squaring) as this is just an estimate.
+ *
+ * As a windowed sinc has a passband ripple commensurate to the stopband attenuation
+ * due to Gibb's phenomenon from truncating the sinc, we derive this value from
+ * the design stopbandAttenuationDb (a positive value).
+ */
+static inline double computeWindowedSincMinimumPassbandValue(
+        double stopBandAttenuationDb) {
+    return 1. - pow(10. /* base */, stopBandAttenuationDb * (-1. / 20.));
+}
+
+/*
+ * Compute the windowed sinc passband ripple from stopband attenuation.
+ *
+ * As a windowed sinc has an passband ripple commensurate to the stopband attenuation
+ * due to Gibb's phenomenon from truncating the sinc, we derive this value from
+ * the design stopbandAttenuationDb (a positive value).
+ */
+static inline double computeWindowedSincPassbandRippleDb(
+        double stopBandAttenuationDb) {
+    return -20. * log10(computeWindowedSincMinimumPassbandValue(stopBandAttenuationDb));
+}
+
+/*
+ * Kaiser window Beta value
+ *
+ * Formula 3.2.5, 3.2.7, Vaidyanathan, _Multirate Systems and Filter Banks_, p. 48
+ * Formula 7.75, Oppenheim and Schafer, _Discrete-time Signal Processing, 3e_, p. 542
+ *
+ * See also: http://melodi.ee.washington.edu/courses/ee518/notes/lec17.pdf
+ *
+ * Kaiser window and beta parameter
+ *
+ *         | 0.1102*(A - 8.7)                         A > 50
+ *  Beta = | 0.5842*(A - 21)^0.4 + 0.07886*(A - 21)   21 < A <= 50
+ *         | 0.                                       A <= 21
+ *
+ * with A is the desired stop-band attenuation in positive dBFS
+ *
+ *    30 dB    2.210
+ *    40 dB    3.384
+ *    50 dB    4.538
+ *    60 dB    5.658
+ *    70 dB    6.764
+ *    80 dB    7.865
+ *    90 dB    8.960
+ *   100 dB   10.056
+ *
+ * For some values of stopBandAttenuationDb the function may be computed
+ * at compile time.
+ */
+static inline constexpr double computeBeta(double stopBandAttenuationDb) {
+    if (stopBandAttenuationDb > 50.) {
+        return 0.1102 * (stopBandAttenuationDb - 8.7);
+    }
+    const double offset = stopBandAttenuationDb - 21.;
+    if (offset > 0.) {
+        return 0.5842 * pow(offset, 0.4) + 0.07886 * offset;
+    }
+    return 0.;
+}
+
+/*
  * Calculates the overall polyphase filter based on a windowed sinc function.
  *
  * The windowed sinc is an odd length symmetric filter of exactly L*halfNumCoef*2+1
@@ -642,31 +714,8 @@
 template <typename T>
 static inline void firKaiserGen(T* coef, int L, int halfNumCoef,
         double stopBandAtten, double fcr, double atten) {
-    //
-    // Formula 3.2.5, 3.2.7, Vaidyanathan, _Multirate Systems and Filter Banks_, p. 48
-    // Formula 7.75, Oppenheim and Schafer, _Discrete-time Signal Processing, 3e_, p. 542
-    //
-    // See also: http://melodi.ee.washington.edu/courses/ee518/notes/lec17.pdf
-    //
-    // Kaiser window and beta parameter
-    //
-    //         | 0.1102*(A - 8.7)                         A > 50
-    //  beta = | 0.5842*(A - 21)^0.4 + 0.07886*(A - 21)   21 <= A <= 50
-    //         | 0.                                       A < 21
-    //
-    // with A is the desired stop-band attenuation in dBFS
-    //
-    //    30 dB    2.210
-    //    40 dB    3.384
-    //    50 dB    4.538
-    //    60 dB    5.658
-    //    70 dB    6.764
-    //    80 dB    7.865
-    //    90 dB    8.960
-    //   100 dB   10.056
-
     const int N = L * halfNumCoef; // non-negative half
-    const double beta = 0.1102 * (stopBandAtten - 8.7); // >= 50dB always
+    const double beta = computeBeta(stopBandAtten);
     const double xstep = (2. * M_PI) * fcr / L;
     const double xfrac = 1. / N;
     const double yscale = atten * L / (I0(beta) * M_PI);
@@ -696,9 +745,9 @@
                 sg.advance();
             }
 
-            if (is_same<T, int16_t>::value) { // int16_t needs noise shaping
+            if (std::is_same<T, int16_t>::value) { // int16_t needs noise shaping
                 *coef++ = static_cast<T>(toint(y, 1ULL<<(sizeof(T)*8-1), err));
-            } else if (is_same<T, int32_t>::value) {
+            } else if (std::is_same<T, int32_t>::value) {
                 *coef++ = static_cast<T>(toint(y, 1ULL<<(sizeof(T)*8-1)));
             } else { // assumed float or double
                 *coef++ = static_cast<T>(y);

diff --git a/media/libaudioprocessing/BufferProviders.cpp b/media/libaudioprocessing/BufferProviders.cpp
index 862fef6..2d9e1cb 100644
--- a/media/libaudioprocessing/BufferProviders.cpp
+++ b/media/libaudioprocessing/BufferProviders.cpp

@@ -183,7 +183,7 @@
      mOutFrameSize =
              audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask);
      status_t status;
-     status = EffectBufferHalInterface::mirror(
+     status = mEffectsFactory->mirrorBuffer(
              nullptr, mInFrameSize * bufferFrameCount, &mInBuffer);
      if (status != 0) {
          ALOGE("DownmixerBufferProvider() error %d while creating input buffer", status);
@@ -191,7 +191,7 @@
          mEffectsFactory.clear();
          return;
      }
-     status = EffectBufferHalInterface::mirror(
+     status = mEffectsFactory->mirrorBuffer(
              nullptr, mOutFrameSize * bufferFrameCount, &mOutBuffer);
      if (status != 0) {
          ALOGE("DownmixerBufferProvider() error %d while creating output buffer", status);
@@ -376,6 +376,23 @@
     memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannelCount);
 }
 
+ClampFloatBufferProvider::ClampFloatBufferProvider(int32_t channelCount, size_t bufferFrameCount) :
+        CopyBufferProvider(
+                channelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT),
+                channelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT),
+                bufferFrameCount),
+        mChannelCount(channelCount)
+{
+    ALOGV("ClampFloatBufferProvider(%p)(%u)", this, channelCount);
+}
+
+void ClampFloatBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
+{
+    memcpy_to_float_from_float_with_clamping((float*)dst, (const float*)src,
+                                             frames * mChannelCount,
+                                             FLOAT_NOMINAL_RANGE_HEADROOM);
+}
+
 TimestretchBufferProvider::TimestretchBufferProvider(int32_t channelCount,
         audio_format_t format, uint32_t sampleRate, const AudioPlaybackRate &playbackRate) :
         mChannelCount(channelCount),

diff --git a/media/libaudioprocessing/OWNERS b/media/libaudioprocessing/OWNERS
new file mode 100644
index 0000000..96d0ea0
--- /dev/null
+++ b/media/libaudioprocessing/OWNERS

@@ -0,0 +1,3 @@
+gkasten@google.com
+hunga@google.com
+rago@google.com

diff --git a/media/libaudioprocessing/tests/build_and_run_all_unit_tests.sh b/media/libaudioprocessing/tests/build_and_run_all_unit_tests.sh
index 704d095..efef417 100755
--- a/media/libaudioprocessing/tests/build_and_run_all_unit_tests.sh
+++ b/media/libaudioprocessing/tests/build_and_run_all_unit_tests.sh

@@ -14,8 +14,8 @@
 
 echo "waiting for device"
 adb root && adb wait-for-device remount
-adb push $OUT/system/lib/libaudioresampler.so /system/lib
-adb push $OUT/system/lib64/libaudioresampler.so /system/lib64
+adb push $OUT/system/lib/libaudioprocessing.so /system/lib
+adb push $OUT/system/lib64/libaudioprocessing.so /system/lib64
 adb push $OUT/data/nativetest/resampler_tests/resampler_tests /data/nativetest/resampler_tests/resampler_tests
 adb push $OUT/data/nativetest64/resampler_tests/resampler_tests /data/nativetest64/resampler_tests/resampler_tests
 

diff --git a/media/libaudioprocessing/tests/resampler_tests.cpp b/media/libaudioprocessing/tests/resampler_tests.cpp
index a23c000..e1623f7 100644
--- a/media/libaudioprocessing/tests/resampler_tests.cpp
+++ b/media/libaudioprocessing/tests/resampler_tests.cpp

@@ -29,6 +29,7 @@
 #include <unistd.h>
 
 #include <iostream>
+#include <memory>
 #include <utility>
 #include <vector>
 
@@ -37,6 +38,8 @@
 #include <media/AudioBufferProvider.h>
 
 #include <media/AudioResampler.h>
+#include "../AudioResamplerDyn.h"
+#include "../AudioResamplerFirGen.h"
 #include "test_utils.h"
 
 template <typename T>
@@ -242,6 +245,60 @@
     delete resampler;
 }
 
+void testFilterResponse(
+        size_t channels, unsigned inputFreq, unsigned outputFreq)
+{
+    // create resampler
+    using ResamplerType = android::AudioResamplerDyn<float, float, float>;
+    std::unique_ptr<ResamplerType> rdyn(
+            static_cast<ResamplerType *>(
+                    android::AudioResampler::create(
+                            AUDIO_FORMAT_PCM_FLOAT,
+                            channels,
+                            outputFreq,
+                            android::AudioResampler::DYN_HIGH_QUALITY)));
+    rdyn->setSampleRate(inputFreq);
+
+    // get design parameters
+    const int phases = rdyn->getPhases();
+    const int halfLength = rdyn->getHalfLength();
+    const float *coefs = rdyn->getFilterCoefs();
+    const double fcr = rdyn->getNormalizedCutoffFrequency();
+    const double tbw = rdyn->getNormalizedTransitionBandwidth();
+    const double attenuation = rdyn->getFilterAttenuation();
+    const double stopbandDb = rdyn->getStopbandAttenuationDb();
+    const double passbandDb = rdyn->getPassbandRippleDb();
+    const double fp = fcr - tbw / 2;
+    const double fs = fcr + tbw / 2;
+
+    printf("inputFreq:%d outputFreq:%d design"
+            " phases:%d halfLength:%d"
+            " fcr:%lf fp:%lf fs:%lf tbw:%lf"
+            " attenuation:%lf stopRipple:%.lf passRipple:%lf"
+            "\n",
+            inputFreq, outputFreq,
+            phases, halfLength,
+            fcr, fp, fs, tbw,
+            attenuation, stopbandDb, passbandDb);
+
+    // verify design parameters
+    constexpr int32_t passSteps = 1000;
+    double passMin, passMax, passRipple, stopMax, stopRipple;
+    android::testFir(coefs, phases, halfLength, fp / phases, fs / phases,
+            passSteps, phases * passSteps /* stopSteps */,
+            passMin, passMax, passRipple,
+            stopMax, stopRipple);
+    printf("inputFreq:%d outputFreq:%d verify"
+            " passMin:%lf passMax:%lf passRipple:%lf stopMax:%lf stopRipple:%lf"
+            "\n",
+            inputFreq, outputFreq,
+            passMin, passMax, passRipple, stopMax, stopRipple);
+
+    ASSERT_GT(stopRipple, 60.);  // enough stopband attenuation
+    ASSERT_LT(passRipple, 0.2);  // small passband ripple
+    ASSERT_GT(passMin, 0.99);    // we do not attenuate the signal (ideally 1.)
+}
+
 /* Buffer increment test
  *
  * We compare a reference output, where we consume and process the entire
@@ -484,3 +541,30 @@
     }
 }
 
+TEST(audioflinger_resampler, filterresponse) {
+    std::vector<int> inSampleRates{
+        8000,
+        11025,
+        12000,
+        16000,
+        22050,
+        24000,
+        32000,
+        44100,
+        48000,
+        88200,
+        96000,
+        176400,
+        192000,
+    };
+    std::vector<int> outSampleRates{
+        48000,
+        96000,
+    };
+
+    for (int outSampleRate : outSampleRates) {
+        for (int inSampleRate : inSampleRates) {
+            testFilterResponse(2 /* channels */, inSampleRate, outSampleRate);
+        }
+    }
+}

diff --git a/media/libaudioprocessing/tests/test-mixer.cpp b/media/libaudioprocessing/tests/test-mixer.cpp
index 75dbf91..bc9d2a6 100644
--- a/media/libaudioprocessing/tests/test-mixer.cpp
+++ b/media/libaudioprocessing/tests/test-mixer.cpp

@@ -143,10 +143,6 @@
         usage(progname);
         return EXIT_FAILURE;
     }
-    if ((unsigned)argc > AudioMixer::MAX_NUM_TRACKS) {
-        fprintf(stderr, "too many tracks: %d > %u", argc, AudioMixer::MAX_NUM_TRACKS);
-        return EXIT_FAILURE;
-    }
 
     size_t outputFrames = 0;
 
@@ -246,9 +242,10 @@
     for (size_t i = 0; i < providers.size(); ++i) {
         //printf("track %d out of %d\n", i, providers.size());
         uint32_t channelMask = audio_channel_out_mask_from_count(providers[i].getNumChannels());
-        int32_t name = mixer->getTrackName(channelMask,
-                formats[i], AUDIO_SESSION_OUTPUT_MIX);
-        ALOG_ASSERT(name >= 0);
+        const int name = i;
+        const status_t status = mixer->create(
+                name, channelMask, formats[i], AUDIO_SESSION_OUTPUT_MIX);
+        LOG_ALWAYS_FATAL_IF(status != OK);
         names[i] = name;
         mixer->setBufferProvider(name, &providers[i]);
         mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
@@ -315,9 +312,10 @@
     writeFile(outputFilename, outputAddr,
             outputSampleRate, outputChannels, outputFrames, useMixerFloat);
     if (auxFilename) {
-        // Aux buffer is always in q4_27 format for now.
-        // memcpy_to_i16_from_q4_27(), but with stereo frame count (not sample count)
-        ditherAndClamp((int32_t*)auxAddr, (int32_t*)auxAddr, outputFrames >> 1);
+        // Aux buffer is always in q4_27 format for O and earlier.
+        // memcpy_to_i16_from_q4_27((int16_t*)auxAddr, (const int32_t*)auxAddr, outputFrames);
+        // Aux buffer is always in float format for P.
+        memcpy_to_i16_from_float((int16_t*)auxAddr, (const float*)auxAddr, outputFrames);
         writeFile(auxFilename, auxAddr, outputSampleRate, 1, outputFrames, false);
     }
 

diff --git a/media/libcpustats/OWNERS b/media/libcpustats/OWNERS
new file mode 100644
index 0000000..f9cb567
--- /dev/null
+++ b/media/libcpustats/OWNERS

@@ -0,0 +1 @@
+gkasten@google.com

diff --git a/media/libeffects/OWNERS b/media/libeffects/OWNERS
index 7e3de13..7f9ae81 100644
--- a/media/libeffects/OWNERS
+++ b/media/libeffects/OWNERS

@@ -1,3 +1,4 @@
+hunga@google.com
 krocard@google.com
 mnaganov@google.com
 rago@google.com

diff --git a/media/libeffects/config/Android.bp b/media/libeffects/config/Android.bp
index 4398a91..5fa9da9 100644
--- a/media/libeffects/config/Android.bp
+++ b/media/libeffects/config/Android.bp

@@ -1,10 +1,15 @@
 // Effect configuration
-cc_library_shared {
+cc_library {
     name: "libeffectsconfig",
     vendor_available: true,
 
     srcs: ["src/EffectsConfig.cpp"],
 
+    cflags: [
+        "-Wall",
+        "-Werror",
+    ],
+
     shared_libs: [
         "liblog",
         "libtinyxml2",

diff --git a/media/libeffects/config/include/media/EffectsConfig.h b/media/libeffects/config/include/media/EffectsConfig.h
index 811730c..fa0415b 100644
--- a/media/libeffects/config/include/media/EffectsConfig.h
+++ b/media/libeffects/config/include/media/EffectsConfig.h

@@ -32,8 +32,13 @@
 namespace android {
 namespace effectsConfig {
 
-/** Default path of effect configuration file. */
-constexpr char DEFAULT_PATH[] = "/vendor/etc/audio_effects.xml";
+/** Default path of effect configuration file. Relative to DEFAULT_LOCATIONS. */
+constexpr const char* DEFAULT_NAME = "audio_effects.xml";
+
+/** Default path of effect configuration file.
+ * The /vendor partition is the recommended one, the others are deprecated.
+ */
+constexpr const char* DEFAULT_LOCATIONS[] = {"/odm/etc", "/vendor/etc", "/system/etc"};
 
 /** Directories where the effect libraries will be search for. */
 constexpr const char* LD_EFFECT_LIBRARY_PATH[] =
@@ -91,13 +96,16 @@
     /** Parsed config, nullptr if the xml lib could not load the file */
     std::unique_ptr<Config> parsedConfig;
     size_t nbSkippedElement; //< Number of skipped invalid library, effect or processing chain
+    const std::string configPath; //< Path to the loaded configuration
 };
 
 /** Parses the provided effect configuration.
  * Parsing do not stop of first invalid element, but continues to the next.
+ * @param[in] path of the configuration file do load
+ *                 if nullptr, look for DEFAULT_NAME in DEFAULT_LOCATIONS.
  * @see ParsingResult::nbSkippedElement
  */
-ParsingResult parse(const char* path = DEFAULT_PATH);
+ParsingResult parse(const char* path = nullptr);
 
 } // namespace effectsConfig
 } // namespace android

diff --git a/media/libeffects/config/src/EffectsConfig.cpp b/media/libeffects/config/src/EffectsConfig.cpp
index 97462f8..351b1ee 100644
--- a/media/libeffects/config/src/EffectsConfig.cpp
+++ b/media/libeffects/config/src/EffectsConfig.cpp

@@ -20,6 +20,7 @@
 #include <cstdint>
 #include <functional>
 #include <string>
+#include <unistd.h>
 
 #include <tinyxml2.h>
 #include <log/log.h>
@@ -85,7 +86,7 @@
 constexpr std::enable_if<false, Enum> STREAM_NAME_MAP;
 
 /** All output stream types which support effects.
- * This need to be kept in sink with the xsd streamOutputType.
+ * This need to be kept in sync with the xsd streamOutputType.
  */
 template <>
 constexpr std::pair<audio_stream_type_t, const char*> STREAM_NAME_MAP<audio_stream_type_t>[] = {
@@ -102,7 +103,7 @@
 };
 
 /** All input stream types which support effects.
- * This need to be kept in sink with the xsd streamOutputType.
+ * This need to be kept in sync with the xsd streamOutputType.
  */
 template <>
 constexpr std::pair<audio_source_t, const char*> STREAM_NAME_MAP<audio_source_t>[] = {
@@ -142,7 +143,7 @@
 }
 
 /** Find an element in a collection by its name.
- * @return nullptr if not found, the ellements address if found.
+ * @return nullptr if not found, the element address if found.
  */
 template <class T>
 T* findByName(const char* name, std::vector<T>& collection) {
@@ -202,7 +203,7 @@
         auto parseProxy = [&xmlEffect, &parseImpl](const char* tag, EffectImpl& proxyLib) {
             auto* xmlProxyLib = xmlEffect.FirstChildElement(tag);
             if (xmlProxyLib == nullptr) {
-                ALOGE("effectProxy must contain a <%s>: %s", tag, dump(*xmlProxyLib));
+                ALOGE("effectProxy must contain a <%s>: %s", tag, dump(xmlEffect));
                 return false;
             }
             return parseImpl(*xmlProxyLib, proxyLib);
@@ -249,15 +250,14 @@
     return true;
 }
 
-}; // namespace
-
-ParsingResult parse(const char* path) {
+/** Internal version of the public parse(const char* path) where path always exist. */
+ParsingResult parseWithPath(std::string&& path) {
     XMLDocument doc;
-    doc.LoadFile(path);
+    doc.LoadFile(path.c_str());
     if (doc.Error()) {
-        ALOGE("Failed to parse %s: Tinyxml2 error (%d): %s %s", path,
-              doc.ErrorID(), doc.GetErrorStr1(), doc.GetErrorStr2());
-        return {nullptr, 0};
+        ALOGE("Failed to parse %s: Tinyxml2 error (%d): %s", path.c_str(),
+              doc.ErrorID(), doc.ErrorStr());
+        return {nullptr, 0, std::move(path)};
     }
 
     auto config = std::make_unique<Config>();
@@ -295,7 +295,29 @@
             }
         }
     }
-    return {std::move(config), nbSkippedElements};
+    return {std::move(config), nbSkippedElements, std::move(path)};
+}
+
+}; // namespace
+
+ParsingResult parse(const char* path) {
+    if (path != nullptr) {
+        return parseWithPath(path);
+    }
+
+    for (std::string location : DEFAULT_LOCATIONS) {
+        std::string defaultPath = location + '/' + DEFAULT_NAME;
+        if (access(defaultPath.c_str(), R_OK) != 0) {
+            continue;
+        }
+        auto result = parseWithPath(std::move(defaultPath));
+        if (result.parsedConfig != nullptr) {
+            return result;
+        }
+    }
+
+    ALOGE("Could not parse effect configuration in any of the default locations.");
+    return {nullptr, 0, ""};
 }
 
 } // namespace effectsConfig

diff --git a/media/libeffects/data/audio_effects.conf b/media/libeffects/data/audio_effects.conf
index 14a171b..dd729c5 100644
--- a/media/libeffects/data/audio_effects.conf
+++ b/media/libeffects/data/audio_effects.conf

@@ -38,6 +38,9 @@
   loudness_enhancer {
     path /vendor/lib/soundfx/libldnhncr.so
   }
+  dynamics_processing {
+    path /vendor/lib/soundfx/libdynproc.so
+  }
 }
 
 # Default pre-processing library. Add to audio_effect.conf "libraries" section if
@@ -129,6 +132,10 @@
     library loudness_enhancer
     uuid fa415329-2034-4bea-b5dc-5b381c8d1e2c
   }
+  dynamics_processing {
+    library dynamics_processing
+    uuid e0e6539b-1781-7261-676f-6d7573696340
+  }
 }
 
 # Default pre-processing effects. Add to audio_effect.conf "effects" section if

diff --git a/media/libeffects/data/audio_effects.xml b/media/libeffects/data/audio_effects.xml
new file mode 100644
index 0000000..3f85052
--- /dev/null
+++ b/media/libeffects/data/audio_effects.xml

@@ -0,0 +1,102 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<audio_effects_conf version="2.0" xmlns="http://schemas.android.com/audio/audio_effects_conf/v2_0">
+    <!-- List of effect libraries to load.
+         Each library element must contain a "name" attribute and a "path" attribute giving the
+         name of a library .so file in /vendor/lib/soundfx on the target
+
+         If offloadable effects are present, the AOSP library libeffectproxy.so must be listed as
+         well as one library for the SW implementation and one library for the DSP implementation:
+         <library name="proxy" path="libeffectproxy.so"/>
+         <library name="some_fx_sw" path="lib_some_fx_sw.so"/>
+         <library name="some_fx_hw" path="lib_some_fx_hw.so"/>
+
+         If the audio HAL implements support for AOSP software audio pre-processing effects,
+         the following library must be added:
+         <library name="pre_processing" path="libaudiopreprocessing.so"/>
+    -->
+    <libraries>
+        <library name="bundle" path="libbundlewrapper.so"/>
+        <library name="reverb" path="libreverbwrapper.so"/>
+        <library name="visualizer" path="libvisualizer.so"/>
+        <library name="downmix" path="libdownmix.so"/>
+        <library name="loudness_enhancer" path="libldnhncr.so"/>
+        <library name="dynamics_processing" path="libdynproc.so"/>
+    </libraries>
+
+    <!-- list of effects to load.
+         Each "effect" element must contain a "name", "library" and a "uuid" attribute.
+         The value of the "library" element must correspond to the name of one library element in
+         the "libraries" element.
+         The "name" attribute is indicative, only the value of the "uuid" attribute designates
+         the effect.
+         The uuid is the implementation specific UUID as specified by the effect vendor. This is not
+         the generic effect type UUID.
+
+         Offloadable effects are described by an "effectProxy" element which contains one "libsw"
+         element containing the "uuid" and "library" for the SW implementation and one "libhw"
+         element containing the "uuid" and "library" for the DSP implementation.
+         The "uuid" value for the "effectProxy" element must be unique and will override the default
+         uuid in the AOSP proxy effect implementation.
+
+         If the audio HAL implements support for AOSP software audio pre-processing effects,
+         the following effects can be added:
+         <effect name="agc" library="pre_processing" uuid="aa8130e0-66fc-11e0-bad0-0002a5d5c51b"/>
+         <effect name="aec" library="pre_processing" uuid="bb392ec0-8d4d-11e0-a896-0002a5d5c51b"/>
+         <effect name="ns" library="pre_processing" uuid="c06c8400-8e06-11e0-9cb6-0002a5d5c51b"/>
+    -->
+
+    <effects>
+        <effect name="bassboost" library="bundle" uuid="8631f300-72e2-11df-b57e-0002a5d5c51b"/>
+        <effect name="virtualizer" library="bundle" uuid="1d4033c0-8557-11df-9f2d-0002a5d5c51b"/>
+        <effect name="equalizer" library="bundle" uuid="ce772f20-847d-11df-bb17-0002a5d5c51b"/>
+        <effect name="volume" library="bundle" uuid="119341a0-8469-11df-81f9-0002a5d5c51b"/>
+        <effect name="reverb_env_aux" library="reverb" uuid="4a387fc0-8ab3-11df-8bad-0002a5d5c51b"/>
+        <effect name="reverb_env_ins" library="reverb" uuid="c7a511a0-a3bb-11df-860e-0002a5d5c51b"/>
+        <effect name="reverb_pre_aux" library="reverb" uuid="f29a1400-a3bb-11df-8ddc-0002a5d5c51b"/>
+        <effect name="reverb_pre_ins" library="reverb" uuid="172cdf00-a3bc-11df-a72f-0002a5d5c51b"/>
+        <effect name="visualizer" library="visualizer" uuid="d069d9e0-8329-11df-9168-0002a5d5c51b"/>
+        <effect name="downmix" library="downmix" uuid="93f04452-e4fe-41cc-91f9-e475b6d1d69f"/>
+        <effect name="loudness_enhancer" library="loudness_enhancer" uuid="fa415329-2034-4bea-b5dc-5b381c8d1e2c"/>
+        <effect name="dynamics_processing" library="dynamics_processing" uuid="e0e6539b-1781-7261-676f-6d7573696340"/>
+    </effects>
+
+    <!-- Audio pre processor configurations.
+         The pre processor configuration is described in a "preprocess" element and consists in a
+         list of elements each describing pre processor settings for a given use case or "stream".
+         Each stream element has a "type" attribute corresponding to the input source used.
+         Valid types are:
+              "mic", "camcorder", "voice_recognition", "voice_communication"
+         Each "stream" element contains a list of "apply" elements indicating one effect to apply.
+         The effect to apply is designated by its name in the "effects" elements.
+
+        <preprocess>
+            <stream type="voice_communication">
+                <apply effect="aec"/>
+                <apply effect="ns"/>
+            </stream>
+        </preprocess>
+    -->
+
+    <!-- Audio post processor configurations.
+         The post processor configuration is described in a "postprocess" element and consists in a
+         list of elements each describing post processor settings for a given use case or "stream".
+         Each stream element has a "type" attribute corresponding to the stream type used.
+         Valid types are:
+              "music", "ring", "alarm", "notification", "voice_call"
+         Each "stream" element contains a list of "apply" elements indicating one effect to apply.
+         The effect to apply is designated by its name in the "effects" elements.
+
+        <postprocess>
+            <stream type="music">
+                <apply effect="music_post_proc"/>
+            </stream>
+            <stream type="voice_call">
+                <apply effect="voice_post_proc"/>
+            </stream>
+            <stream type="notification">
+                <apply effect="notification_post_proc"/>
+            </stream>
+        </postprocess>
+    -->
+
+</audio_effects_conf>

diff --git a/media/libeffects/dynamicsproc/Android.mk b/media/libeffects/dynamicsproc/Android.mk
new file mode 100644
index 0000000..7be0c49
--- /dev/null
+++ b/media/libeffects/dynamicsproc/Android.mk

@@ -0,0 +1,43 @@
+# Copyright (C) 2018 The Android Open Source Project
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#      http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+LOCAL_PATH:= $(call my-dir)
+
+# DynamicsProcessing library
+include $(CLEAR_VARS)
+
+LOCAL_VENDOR_MODULE := true
+
+EIGEN_PATH := external/eigen
+LOCAL_C_INCLUDES += $(EIGEN_PATH)
+
+LOCAL_SRC_FILES:= \
+    EffectDynamicsProcessing.cpp \
+    dsp/DPBase.cpp \
+    dsp/DPFrequency.cpp
+
+LOCAL_CFLAGS+= -O2 -fvisibility=hidden
+LOCAL_CFLAGS += -Wall -Werror
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    liblog \
+
+LOCAL_MODULE_RELATIVE_PATH := soundfx
+LOCAL_MODULE:= libdynproc
+
+LOCAL_HEADER_LIBRARIES := \
+    libaudioeffects
+
+include $(BUILD_SHARED_LIBRARY)

diff --git a/media/libeffects/dynamicsproc/EffectDynamicsProcessing.cpp b/media/libeffects/dynamicsproc/EffectDynamicsProcessing.cpp
new file mode 100644
index 0000000..0b883f1
--- /dev/null
+++ b/media/libeffects/dynamicsproc/EffectDynamicsProcessing.cpp

@@ -0,0 +1,1300 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "EffectDP"
+//#define LOG_NDEBUG 0
+
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+#include <time.h>
+#include <new>
+
+#include <log/log.h>
+
+#include <audio_effects/effect_dynamicsprocessing.h>
+#include <dsp/DPBase.h>
+#include <dsp/DPFrequency.h>
+
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while (false)
+#endif
+
+// union to hold command values
+using value_t = union {
+    int32_t i;
+    float f;
+};
+
+// effect_handle_t interface implementation for DP effect
+extern const struct effect_interface_s gDPInterface;
+
+// AOSP Dynamics Processing UUID: e0e6539b-1781-7261-676f-6d7573696340
+const effect_descriptor_t gDPDescriptor = {
+        {0x7261676f, 0x6d75, 0x7369, 0x6364, {0x28, 0xe2, 0xfd, 0x3a, 0xc3, 0x9e}}, // type
+        {0xe0e6539b, 0x1781, 0x7261, 0x676f, {0x6d, 0x75, 0x73, 0x69, 0x63, 0x40}}, // uuid
+        EFFECT_CONTROL_API_VERSION,
+        (EFFECT_FLAG_TYPE_INSERT | EFFECT_FLAG_INSERT_LAST | EFFECT_FLAG_VOLUME_CTRL),
+        0, // TODO
+        1,
+        "Dynamics Processing",
+        "The Android Open Source Project",
+};
+
+enum dp_state_e {
+    DYNAMICS_PROCESSING_STATE_UNINITIALIZED,
+    DYNAMICS_PROCESSING_STATE_INITIALIZED,
+    DYNAMICS_PROCESSING_STATE_ACTIVE,
+};
+
+struct DynamicsProcessingContext {
+    const struct effect_interface_s *mItfe;
+    effect_config_t mConfig;
+    uint8_t mState;
+
+    dp_fx::DPBase * mPDynamics; //the effect (or current effect)
+    int32_t mCurrentVariant;
+    float mPreferredFrameDuration;
+};
+
+// The value offset of an effect parameter is computed by rounding up
+// the parameter size to the next 32 bit alignment.
+static inline uint32_t computeParamVOffset(const effect_param_t *p) {
+    return ((p->psize + sizeof(int32_t) - 1) / sizeof(int32_t)) *
+            sizeof(int32_t);
+}
+
+//--- local function prototypes
+int DP_setParameter(DynamicsProcessingContext *pContext,
+        uint32_t paramSize,
+        void *pParam,
+        uint32_t valueSize,
+        void *pValue);
+int DP_getParameter(DynamicsProcessingContext *pContext,
+        uint32_t paramSize,
+        void *pParam,
+        uint32_t *pValueSize,
+        void *pValue);
+int DP_getParameterCmdSize(uint32_t paramSize,
+        void *pParam);
+void DP_expectedParamValueSizes(uint32_t paramSize,
+        void *pParam,
+        bool isSet,
+        uint32_t *pCmdSize,
+        uint32_t *pValueSize);
+//
+//--- Local functions (not directly used by effect interface)
+//
+
+void DP_reset(DynamicsProcessingContext *pContext)
+{
+    ALOGV("> DP_reset(%p)", pContext);
+    if (pContext->mPDynamics != NULL) {
+        pContext->mPDynamics->reset();
+    } else {
+        ALOGE("DP_reset(%p): null DynamicsProcessing", pContext);
+    }
+}
+
+//----------------------------------------------------------------------------
+// DP_setConfig()
+//----------------------------------------------------------------------------
+// Purpose: Set input and output audio configuration.
+//
+// Inputs:
+//  pContext:   effect engine context
+//  pConfig:    pointer to effect_config_t structure holding input and output
+//      configuration parameters
+//
+// Outputs:
+//
+//----------------------------------------------------------------------------
+
+int DP_setConfig(DynamicsProcessingContext *pContext, effect_config_t *pConfig)
+{
+    ALOGV("DP_setConfig(%p)", pContext);
+
+    if (pConfig->inputCfg.samplingRate != pConfig->outputCfg.samplingRate) return -EINVAL;
+    if (pConfig->inputCfg.channels != pConfig->outputCfg.channels) return -EINVAL;
+    if (pConfig->inputCfg.format != pConfig->outputCfg.format) return -EINVAL;
+    if (pConfig->outputCfg.accessMode != EFFECT_BUFFER_ACCESS_WRITE &&
+            pConfig->outputCfg.accessMode != EFFECT_BUFFER_ACCESS_ACCUMULATE) return -EINVAL;
+    if (pConfig->inputCfg.format != AUDIO_FORMAT_PCM_FLOAT) return -EINVAL;
+
+    pContext->mConfig = *pConfig;
+
+    DP_reset(pContext);
+
+    return 0;
+}
+
+//----------------------------------------------------------------------------
+// DP_getConfig()
+//----------------------------------------------------------------------------
+// Purpose: Get input and output audio configuration.
+//
+// Inputs:
+//  pContext:   effect engine context
+//  pConfig:    pointer to effect_config_t structure holding input and output
+//      configuration parameters
+//
+// Outputs:
+//
+//----------------------------------------------------------------------------
+
+void DP_getConfig(DynamicsProcessingContext *pContext, effect_config_t *pConfig)
+{
+    *pConfig = pContext->mConfig;
+}
+
+//----------------------------------------------------------------------------
+// DP_init()
+//----------------------------------------------------------------------------
+// Purpose: Initialize engine with default configuration.
+//
+// Inputs:
+//  pContext:   effect engine context
+//
+// Outputs:
+//
+//----------------------------------------------------------------------------
+
+int DP_init(DynamicsProcessingContext *pContext)
+{
+    ALOGV("DP_init(%p)", pContext);
+
+    pContext->mItfe = &gDPInterface;
+    pContext->mPDynamics = NULL;
+    pContext->mState = DYNAMICS_PROCESSING_STATE_UNINITIALIZED;
+
+    pContext->mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
+    pContext->mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
+    pContext->mConfig.inputCfg.format = AUDIO_FORMAT_PCM_FLOAT;
+    pContext->mConfig.inputCfg.samplingRate = 48000;
+    pContext->mConfig.inputCfg.bufferProvider.getBuffer = NULL;
+    pContext->mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
+    pContext->mConfig.inputCfg.bufferProvider.cookie = NULL;
+    pContext->mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
+    pContext->mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
+    pContext->mConfig.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
+    pContext->mConfig.outputCfg.format = AUDIO_FORMAT_PCM_FLOAT;
+    pContext->mConfig.outputCfg.samplingRate = 48000;
+    pContext->mConfig.outputCfg.bufferProvider.getBuffer = NULL;
+    pContext->mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
+    pContext->mConfig.outputCfg.bufferProvider.cookie = NULL;
+    pContext->mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
+
+    pContext->mCurrentVariant = -1; //none
+    pContext->mPreferredFrameDuration = 0; //none
+
+    DP_setConfig(pContext, &pContext->mConfig);
+    pContext->mState = DYNAMICS_PROCESSING_STATE_INITIALIZED;
+    return 0;
+}
+
+void DP_changeVariant(DynamicsProcessingContext *pContext, int newVariant) {
+    ALOGV("DP_changeVariant from %d to %d", pContext->mCurrentVariant, newVariant);
+    switch(newVariant) {
+    case VARIANT_FAVOR_FREQUENCY_RESOLUTION: {
+        pContext->mCurrentVariant = VARIANT_FAVOR_FREQUENCY_RESOLUTION;
+        delete pContext->mPDynamics;
+        pContext->mPDynamics = new dp_fx::DPFrequency();
+        break;
+    }
+    default: {
+        ALOGW("DynamicsProcessing variant %d not available for creation", newVariant);
+        break;
+    }
+    } //switch
+}
+
+static inline bool isPowerOf2(unsigned long n) {
+    return (n & (n - 1)) == 0;
+}
+
+void DP_configureVariant(DynamicsProcessingContext *pContext, int newVariant) {
+    ALOGV("DP_configureVariant %d", newVariant);
+    switch(newVariant) {
+    case VARIANT_FAVOR_FREQUENCY_RESOLUTION: {
+        int32_t minBlockSize = (int32_t)dp_fx::DPFrequency::getMinBockSize();
+        int32_t desiredBlock = pContext->mPreferredFrameDuration *
+                pContext->mConfig.inputCfg.samplingRate / 1000.0f;
+        int32_t currentBlock = desiredBlock;
+        ALOGV(" sampling rate: %d, desiredBlock size %0.2f (%d) samples",
+                pContext->mConfig.inputCfg.samplingRate, pContext->mPreferredFrameDuration,
+                desiredBlock);
+        if (desiredBlock < minBlockSize) {
+            currentBlock = minBlockSize;
+        } else if (!isPowerOf2(desiredBlock)) {
+            //find next highest power of 2.
+            currentBlock = 1 << (32 - __builtin_clz(desiredBlock));
+        }
+        ((dp_fx::DPFrequency*)pContext->mPDynamics)->configure(currentBlock,
+                currentBlock/2,
+                pContext->mConfig.inputCfg.samplingRate);
+        break;
+    }
+    default: {
+        ALOGE("DynamicsProcessing variant %d not available to configure", newVariant);
+        break;
+    }
+    }
+}
+
+//
+//--- Effect Library Interface Implementation
+//
+
+int DPLib_Release(effect_handle_t handle) {
+    DynamicsProcessingContext * pContext = (DynamicsProcessingContext *)handle;
+
+    ALOGV("DPLib_Release %p", handle);
+    if (pContext == NULL) {
+        return -EINVAL;
+    }
+    delete pContext->mPDynamics;
+    delete pContext;
+
+    return 0;
+}
+
+int DPLib_Create(const effect_uuid_t *uuid,
+                         int32_t sessionId __unused,
+                         int32_t ioId __unused,
+                         effect_handle_t *pHandle) {
+    ALOGV("DPLib_Create()");
+
+    if (pHandle == NULL || uuid == NULL) {
+        return -EINVAL;
+    }
+
+    if (memcmp(uuid, &gDPDescriptor.uuid, sizeof(*uuid)) != 0) {
+        return -EINVAL;
+    }
+
+    DynamicsProcessingContext *pContext = new DynamicsProcessingContext;
+    *pHandle = (effect_handle_t)pContext;
+    int ret = DP_init(pContext);
+    if (ret < 0) {
+        ALOGW("DPLib_Create() init failed");
+        DPLib_Release(*pHandle);
+        return ret;
+    }
+
+    ALOGV("DPLib_Create context is %p", pContext);
+    return 0;
+}
+
+int DPLib_GetDescriptor(const effect_uuid_t *uuid,
+                                effect_descriptor_t *pDescriptor) {
+
+    if (pDescriptor == NULL || uuid == NULL){
+        ALOGE("DPLib_GetDescriptor() called with NULL pointer");
+        return -EINVAL;
+    }
+
+    if (memcmp(uuid, &gDPDescriptor.uuid, sizeof(*uuid)) == 0) {
+        *pDescriptor = gDPDescriptor;
+        return 0;
+    }
+
+    return -EINVAL;
+} /* end DPLib_GetDescriptor */
+
+//
+//--- Effect Control Interface Implementation
+//
+int DP_process(effect_handle_t self, audio_buffer_t *inBuffer,
+        audio_buffer_t *outBuffer) {
+    DynamicsProcessingContext * pContext = (DynamicsProcessingContext *)self;
+
+    if (pContext == NULL) {
+        ALOGE("DP_process() called with NULL context");
+        return -EINVAL;
+    }
+
+    if (inBuffer == NULL || inBuffer->raw == NULL ||
+        outBuffer == NULL || outBuffer->raw == NULL ||
+        inBuffer->frameCount != outBuffer->frameCount ||
+        inBuffer->frameCount == 0) {
+        ALOGE("inBuffer or outBuffer are NULL or have problems with frame count");
+        return -EINVAL;
+    }
+    if (pContext->mState != DYNAMICS_PROCESSING_STATE_ACTIVE) {
+        ALOGE("mState is not DYNAMICS_PROCESSING_STATE_ACTIVE. Current mState %d",
+                pContext->mState);
+        return -ENODATA;
+    }
+    //if dynamics exist...
+    if (pContext->mPDynamics != NULL) {
+        int32_t channelCount = (int32_t)audio_channel_count_from_out_mask(
+                        pContext->mConfig.inputCfg.channels);
+        pContext->mPDynamics->processSamples(inBuffer->f32, inBuffer->f32,
+                inBuffer->frameCount * channelCount);
+
+        if (inBuffer->raw != outBuffer->raw) {
+            if (pContext->mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
+                for (size_t i = 0; i < outBuffer->frameCount * channelCount; i++) {
+                    outBuffer->f32[i] += inBuffer->f32[i];
+                }
+            } else {
+                memcpy(outBuffer->raw, inBuffer->raw,
+                        outBuffer->frameCount * channelCount * sizeof(float));
+            }
+        }
+    } else {
+        //do nothing. no effect created yet. warning.
+        ALOGW("Warning: no DynamicsProcessing engine available");
+        return -EINVAL;
+    }
+    return 0;
+}
+
+//helper function
+bool DP_checkSizesInt(uint32_t paramSize, uint32_t valueSize, uint32_t expectedParams,
+        uint32_t expectedValues) {
+    if (paramSize < expectedParams * sizeof(int32_t)) {
+        ALOGE("Invalid paramSize: %u expected %u", paramSize,
+                (uint32_t)(expectedParams * sizeof(int32_t)));
+        return false;
+    }
+    if (valueSize < expectedValues * sizeof(int32_t)) {
+        ALOGE("Invalid valueSize %u expected %u", valueSize,
+                (uint32_t)(expectedValues * sizeof(int32_t)));
+        return false;
+    }
+    return true;
+}
+
+static dp_fx::DPChannel* DP_getChannel(DynamicsProcessingContext *pContext,
+        int32_t channel) {
+    if (pContext->mPDynamics == NULL) {
+        return NULL;
+    }
+    dp_fx::DPChannel *pChannel = pContext->mPDynamics->getChannel(channel);
+    ALOGE_IF(pChannel == NULL, "DPChannel NULL. invalid channel %d", channel);
+    return pChannel;
+}
+
+static dp_fx::DPEq* DP_getEq(DynamicsProcessingContext *pContext, int32_t channel,
+        int32_t eqType) {
+    dp_fx::DPChannel *pChannel = DP_getChannel(pContext, channel);
+    if (pChannel == NULL) {
+        return NULL;
+    }
+    dp_fx::DPEq *pEq = (eqType == DP_PARAM_PRE_EQ ? pChannel->getPreEq() :
+            (eqType == DP_PARAM_POST_EQ ? pChannel->getPostEq() : NULL));
+    ALOGE_IF(pEq == NULL,"DPEq NULL invalid eq");
+    return pEq;
+}
+
+static dp_fx::DPEqBand* DP_getEqBand(DynamicsProcessingContext *pContext, int32_t channel,
+        int32_t eqType, int32_t band) {
+    dp_fx::DPEq *pEq = DP_getEq(pContext, channel, eqType);
+    if (pEq == NULL) {
+        return NULL;
+    }
+    dp_fx::DPEqBand *pEqBand = pEq->getBand(band);
+    ALOGE_IF(pEqBand == NULL, "DPEqBand NULL. invalid band %d", band);
+    return pEqBand;
+}
+
+static dp_fx::DPMbc* DP_getMbc(DynamicsProcessingContext *pContext, int32_t channel) {
+    dp_fx::DPChannel * pChannel = DP_getChannel(pContext, channel);
+    if (pChannel == NULL) {
+        return NULL;
+    }
+    dp_fx::DPMbc *pMbc = pChannel->getMbc();
+    ALOGE_IF(pMbc == NULL, "DPMbc NULL invalid MBC");
+    return pMbc;
+}
+
+static dp_fx::DPMbcBand* DP_getMbcBand(DynamicsProcessingContext *pContext, int32_t channel,
+        int32_t band) {
+    dp_fx::DPMbc *pMbc = DP_getMbc(pContext, channel);
+    if (pMbc == NULL) {
+        return NULL;
+    }
+    dp_fx::DPMbcBand *pMbcBand = pMbc->getBand(band);
+    ALOGE_IF(pMbcBand == NULL, "pMbcBand NULL. invalid band %d", band);
+    return pMbcBand;
+}
+
+int DP_command(effect_handle_t self, uint32_t cmdCode, uint32_t cmdSize,
+        void *pCmdData, uint32_t *replySize, void *pReplyData) {
+
+    DynamicsProcessingContext * pContext = (DynamicsProcessingContext *)self;
+
+    if (pContext == NULL || pContext->mState == DYNAMICS_PROCESSING_STATE_UNINITIALIZED) {
+        ALOGE("DP_command() called with NULL context or uninitialized state.");
+        return -EINVAL;
+    }
+
+    ALOGV("DP_command command %d cmdSize %d",cmdCode, cmdSize);
+    switch (cmdCode) {
+    case EFFECT_CMD_INIT:
+        if (pReplyData == NULL || *replySize != sizeof(int)) {
+            ALOGE("EFFECT_CMD_INIT wrong replyData or repySize");
+            return -EINVAL;
+        }
+        *(int *) pReplyData = DP_init(pContext);
+        break;
+    case EFFECT_CMD_SET_CONFIG:
+        if (pCmdData == NULL || cmdSize != sizeof(effect_config_t)
+                || pReplyData == NULL || replySize == NULL || *replySize != sizeof(int)) {
+            ALOGE("EFFECT_CMD_SET_CONFIG error with pCmdData, cmdSize, pReplyData or replySize");
+            return -EINVAL;
+        }
+        *(int *) pReplyData = DP_setConfig(pContext,
+                (effect_config_t *) pCmdData);
+        break;
+    case EFFECT_CMD_GET_CONFIG:
+        if (pReplyData == NULL ||
+            *replySize != sizeof(effect_config_t)) {
+            ALOGE("EFFECT_CMD_GET_CONFIG wrong replyData or repySize");
+            return -EINVAL;
+        }
+        DP_getConfig(pContext, (effect_config_t *)pReplyData);
+        break;
+    case EFFECT_CMD_RESET:
+        DP_reset(pContext);
+        break;
+    case EFFECT_CMD_ENABLE:
+        if (pReplyData == NULL || replySize == NULL || *replySize != sizeof(int)) {
+            ALOGE("EFFECT_CMD_ENABLE wrong replyData or repySize");
+            return -EINVAL;
+        }
+        if (pContext->mState != DYNAMICS_PROCESSING_STATE_INITIALIZED) {
+            ALOGE("EFFECT_CMD_ENABLE state not initialized");
+            *(int *)pReplyData = -ENOSYS;
+        } else {
+            pContext->mState = DYNAMICS_PROCESSING_STATE_ACTIVE;
+            ALOGV("EFFECT_CMD_ENABLE() OK");
+            *(int *)pReplyData = 0;
+        }
+        break;
+    case EFFECT_CMD_DISABLE:
+        if (pReplyData == NULL || replySize == NULL || *replySize != sizeof(int)) {
+            ALOGE("EFFECT_CMD_DISABLE wrong replyData or repySize");
+            return -EINVAL;
+        }
+        if (pContext->mState != DYNAMICS_PROCESSING_STATE_ACTIVE) {
+            ALOGE("EFFECT_CMD_DISABLE state not active");
+            *(int *)pReplyData = -ENOSYS;
+        } else {
+            pContext->mState = DYNAMICS_PROCESSING_STATE_INITIALIZED;
+            ALOGV("EFFECT_CMD_DISABLE() OK");
+            *(int *)pReplyData = 0;
+        }
+        break;
+    case EFFECT_CMD_GET_PARAM: {
+        if (pCmdData == NULL || pReplyData == NULL || replySize == NULL) {
+            ALOGE("null pCmdData or pReplyData or replySize");
+            return -EINVAL;
+        }
+        effect_param_t *pEffectParam = (effect_param_t *) pCmdData;
+        uint32_t expectedCmdSize = DP_getParameterCmdSize(pEffectParam->psize,
+                pEffectParam->data);
+        if (cmdSize != expectedCmdSize || *replySize < expectedCmdSize) {
+            ALOGE("error cmdSize: %d, expetedCmdSize: %d, replySize: %d",
+                    cmdSize, expectedCmdSize, *replySize);
+            return -EINVAL;
+        }
+
+        ALOGVV("DP_command expectedCmdSize: %d", expectedCmdSize);
+        memcpy(pReplyData, pCmdData, expectedCmdSize);
+        effect_param_t *p = (effect_param_t *)pReplyData;
+
+        uint32_t voffset = computeParamVOffset(p);
+
+        p->status = DP_getParameter(pContext,
+                p->psize,
+                p->data,
+                &p->vsize,
+                p->data + voffset);
+        *replySize = sizeof(effect_param_t) + voffset + p->vsize;
+
+        ALOGVV("DP_command replysize %u, status %d" , *replySize, p->status);
+        break;
+    }
+    case EFFECT_CMD_SET_PARAM: {
+        if (pCmdData == NULL ||
+                cmdSize < (sizeof(effect_param_t) + sizeof(int32_t) + sizeof(int32_t)) ||
+                pReplyData == NULL || replySize == NULL || *replySize != sizeof(int32_t)) {
+            ALOGE("\tLVM_ERROR : DynamicsProcessing cmdCode Case: "
+                    "EFFECT_CMD_SET_PARAM: ERROR");
+            return -EINVAL;
+        }
+
+        effect_param_t * const p = (effect_param_t *) pCmdData;
+        const uint32_t voffset = computeParamVOffset(p);
+
+        *(int *)pReplyData = DP_setParameter(pContext,
+                p->psize,
+                (void *)p->data,
+                p->vsize,
+                p->data + voffset);
+        break;
+    }
+    case EFFECT_CMD_SET_VOLUME: {
+        ALOGV("EFFECT_CMD_SET_VOLUME");
+        // if pReplyData is NULL, VOL_CTRL is delegated to another effect
+        if (pReplyData == NULL || replySize == NULL || *replySize < ((int)sizeof(int32_t) * 2)) {
+            ALOGV("no VOLUME data to return");
+            break;
+        }
+        if (pCmdData == NULL || cmdSize < ((int)sizeof(uint32_t) * 2)) {
+            ALOGE("\tLVM_ERROR : DynamicsProcessing EFFECT_CMD_SET_VOLUME ERROR");
+            return -EINVAL;
+        }
+
+        const int32_t unityGain = 1 << 24;
+        //channel count
+        int32_t channelCount = (int32_t)audio_channel_count_from_out_mask(
+                pContext->mConfig.inputCfg.channels);
+        for (int32_t ch = 0; ch < channelCount; ch++) {
+
+            dp_fx::DPChannel * pChannel = DP_getChannel(pContext, ch);
+            if (pChannel == NULL) {
+                ALOGE("%s EFFECT_CMD_SET_VOLUME invalid channel %d", __func__, ch);
+                return -EINVAL;
+                break;
+            }
+
+            int32_t offset = ch;
+            if (ch > 1) {
+                // FIXME: limited to 2 unique channels. If more channels present, use value for
+                // first channel
+                offset = 0;
+            }
+            const float gain = (float)*((uint32_t *)pCmdData + offset) / unityGain;
+            const float gainDb = linearToDb(gain);
+            ALOGVV("%s EFFECT_CMD_SET_VOLUME channel %d, engine outputlevel %f (%0.2f dB)",
+                    __func__, ch, gain, gainDb);
+            pChannel->setOutputGain(gainDb);
+        }
+
+        const int32_t  volRet[2] = {unityGain, unityGain}; // Apply no volume before effect.
+        memcpy(pReplyData, volRet, sizeof(volRet));
+        break;
+    }
+    case EFFECT_CMD_SET_DEVICE:
+    case EFFECT_CMD_SET_AUDIO_MODE:
+        break;
+
+    default:
+        ALOGW("DP_command invalid command %d",cmdCode);
+        return -EINVAL;
+    }
+
+    return 0;
+}
+
+//register expected cmd size
+int DP_getParameterCmdSize(uint32_t paramSize,
+        void *pParam) {
+    if (paramSize < sizeof(int32_t)) {
+        return 0;
+    }
+    int32_t param = *(int32_t*)pParam;
+    switch(param) {
+    case DP_PARAM_GET_CHANNEL_COUNT: //paramcmd
+    case DP_PARAM_ENGINE_ARCHITECTURE:
+        //effect + param
+        return (int)(sizeof(effect_param_t) + sizeof(uint32_t));
+    case DP_PARAM_INPUT_GAIN: //paramcmd + param
+    case DP_PARAM_LIMITER:
+    case DP_PARAM_PRE_EQ:
+    case DP_PARAM_POST_EQ:
+    case DP_PARAM_MBC:
+        //effect + param
+        return (int)(sizeof(effect_param_t) + 2 * sizeof(uint32_t));
+    case DP_PARAM_PRE_EQ_BAND:
+    case DP_PARAM_POST_EQ_BAND:
+    case DP_PARAM_MBC_BAND:
+        return (int)(sizeof(effect_param_t) + 3 * sizeof(uint32_t));
+    }
+    return 0;
+}
+
+int DP_getParameter(DynamicsProcessingContext *pContext,
+                           uint32_t paramSize,
+                           void *pParam,
+                           uint32_t *pValueSize,
+                           void *pValue) {
+    int status = 0;
+    int32_t *params = (int32_t *)pParam;
+    static_assert(sizeof(float) == sizeof(int32_t) && sizeof(float) == sizeof(value_t) &&
+            alignof(float) == alignof(int32_t) && alignof(float) == alignof(value_t),
+            "Size/alignment mismatch for float/int32_t/value_t");
+    value_t *values = reinterpret_cast<value_t*>(pValue);
+
+    ALOGVV("%s start", __func__);
+#ifdef VERY_VERY_VERBOSE_LOGGING
+    for (size_t i = 0; i < paramSize/sizeof(int32_t); i++) {
+        ALOGVV("Param[%zu] %d", i, params[i]);
+    }
+#endif
+    if (paramSize < sizeof(int32_t)) {
+        ALOGE("%s invalid paramSize: %u", __func__, paramSize);
+        return -EINVAL;
+    }
+    const int32_t command = params[0];
+    switch (command) {
+    case DP_PARAM_GET_CHANNEL_COUNT: {
+        if (!DP_checkSizesInt(paramSize,*pValueSize, 1 /*params*/, 1 /*values*/)) {
+            ALOGE("%s DP_PARAM_GET_CHANNEL_COUNT (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+        *pValueSize = sizeof(uint32_t);
+        *(uint32_t *)pValue = (uint32_t)audio_channel_count_from_out_mask(
+                pContext->mConfig.inputCfg.channels);
+        ALOGVV("%s DP_PARAM_GET_CHANNEL_COUNT channels %d", __func__, *(int32_t *)pValue);
+        break;
+    }
+    case DP_PARAM_ENGINE_ARCHITECTURE: {
+        ALOGVV("engine architecture paramsize: %d valuesize %d",paramSize, *pValueSize);
+        if (!DP_checkSizesInt(paramSize, *pValueSize, 1 /*params*/, 9 /*values*/)) {
+            ALOGE("%s DP_PARAM_ENGINE_ARCHITECTURE (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+//        Number[] params = { PARAM_ENGINE_ARCHITECTURE };
+//        Number[] values = { 0 /*0 variant */,
+//                0.0f /* 1 preferredFrameDuration */,
+//                0 /*2 preEqInUse */,
+//                0 /*3 preEqBandCount */,
+//                0 /*4 mbcInUse */,
+//                0 /*5 mbcBandCount*/,
+//                0 /*6 postEqInUse */,
+//                0 /*7 postEqBandCount */,
+//                0 /*8 limiterInUse */};
+        if (pContext->mPDynamics == NULL) {
+            ALOGE("%s DP_PARAM_ENGINE_ARCHITECTURE error mPDynamics is NULL", __func__);
+            status = -EINVAL;
+            break;
+        }
+        values[0].i = pContext->mCurrentVariant;
+        values[1].f = pContext->mPreferredFrameDuration;
+        values[2].i = pContext->mPDynamics->isPreEQInUse();
+        values[3].i = pContext->mPDynamics->getPreEqBandCount();
+        values[4].i = pContext->mPDynamics->isMbcInUse();
+        values[5].i = pContext->mPDynamics->getMbcBandCount();
+        values[6].i = pContext->mPDynamics->isPostEqInUse();
+        values[7].i = pContext->mPDynamics->getPostEqBandCount();
+        values[8].i = pContext->mPDynamics->isLimiterInUse();
+
+        *pValueSize = sizeof(value_t) * 9;
+
+        ALOGVV(" variant %d, preferredFrameDuration: %f, preEqInuse %d, bands %d, mbcinuse %d,"
+                "mbcbands %d, posteqInUse %d, bands %d, limiterinuse %d",
+                values[0].i, values[1].f, values[2].i, values[3].i, values[4].i, values[5].i,
+                values[6].i, values[7].i, values[8].i);
+        break;
+    }
+    case DP_PARAM_INPUT_GAIN: {
+        ALOGVV("engine get PARAM_INPUT_GAIN paramsize: %d valuesize %d",paramSize, *pValueSize);
+        if (!DP_checkSizesInt(paramSize, *pValueSize, 2 /*params*/, 1 /*values*/)) {
+            ALOGE("%s get PARAM_INPUT_GAIN invalid sizes.", __func__);
+            status = -EINVAL;
+            break;
+        }
+
+        const int32_t channel = params[1];
+        dp_fx::DPChannel * pChannel = DP_getChannel(pContext, channel);
+        if (pChannel == NULL) {
+            ALOGE("%s get PARAM_INPUT_GAIN invalid channel %d", __func__, channel);
+            status = -EINVAL;
+            break;
+        }
+        values[0].f = pChannel->getInputGain();
+        *pValueSize = sizeof(value_t) * 1;
+
+        ALOGVV(" channel: %d, input gain %f\n", channel, values[0].f);
+        break;
+    }
+    case DP_PARAM_PRE_EQ:
+    case DP_PARAM_POST_EQ: {
+        ALOGVV("engine get PARAM_*_EQ paramsize: %d valuesize %d",paramSize, *pValueSize);
+        if (!DP_checkSizesInt(paramSize, *pValueSize, 2 /*params*/, 3 /*values*/)) {
+            ALOGE("%s get PARAM_*_EQ (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+//        Number[] params = {paramSet == PARAM_PRE_EQ ? PARAM_PRE_EQ : PARAM_POST_EQ,
+//                       channelIndex};
+//               Number[] values = {0 /*0 in use */,
+//                                   0 /*1 enabled*/,
+//                                   0 /*2 band count */};
+        const int32_t channel = params[1];
+
+        dp_fx::DPEq *pEq = DP_getEq(pContext, channel, command);
+        if (pEq == NULL) {
+            ALOGE("%s get PARAM_*_EQ invalid eq", __func__);
+            status = -EINVAL;
+            break;
+        }
+        values[0].i = pEq->isInUse();
+        values[1].i = pEq->isEnabled();
+        values[2].i = pEq->getBandCount();
+        *pValueSize = sizeof(value_t) * 3;
+
+        ALOGVV(" %s channel: %d, inUse::%d, enabled:%d, bandCount:%d\n",
+                (command == DP_PARAM_PRE_EQ ? "preEq" : "postEq"), channel,
+                values[0].i, values[1].i, values[2].i);
+        break;
+    }
+    case DP_PARAM_PRE_EQ_BAND:
+    case DP_PARAM_POST_EQ_BAND: {
+        ALOGVV("engine get PARAM_*_EQ_BAND paramsize: %d valuesize %d",paramSize, *pValueSize);
+        if (!DP_checkSizesInt(paramSize, *pValueSize, 3 /*params*/, 3 /*values*/)) {
+            ALOGE("%s get PARAM_*_EQ_BAND (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+//        Number[] params = {paramSet,
+//                channelIndex,
+//                bandIndex};
+//        Number[] values = {(eqBand.isEnabled() ? 1 : 0),
+//              eqBand.getCutoffFrequency(),
+//              eqBand.getGain()};
+        const int32_t channel = params[1];
+        const int32_t band = params[2];
+        int eqCommand = (command == DP_PARAM_PRE_EQ_BAND ? DP_PARAM_PRE_EQ :
+                (command == DP_PARAM_POST_EQ_BAND ? DP_PARAM_POST_EQ : -1));
+
+        dp_fx::DPEqBand *pEqBand = DP_getEqBand(pContext, channel, eqCommand, band);
+        if (pEqBand == NULL) {
+            ALOGE("%s get PARAM_*_EQ_BAND invalid channel %d or band %d", __func__, channel, band);
+            status = -EINVAL;
+            break;
+        }
+
+        values[0].i = pEqBand->isEnabled();
+        values[1].f = pEqBand->getCutoffFrequency();
+        values[2].f = pEqBand->getGain();
+        *pValueSize = sizeof(value_t) * 3;
+
+        ALOGVV("%s channel: %d, band::%d, enabled:%d, cutoffFrequency:%f, gain%f\n",
+                (command == DP_PARAM_PRE_EQ_BAND ? "preEqBand" : "postEqBand"), channel, band,
+                values[0].i, values[1].f, values[2].f);
+        break;
+    }
+    case DP_PARAM_MBC: {
+        ALOGVV("engine get PDP_PARAM_MBC paramsize: %d valuesize %d",paramSize, *pValueSize);
+        if (!DP_checkSizesInt(paramSize, *pValueSize, 2 /*params*/, 3 /*values*/)) {
+            ALOGE("%s get PDP_PARAM_MBC (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+
+//           Number[] params = {PARAM_MBC,
+//                    channelIndex};
+//            Number[] values = {0 /*0 in use */,
+//                                0 /*1 enabled*/,
+//                                0 /*2 band count */};
+
+        const int32_t channel = params[1];
+
+        dp_fx::DPMbc *pMbc = DP_getMbc(pContext, channel);
+        if (pMbc == NULL) {
+            ALOGE("%s get PDP_PARAM_MBC invalid MBC", __func__);
+            status = -EINVAL;
+            break;
+        }
+
+        values[0].i = pMbc->isInUse();
+        values[1].i = pMbc->isEnabled();
+        values[2].i = pMbc->getBandCount();
+        *pValueSize = sizeof(value_t) * 3;
+
+        ALOGVV("DP_PARAM_MBC channel: %d, inUse::%d, enabled:%d, bandCount:%d\n", channel,
+                values[0].i, values[1].i, values[2].i);
+        break;
+    }
+    case DP_PARAM_MBC_BAND: {
+        ALOGVV("engine get DP_PARAM_MBC_BAND paramsize: %d valuesize %d",paramSize, *pValueSize);
+        if (!DP_checkSizesInt(paramSize, *pValueSize, 3 /*params*/, 11 /*values*/)) {
+            ALOGE("%s get DP_PARAM_MBC_BAND (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+//        Number[] params = {PARAM_MBC_BAND,
+//                        channelIndex,
+//                        bandIndex};
+//                Number[] values = {0 /*0 enabled */,
+//                        0.0f /*1 cutoffFrequency */,
+//                        0.0f /*2 AttackTime */,
+//                        0.0f /*3 ReleaseTime */,
+//                        0.0f /*4 Ratio */,
+//                        0.0f /*5 Threshold */,
+//                        0.0f /*6 KneeWidth */,
+//                        0.0f /*7 NoiseGateThreshold */,
+//                        0.0f /*8 ExpanderRatio */,
+//                        0.0f /*9 PreGain */,
+//                        0.0f /*10 PostGain*/};
+
+        const int32_t channel = params[1];
+        const int32_t band = params[2];
+
+        dp_fx::DPMbcBand *pMbcBand = DP_getMbcBand(pContext, channel, band);
+        if (pMbcBand == NULL) {
+            ALOGE("%s get PARAM_MBC_BAND invalid channel %d or band %d", __func__, channel, band);
+            status = -EINVAL;
+            break;
+        }
+
+        values[0].i = pMbcBand->isEnabled();
+        values[1].f = pMbcBand->getCutoffFrequency();
+        values[2].f = pMbcBand->getAttackTime();
+        values[3].f = pMbcBand->getReleaseTime();
+        values[4].f = pMbcBand->getRatio();
+        values[5].f = pMbcBand->getThreshold();
+        values[6].f = pMbcBand->getKneeWidth();
+        values[7].f = pMbcBand->getNoiseGateThreshold();
+        values[8].f = pMbcBand->getExpanderRatio();
+        values[9].f = pMbcBand->getPreGain();
+        values[10].f = pMbcBand->getPostGain();
+
+        *pValueSize = sizeof(value_t) * 11;
+        ALOGVV(" mbcBand channel: %d, band::%d, enabled:%d, cutoffFrequency:%f, attackTime:%f,"
+                "releaseTime:%f, ratio:%f, threshold:%f, kneeWidth:%f, noiseGateThreshold:%f,"
+                "expanderRatio:%f, preGain:%f, postGain:%f\n", channel, band, values[0].i,
+                values[1].f, values[2].f, values[3].f, values[4].f, values[5].f, values[6].f,
+                values[7].f, values[8].f, values[9].f, values[10].f);
+        break;
+    }
+    case DP_PARAM_LIMITER: {
+        ALOGVV("engine get DP_PARAM_LIMITER paramsize: %d valuesize %d",paramSize, *pValueSize);
+        if (!DP_checkSizesInt(paramSize, *pValueSize, 2 /*params*/, 8 /*values*/)) {
+            ALOGE("%s DP_PARAM_LIMITER (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+
+        int32_t channel = params[1];
+//      Number[] values = {0 /*0 in use (int)*/,
+//              0 /*1 enabled (int)*/,
+//              0 /*2 link group (int)*/,
+//              0.0f /*3 attack time (float)*/,
+//              0.0f /*4 release time (float)*/,
+//              0.0f /*5 ratio (float)*/,
+//              0.0f /*6 threshold (float)*/,
+//              0.0f /*7 post gain(float)*/};
+        dp_fx::DPChannel * pChannel = DP_getChannel(pContext, channel);
+        if (pChannel == NULL) {
+            ALOGE("%s DP_PARAM_LIMITER invalid channel %d", __func__, channel);
+            status = -EINVAL;
+            break;
+        }
+        dp_fx::DPLimiter *pLimiter = pChannel->getLimiter();
+        if (pLimiter == NULL) {
+            ALOGE("%s DP_PARAM_LIMITER null LIMITER", __func__);
+            status = -EINVAL;
+            break;
+        }
+        values[0].i = pLimiter->isInUse();
+        values[1].i = pLimiter->isEnabled();
+        values[2].i = pLimiter->getLinkGroup();
+        values[3].f = pLimiter->getAttackTime();
+        values[4].f = pLimiter->getReleaseTime();
+        values[5].f = pLimiter->getRatio();
+        values[6].f = pLimiter->getThreshold();
+        values[7].f = pLimiter->getPostGain();
+
+        *pValueSize = sizeof(value_t) * 8;
+
+        ALOGVV(" Limiter channel: %d, inUse::%d, enabled:%d, linkgroup:%d attackTime:%f,"
+                "releaseTime:%f, ratio:%f, threshold:%f, postGain:%f\n",
+                channel, values[0].i/*inUse*/, values[1].i/*enabled*/, values[2].i/*linkGroup*/,
+                values[3].f/*attackTime*/, values[4].f/*releaseTime*/,
+                values[5].f/*ratio*/, values[6].f/*threshold*/,
+                values[7].f/*postGain*/);
+        break;
+    }
+    default:
+        ALOGE("%s invalid param %d", __func__, params[0]);
+        status = -EINVAL;
+        break;
+    }
+
+    ALOGVV("%s end param: %d, status: %d", __func__, params[0], status);
+    return status;
+} /* end DP_getParameter */
+
+int DP_setParameter(DynamicsProcessingContext *pContext,
+                           uint32_t paramSize,
+                           void *pParam,
+                           uint32_t valueSize,
+                           void *pValue) {
+    int status = 0;
+    int32_t *params = (int32_t *)pParam;
+    static_assert(sizeof(float) == sizeof(int32_t) && sizeof(float) == sizeof(value_t) &&
+            alignof(float) == alignof(int32_t) && alignof(float) == alignof(value_t),
+            "Size/alignment mismatch for float/int32_t/value_t");
+    value_t *values = reinterpret_cast<value_t*>(pValue);
+
+    ALOGVV("%s start", __func__);
+    if (paramSize < sizeof(int32_t)) {
+        ALOGE("%s invalid paramSize: %u", __func__, paramSize);
+        return -EINVAL;
+    }
+    const int32_t command = params[0];
+    switch (command) {
+    case DP_PARAM_ENGINE_ARCHITECTURE: {
+        ALOGVV("engine architecture paramsize: %d valuesize %d",paramSize, valueSize);
+        if (!DP_checkSizesInt(paramSize, valueSize, 1 /*params*/, 9 /*values*/)) {
+            ALOGE("%s DP_PARAM_ENGINE_ARCHITECTURE (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+//        Number[] params = { PARAM_ENGINE_ARCHITECTURE };
+//        Number[] values = { variant /* variant */,
+//                preferredFrameDuration,
+//                (preEqInUse ? 1 : 0),
+//                preEqBandCount,
+//                (mbcInUse ? 1 : 0),
+//                mbcBandCount,
+//                (postEqInUse ? 1 : 0),
+//                postEqBandCount,
+//                (limiterInUse ? 1 : 0)};
+        const int32_t variant = values[0].i;
+        const float preferredFrameDuration = values[1].f;
+        const int32_t preEqInUse = values[2].i;
+        const int32_t preEqBandCount = values[3].i;
+        const int32_t mbcInUse = values[4].i;
+        const int32_t mbcBandCount = values[5].i;
+        const int32_t postEqInUse = values[6].i;
+        const int32_t postEqBandCount = values[7].i;
+        const int32_t limiterInUse = values[8].i;
+        ALOGVV("variant %d, preEqInuse %d, bands %d, mbcinuse %d, mbcbands %d, posteqInUse %d,"
+                "bands %d, limiterinuse %d", variant, preEqInUse, preEqBandCount, mbcInUse,
+                mbcBandCount, postEqInUse, postEqBandCount, limiterInUse);
+
+        //set variant (instantiate effect)
+        //initArchitecture for effect
+        DP_changeVariant(pContext, variant);
+        if (pContext->mPDynamics == NULL) {
+            ALOGE("%s DP_PARAM_ENGINE_ARCHITECTURE error setting variant %d", __func__, variant);
+            status = -EINVAL;
+            break;
+        }
+        pContext->mPreferredFrameDuration = preferredFrameDuration;
+        pContext->mPDynamics->init((uint32_t)audio_channel_count_from_out_mask(
+                pContext->mConfig.inputCfg.channels),
+                preEqInUse != 0, (uint32_t)preEqBandCount,
+                mbcInUse != 0, (uint32_t)mbcBandCount,
+                postEqInUse != 0, (uint32_t)postEqBandCount,
+                limiterInUse != 0);
+
+        DP_configureVariant(pContext, variant);
+        break;
+    }
+    case DP_PARAM_INPUT_GAIN: {
+        ALOGVV("engine DP_PARAM_INPUT_GAIN paramsize: %d valuesize %d",paramSize, valueSize);
+        if (!DP_checkSizesInt(paramSize, valueSize, 2 /*params*/, 1 /*values*/)) {
+            ALOGE("%s DP_PARAM_INPUT_GAIN invalid sizes.", __func__);
+            status = -EINVAL;
+            break;
+        }
+
+        const int32_t channel = params[1];
+        dp_fx::DPChannel * pChannel = DP_getChannel(pContext, channel);
+        if (pChannel == NULL) {
+            ALOGE("%s DP_PARAM_INPUT_GAIN invalid channel %d", __func__, channel);
+            status = -EINVAL;
+            break;
+        }
+        const float gain = values[0].f;
+        ALOGVV("%s DP_PARAM_INPUT_GAIN channel %d, level %f", __func__, channel, gain);
+        pChannel->setInputGain(gain);
+        break;
+    }
+    case DP_PARAM_PRE_EQ:
+    case DP_PARAM_POST_EQ: {
+        ALOGVV("engine DP_PARAM_*_EQ paramsize: %d valuesize %d",paramSize, valueSize);
+        if (!DP_checkSizesInt(paramSize, valueSize, 2 /*params*/, 3 /*values*/)) {
+            ALOGE("%s DP_PARAM_*_EQ (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+//        Number[] params = {paramSet,
+//                channelIndex};
+//        Number[] values = { (eq.isInUse() ? 1 : 0),
+//                (eq.isEnabled() ? 1 : 0),
+//                bandCount};
+        const int32_t channel = params[1];
+
+        const int32_t enabled = values[1].i;
+        const int32_t bandCount = values[2].i;
+        ALOGVV(" %s channel: %d, inUse::%d, enabled:%d, bandCount:%d\n",
+                (command == DP_PARAM_PRE_EQ ? "preEq" : "postEq"), channel, values[0].i,
+                values[2].i, bandCount);
+
+        dp_fx::DPEq *pEq = DP_getEq(pContext, channel, command);
+        if (pEq == NULL) {
+            ALOGE("%s set PARAM_*_EQ invalid channel %d or command %d", __func__, channel,
+                    command);
+            status = -EINVAL;
+            break;
+        }
+
+        pEq->setEnabled(enabled != 0);
+        //fail if bandcountis different? maybe.
+        if ((int32_t)pEq->getBandCount() != bandCount) {
+            ALOGW("%s warning, trying to set different bandcount from %d to %d", __func__,
+                    pEq->getBandCount(), bandCount);
+        }
+        break;
+    }
+    case DP_PARAM_PRE_EQ_BAND:
+    case DP_PARAM_POST_EQ_BAND: {
+        ALOGVV("engine set PARAM_*_EQ_BAND paramsize: %d valuesize %d",paramSize, valueSize);
+        if (!DP_checkSizesInt(paramSize, valueSize, 3 /*params*/, 3 /*values*/)) {
+            ALOGE("%s PARAM_*_EQ_BAND (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+//        Number[] values = { channelIndex,
+//                bandIndex,
+//                (eqBand.isEnabled() ? 1 : 0),
+//                eqBand.getCutoffFrequency(),
+//                eqBand.getGain()};
+
+//        Number[] params = {paramSet,
+//                channelIndex,
+//                bandIndex};
+//        Number[] values = {(eqBand.isEnabled() ? 1 : 0),
+//              eqBand.getCutoffFrequency(),
+//              eqBand.getGain()};
+
+        const int32_t channel = params[1];
+        const int32_t band = params[2];
+
+        const int32_t enabled = values[0].i;
+        const float cutoffFrequency = values[1].f;
+        const float gain = values[2].f;
+
+
+        ALOGVV(" %s channel: %d, band::%d, enabled:%d, cutoffFrequency:%f, gain%f\n",
+                (command == DP_PARAM_PRE_EQ_BAND ? "preEqBand" : "postEqBand"), channel, band,
+                enabled, cutoffFrequency, gain);
+
+        int eqCommand = (command == DP_PARAM_PRE_EQ_BAND ? DP_PARAM_PRE_EQ :
+                (command == DP_PARAM_POST_EQ_BAND ? DP_PARAM_POST_EQ : -1));
+        dp_fx::DPEq *pEq = DP_getEq(pContext, channel, eqCommand);
+        if (pEq == NULL) {
+            ALOGE("%s set PARAM_*_EQ_BAND invalid channel %d or command %d", __func__, channel,
+                    command);
+            status = -EINVAL;
+            break;
+        }
+
+        dp_fx::DPEqBand eqBand;
+        eqBand.init(enabled != 0, cutoffFrequency, gain);
+        pEq->setBand(band, eqBand);
+        break;
+    }
+    case DP_PARAM_MBC: {
+        ALOGVV("engine DP_PARAM_MBC paramsize: %d valuesize %d",paramSize, valueSize);
+        if (!DP_checkSizesInt(paramSize, valueSize, 2 /*params*/, 3 /*values*/)) {
+            ALOGE("%s DP_PARAM_MBC (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+//            Number[] params = { PARAM_MBC,
+//                    channelIndex};
+//            Number[] values = {(mbc.isInUse() ? 1 : 0),
+//                    (mbc.isEnabled() ? 1 : 0),
+//                    bandCount};
+        const int32_t channel = params[1];
+
+        const int32_t enabled = values[1].i;
+        const int32_t bandCount = values[2].i;
+        ALOGVV("MBC channel: %d, inUse::%d, enabled:%d, bandCount:%d\n", channel, values[0].i,
+                enabled, bandCount);
+
+        dp_fx::DPMbc *pMbc = DP_getMbc(pContext, channel);
+        if (pMbc == NULL) {
+            ALOGE("%s set DP_PARAM_MBC invalid channel %d ", __func__, channel);
+            status = -EINVAL;
+            break;
+        }
+
+        pMbc->setEnabled(enabled != 0);
+        //fail if bandcountis different? maybe.
+        if ((int32_t)pMbc->getBandCount() != bandCount) {
+            ALOGW("%s warning, trying to set different bandcount from %d to %d", __func__,
+                    pMbc->getBandCount(), bandCount);
+        }
+        break;
+    }
+    case DP_PARAM_MBC_BAND: {
+        ALOGVV("engine set DP_PARAM_MBC_BAND paramsize: %d valuesize %d ",paramSize, valueSize);
+        if (!DP_checkSizesInt(paramSize, valueSize, 3 /*params*/, 11 /*values*/)) {
+            ALOGE("%s DP_PARAM_MBC_BAND: (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+//        Number[] params = { PARAM_MBC_BAND,
+//                channelIndex,
+//                bandIndex};
+//        Number[] values = {(mbcBand.isEnabled() ? 1 : 0),
+//                mbcBand.getCutoffFrequency(),
+//                mbcBand.getAttackTime(),
+//                mbcBand.getReleaseTime(),
+//                mbcBand.getRatio(),
+//                mbcBand.getThreshold(),
+//                mbcBand.getKneeWidth(),
+//                mbcBand.getNoiseGateThreshold(),
+//                mbcBand.getExpanderRatio(),
+//                mbcBand.getPreGain(),
+//                mbcBand.getPostGain()};
+
+        const int32_t channel = params[1];
+        const int32_t band = params[2];
+
+        const int32_t enabled = values[0].i;
+        const float cutoffFrequency = values[1].f;
+        const float attackTime = values[2].f;
+        const float releaseTime = values[3].f;
+        const float ratio = values[4].f;
+        const float threshold = values[5].f;
+        const float kneeWidth = values[6].f;
+        const float noiseGateThreshold = values[7].f;
+        const float expanderRatio = values[8].f;
+        const float preGain = values[9].f;
+        const float postGain = values[10].f;
+
+        ALOGVV(" mbcBand channel: %d, band::%d, enabled:%d, cutoffFrequency:%f, attackTime:%f,"
+                "releaseTime:%f, ratio:%f, threshold:%f, kneeWidth:%f, noiseGateThreshold:%f,"
+                "expanderRatio:%f, preGain:%f, postGain:%f\n",
+                channel, band, enabled, cutoffFrequency, attackTime, releaseTime, ratio,
+                threshold, kneeWidth, noiseGateThreshold, expanderRatio, preGain, postGain);
+
+        dp_fx::DPMbc *pMbc = DP_getMbc(pContext, channel);
+        if (pMbc == NULL) {
+            ALOGE("%s set DP_PARAM_MBC_BAND invalid channel %d", __func__, channel);
+            status = -EINVAL;
+            break;
+        }
+
+        dp_fx::DPMbcBand mbcBand;
+        mbcBand.init(enabled != 0, cutoffFrequency, attackTime, releaseTime, ratio, threshold,
+                kneeWidth, noiseGateThreshold, expanderRatio, preGain, postGain);
+        pMbc->setBand(band, mbcBand);
+        break;
+    }
+    case DP_PARAM_LIMITER: {
+        ALOGVV("engine DP_PARAM_LIMITER paramsize: %d valuesize %d",paramSize, valueSize);
+        if (!DP_checkSizesInt(paramSize, valueSize, 2 /*params*/, 8 /*values*/)) {
+            ALOGE("%s DP_PARAM_LIMITER (cmd %d) invalid sizes.", __func__, command);
+            status = -EINVAL;
+            break;
+        }
+//            Number[] params = { PARAM_LIMITER,
+//                             channelIndex};
+//                     Number[] values = {(limiter.isInUse() ? 1 : 0),
+//                             (limiter.isEnabled() ? 1 : 0),
+//                             limiter.getLinkGroup(),
+//                             limiter.getAttackTime(),
+//                             limiter.getReleaseTime(),
+//                             limiter.getRatio(),
+//                             limiter.getThreshold(),
+//                             limiter.getPostGain()};
+
+        const int32_t channel = params[1];
+
+        const int32_t inUse = values[0].i;
+        const int32_t enabled = values[1].i;
+        const int32_t linkGroup = values[2].i;
+        const float attackTime = values[3].f;
+        const float releaseTime = values[4].f;
+        const float ratio = values[5].f;
+        const float threshold = values[6].f;
+        const float postGain = values[7].f;
+
+        ALOGVV(" Limiter channel: %d, inUse::%d, enabled:%d, linkgroup:%d attackTime:%f,"
+                "releaseTime:%f, ratio:%f, threshold:%f, postGain:%f\n", channel, inUse,
+                enabled, linkGroup, attackTime, releaseTime, ratio, threshold, postGain);
+
+        dp_fx::DPChannel * pChannel = DP_getChannel(pContext, channel);
+        if (pChannel == NULL) {
+            ALOGE("%s DP_PARAM_LIMITER invalid channel %d", __func__, channel);
+            status = -EINVAL;
+            break;
+        }
+        dp_fx::DPLimiter limiter;
+        limiter.init(inUse != 0, enabled != 0, linkGroup, attackTime, releaseTime, ratio,
+                threshold, postGain);
+        pChannel->setLimiter(limiter);
+        break;
+    }
+    default:
+        ALOGE("%s invalid param %d", __func__, params[0]);
+        status = -EINVAL;
+        break;
+    }
+
+    ALOGVV("%s end param: %d, status: %d", __func__, params[0], status);
+    return status;
+} /* end DP_setParameter */
+
+/* Effect Control Interface Implementation: get_descriptor */
+int DP_getDescriptor(effect_handle_t self,
+        effect_descriptor_t *pDescriptor)
+{
+    DynamicsProcessingContext * pContext = (DynamicsProcessingContext *) self;
+
+    if (pContext == NULL || pDescriptor == NULL) {
+        ALOGE("DP_getDescriptor() invalid param");
+        return -EINVAL;
+    }
+
+    *pDescriptor = gDPDescriptor;
+
+    return 0;
+} /* end DP_getDescriptor */
+
+
+// effect_handle_t interface implementation for Dynamics Processing effect
+const struct effect_interface_s gDPInterface = {
+        DP_process,
+        DP_command,
+        DP_getDescriptor,
+        NULL,
+};
+
+extern "C" {
+// This is the only symbol that needs to be exported
+__attribute__ ((visibility ("default")))
+audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
+    .tag = AUDIO_EFFECT_LIBRARY_TAG,
+    .version = EFFECT_LIBRARY_API_VERSION,
+    .name = "Dynamics Processing Library",
+    .implementor = "The Android Open Source Project",
+    .create_effect = DPLib_Create,
+    .release_effect = DPLib_Release,
+    .get_descriptor = DPLib_GetDescriptor,
+};
+
+}; // extern "C"
+

diff --git a/media/libstagefright/matroska/MODULE_LICENSE_APACHE2 b/media/libeffects/dynamicsproc/MODULE_LICENSE_APACHE2
similarity index 100%
rename from media/libstagefright/matroska/MODULE_LICENSE_APACHE2
rename to media/libeffects/dynamicsproc/MODULE_LICENSE_APACHE2


diff --git a/media/libeffects/dynamicsproc/NOTICE b/media/libeffects/dynamicsproc/NOTICE
new file mode 100644
index 0000000..31cc6e9
--- /dev/null
+++ b/media/libeffects/dynamicsproc/NOTICE

@@ -0,0 +1,190 @@
+
+   Copyright (c) 2005-2018, The Android Open Source Project
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+
+
+                                 Apache License
+                           Version 2.0, January 2004
+                        http://www.apache.org/licenses/
+
+   TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
+
+   1. Definitions.
+
+      "License" shall mean the terms and conditions for use, reproduction,
+      and distribution as defined by Sections 1 through 9 of this document.
+
+      "Licensor" shall mean the copyright owner or entity authorized by
+      the copyright owner that is granting the License.
+
+      "Legal Entity" shall mean the union of the acting entity and all
+      other entities that control, are controlled by, or are under common
+      control with that entity. For the purposes of this definition,
+      "control" means (i) the power, direct or indirect, to cause the
+      direction or management of such entity, whether by contract or
+      otherwise, or (ii) ownership of fifty percent (50%) or more of the
+      outstanding shares, or (iii) beneficial ownership of such entity.
+
+      "You" (or "Your") shall mean an individual or Legal Entity
+      exercising permissions granted by this License.
+
+      "Source" form shall mean the preferred form for making modifications,
+      including but not limited to software source code, documentation
+      source, and configuration files.
+
+      "Object" form shall mean any form resulting from mechanical
+      transformation or translation of a Source form, including but
+      not limited to compiled object code, generated documentation,
+      and conversions to other media types.
+
+      "Work" shall mean the work of authorship, whether in Source or
+      Object form, made available under the License, as indicated by a
+      copyright notice that is included in or attached to the work
+      (an example is provided in the Appendix below).
+
+      "Derivative Works" shall mean any work, whether in Source or Object
+      form, that is based on (or derived from) the Work and for which the
+      editorial revisions, annotations, elaborations, or other modifications
+      represent, as a whole, an original work of authorship. For the purposes
+      of this License, Derivative Works shall not include works that remain
+      separable from, or merely link (or bind by name) to the interfaces of,
+      the Work and Derivative Works thereof.
+
+      "Contribution" shall mean any work of authorship, including
+      the original version of the Work and any modifications or additions
+      to that Work or Derivative Works thereof, that is intentionally
+      submitted to Licensor for inclusion in the Work by the copyright owner
+      or by an individual or Legal Entity authorized to submit on behalf of
+      the copyright owner. For the purposes of this definition, "submitted"
+      means any form of electronic, verbal, or written communication sent
+      to the Licensor or its representatives, including but not limited to
+      communication on electronic mailing lists, source code control systems,
+      and issue tracking systems that are managed by, or on behalf of, the
+      Licensor for the purpose of discussing and improving the Work, but
+      excluding communication that is conspicuously marked or otherwise
+      designated in writing by the copyright owner as "Not a Contribution."
+
+      "Contributor" shall mean Licensor and any individual or Legal Entity
+      on behalf of whom a Contribution has been received by Licensor and
+      subsequently incorporated within the Work.
+
+   2. Grant of Copyright License. Subject to the terms and conditions of
+      this License, each Contributor hereby grants to You a perpetual,
+      worldwide, non-exclusive, no-charge, royalty-free, irrevocable
+      copyright license to reproduce, prepare Derivative Works of,
+      publicly display, publicly perform, sublicense, and distribute the
+      Work and such Derivative Works in Source or Object form.
+
+   3. Grant of Patent License. Subject to the terms and conditions of
+      this License, each Contributor hereby grants to You a perpetual,
+      worldwide, non-exclusive, no-charge, royalty-free, irrevocable
+      (except as stated in this section) patent license to make, have made,
+      use, offer to sell, sell, import, and otherwise transfer the Work,
+      where such license applies only to those patent claims licensable
+      by such Contributor that are necessarily infringed by their
+      Contribution(s) alone or by combination of their Contribution(s)
+      with the Work to which such Contribution(s) was submitted. If You
+      institute patent litigation against any entity (including a
+      cross-claim or counterclaim in a lawsuit) alleging that the Work
+      or a Contribution incorporated within the Work constitutes direct
+      or contributory patent infringement, then any patent licenses
+      granted to You under this License for that Work shall terminate
+      as of the date such litigation is filed.
+
+   4. Redistribution. You may reproduce and distribute copies of the
+      Work or Derivative Works thereof in any medium, with or without
+      modifications, and in Source or Object form, provided that You
+      meet the following conditions:
+
+      (a) You must give any other recipients of the Work or
+          Derivative Works a copy of this License; and
+
+      (b) You must cause any modified files to carry prominent notices
+          stating that You changed the files; and
+
+      (c) You must retain, in the Source form of any Derivative Works
+          that You distribute, all copyright, patent, trademark, and
+          attribution notices from the Source form of the Work,
+          excluding those notices that do not pertain to any part of
+          the Derivative Works; and
+
+      (d) If the Work includes a "NOTICE" text file as part of its
+          distribution, then any Derivative Works that You distribute must
+          include a readable copy of the attribution notices contained
+          within such NOTICE file, excluding those notices that do not
+          pertain to any part of the Derivative Works, in at least one
+          of the following places: within a NOTICE text file distributed
+          as part of the Derivative Works; within the Source form or
+          documentation, if provided along with the Derivative Works; or,
+          within a display generated by the Derivative Works, if and
+          wherever such third-party notices normally appear. The contents
+          of the NOTICE file are for informational purposes only and
+          do not modify the License. You may add Your own attribution
+          notices within Derivative Works that You distribute, alongside
+          or as an addendum to the NOTICE text from the Work, provided
+          that such additional attribution notices cannot be construed
+          as modifying the License.
+
+      You may add Your own copyright statement to Your modifications and
+      may provide additional or different license terms and conditions
+      for use, reproduction, or distribution of Your modifications, or
+      for any such Derivative Works as a whole, provided Your use,
+      reproduction, and distribution of the Work otherwise complies with
+      the conditions stated in this License.
+
+   5. Submission of Contributions. Unless You explicitly state otherwise,
+      any Contribution intentionally submitted for inclusion in the Work
+      by You to the Licensor shall be under the terms and conditions of
+      this License, without any additional terms or conditions.
+      Notwithstanding the above, nothing herein shall supersede or modify
+      the terms of any separate license agreement you may have executed
+      with Licensor regarding such Contributions.
+
+   6. Trademarks. This License does not grant permission to use the trade
+      names, trademarks, service marks, or product names of the Licensor,
+      except as required for reasonable and customary use in describing the
+      origin of the Work and reproducing the content of the NOTICE file.
+
+   7. Disclaimer of Warranty. Unless required by applicable law or
+      agreed to in writing, Licensor provides the Work (and each
+      Contributor provides its Contributions) on an "AS IS" BASIS,
+      WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or
+      implied, including, without limitation, any warranties or conditions
+      of TITLE, NON-INFRINGEMENT, MERCHANTABILITY, or FITNESS FOR A
+      PARTICULAR PURPOSE. You are solely responsible for determining the
+      appropriateness of using or redistributing the Work and assume any
+      risks associated with Your exercise of permissions under this License.
+
+   8. Limitation of Liability. In no event and under no legal theory,
+      whether in tort (including negligence), contract, or otherwise,
+      unless required by applicable law (such as deliberate and grossly
+      negligent acts) or agreed to in writing, shall any Contributor be
+      liable to You for damages, including any direct, indirect, special,
+      incidental, or consequential damages of any character arising as a
+      result of this License or out of the use or inability to use the
+      Work (including but not limited to damages for loss of goodwill,
+      work stoppage, computer failure or malfunction, or any and all
+      other commercial damages or losses), even if such Contributor
+      has been advised of the possibility of such damages.
+
+   9. Accepting Warranty or Additional Liability. While redistributing
+      the Work or Derivative Works thereof, You may choose to offer,
+      and charge a fee for, acceptance of support, warranty, indemnity,
+      or other liability obligations and/or rights consistent with this
+      License. However, in accepting such obligations, You may act only
+      on Your own behalf and on Your sole responsibility, not on behalf
+      of any other Contributor, and only if You agree to indemnify,
+      defend, and hold each Contributor harmless for any liability
+      incurred by, or claims asserted against, such Contributor by reason
+      of your accepting any such warranty or additional liability.
+
+   END OF TERMS AND CONDITIONS
+

diff --git a/media/libeffects/dynamicsproc/dsp/DPBase.cpp b/media/libeffects/dynamicsproc/dsp/DPBase.cpp
new file mode 100644
index 0000000..ac758e0
--- /dev/null
+++ b/media/libeffects/dynamicsproc/dsp/DPBase.cpp

@@ -0,0 +1,265 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "DPBase"
+//#define LOG_NDEBUG 0
+
+#include <log/log.h>
+#include "DPBase.h"
+#include "DPFrequency.h"
+
+namespace dp_fx {
+
+DPStage::DPStage() : mInUse(DP_DEFAULT_STAGE_INUSE),
+        mEnabled(DP_DEFAULT_STAGE_ENABLED) {
+}
+
+void DPStage::init(bool inUse, bool enabled) {
+    mInUse = inUse;
+    mEnabled = enabled;
+}
+
+//----
+DPBandStage::DPBandStage() : mBandCount(0) {
+}
+
+void DPBandStage::init(bool inUse, bool enabled, int bandCount) {
+    DPStage::init(inUse, enabled);
+    mBandCount = inUse ? bandCount : 0;
+}
+
+//---
+DPBandBase::DPBandBase() {
+    init(DP_DEFAULT_BAND_ENABLED,
+            DP_DEFAULT_BAND_CUTOFF_FREQUENCY_HZ);
+}
+
+void DPBandBase::init(bool enabled, float cutoffFrequency){
+    mEnabled = enabled;
+    mCutoofFrequencyHz = cutoffFrequency;
+}
+
+//-----
+DPEqBand::DPEqBand() {
+    init(DP_DEFAULT_BAND_ENABLED,
+            DP_DEFAULT_BAND_CUTOFF_FREQUENCY_HZ,
+            DP_DEFAULT_GAIN_DB);
+}
+
+void DPEqBand::init(bool enabled, float cutoffFrequency, float gain) {
+    DPBandBase::init(enabled, cutoffFrequency);
+    setGain(gain);
+}
+
+float DPEqBand::getGain() const{
+    return mGainDb;
+}
+
+void DPEqBand::setGain(float gain) {
+    mGainDb = gain;
+}
+
+//------
+DPMbcBand::DPMbcBand() {
+    init(DP_DEFAULT_BAND_ENABLED,
+            DP_DEFAULT_BAND_CUTOFF_FREQUENCY_HZ,
+            DP_DEFAULT_ATTACK_TIME_MS,
+            DP_DEFAULT_RELEASE_TIME_MS,
+            DP_DEFAULT_RATIO,
+            DP_DEFAULT_THRESHOLD_DB,
+            DP_DEFAULT_KNEE_WIDTH_DB,
+            DP_DEFAULT_NOISE_GATE_THRESHOLD_DB,
+            DP_DEFAULT_EXPANDER_RATIO,
+            DP_DEFAULT_GAIN_DB,
+            DP_DEFAULT_GAIN_DB);
+}
+
+void DPMbcBand::init(bool enabled, float cutoffFrequency, float attackTime, float releaseTime,
+        float ratio, float threshold, float kneeWidth, float noiseGateThreshold,
+        float expanderRatio, float preGain, float postGain) {
+    DPBandBase::init(enabled, cutoffFrequency);
+    setAttackTime(attackTime);
+    setReleaseTime(releaseTime);
+    setRatio(ratio);
+    setThreshold(threshold);
+    setKneeWidth(kneeWidth);
+    setNoiseGateThreshold(noiseGateThreshold);
+    setExpanderRatio(expanderRatio);
+    setPreGain(preGain);
+    setPostGain(postGain);
+}
+
+//------
+DPEq::DPEq() {
+}
+
+void DPEq::init(bool inUse, bool enabled, uint32_t bandCount) {
+    DPBandStage::init(inUse, enabled, bandCount);
+    mBands.resize(getBandCount());
+}
+
+DPEqBand * DPEq::getBand(uint32_t band) {
+    if (band < getBandCount()) {
+        return &mBands[band];
+    }
+    return NULL;
+}
+
+void DPEq::setBand(uint32_t band, DPEqBand &src) {
+    if (band < getBandCount()) {
+        mBands[band] = src;
+    }
+}
+
+//------
+DPMbc::DPMbc() {
+}
+
+void DPMbc::init(bool inUse, bool enabled, uint32_t bandCount) {
+    DPBandStage::init(inUse, enabled, bandCount);
+    if (isInUse()) {
+        mBands.resize(bandCount);
+    } else {
+        mBands.resize(0);
+    }
+}
+
+DPMbcBand * DPMbc::getBand(uint32_t band) {
+    if (band < getBandCount()) {
+        return &mBands[band];
+    }
+    return NULL;
+}
+
+void DPMbc::setBand(uint32_t band, DPMbcBand &src) {
+    if (band < getBandCount()) {
+        mBands[band] = src;
+    }
+}
+
+//------
+DPLimiter::DPLimiter() {
+    init(DP_DEFAULT_STAGE_INUSE,
+            DP_DEFAULT_STAGE_ENABLED,
+            DP_DEFAULT_LINK_GROUP,
+            DP_DEFAULT_ATTACK_TIME_MS,
+            DP_DEFAULT_RELEASE_TIME_MS,
+            DP_DEFAULT_RATIO,
+            DP_DEFAULT_THRESHOLD_DB,
+            DP_DEFAULT_GAIN_DB);
+}
+
+void DPLimiter::init(bool inUse, bool enabled, uint32_t linkGroup, float attackTime, float releaseTime,
+        float ratio, float threshold, float postGain) {
+    DPStage::init(inUse, enabled);
+    setLinkGroup(linkGroup);
+    setAttackTime(attackTime);
+    setReleaseTime(releaseTime);
+    setRatio(ratio);
+    setThreshold(threshold);
+    setPostGain(postGain);
+}
+
+//----
+DPChannel::DPChannel() : mInitialized(false), mInputGainDb(0), mOutputGainDb(0),
+        mPreEqInUse(false), mMbcInUse(false), mPostEqInUse(false), mLimiterInUse(false) {
+}
+
+void DPChannel::init(float inputGain, bool preEqInUse, uint32_t preEqBandCount,
+        bool mbcInUse, uint32_t mbcBandCount, bool postEqInUse, uint32_t postEqBandCount,
+        bool limiterInUse) {
+    setInputGain(inputGain);
+    mPreEqInUse = preEqInUse;
+    mMbcInUse = mbcInUse;
+    mPostEqInUse = postEqInUse;
+    mLimiterInUse = limiterInUse;
+
+    mPreEq.init(mPreEqInUse, false, preEqBandCount);
+    mMbc.init(mMbcInUse, false, mbcBandCount);
+    mPostEq.init(mPostEqInUse, false, postEqBandCount);
+    mLimiter.init(mLimiterInUse, false, 0, 50, 120, 2, -30, 0);
+    mInitialized = true;
+}
+
+DPEq* DPChannel::getPreEq() {
+    if (!mInitialized) {
+        return NULL;
+    }
+    return &mPreEq;
+}
+
+DPMbc* DPChannel::getMbc() {
+    if (!mInitialized) {
+        return NULL;
+    }
+    return &mMbc;
+}
+
+DPEq* DPChannel::getPostEq() {
+    if (!mInitialized) {
+        return NULL;
+    }
+    return &mPostEq;
+}
+
+DPLimiter* DPChannel::getLimiter() {
+    if (!mInitialized) {
+        return NULL;
+    }
+    return &mLimiter;
+}
+
+void DPChannel::setLimiter(DPLimiter &limiter) {
+    if (!mInitialized) {
+        return;
+    }
+    mLimiter = limiter;
+}
+
+//----
+DPBase::DPBase() : mInitialized(false), mChannelCount(0), mPreEqInUse(false), mPreEqBandCount(0),
+        mMbcInUse(false), mMbcBandCount(0), mPostEqInUse(false), mPostEqBandCount(0),
+        mLimiterInUse(false) {
+}
+
+void DPBase::init(uint32_t channelCount, bool preEqInUse, uint32_t preEqBandCount,
+        bool mbcInUse, uint32_t mbcBandCount, bool postEqInUse, uint32_t postEqBandCount,
+        bool limiterInUse) {
+    ALOGV("DPBase::init");
+    mChannelCount = channelCount;
+    mPreEqInUse = preEqInUse;
+    mPreEqBandCount = preEqBandCount;
+    mMbcInUse = mbcInUse;
+    mMbcBandCount = mbcBandCount;
+    mPostEqInUse = postEqInUse;
+    mPostEqBandCount = postEqBandCount;
+    mLimiterInUse = limiterInUse;
+    mChannel.resize(mChannelCount);
+    for (size_t ch = 0; ch < mChannelCount; ch++) {
+        mChannel[ch].init(0, preEqInUse, preEqBandCount, mbcInUse, mbcBandCount,
+                postEqInUse, postEqBandCount, limiterInUse);
+    }
+    mInitialized = true;
+}
+
+DPChannel* DPBase::getChannel(uint32_t channelIndex) {
+    if (!mInitialized || channelIndex < 0 || channelIndex >= mChannel.size()) {
+        return NULL;
+    }
+    return & mChannel[channelIndex];
+}
+
+} //namespace dp_fx

diff --git a/media/libeffects/dynamicsproc/dsp/DPBase.h b/media/libeffects/dynamicsproc/dsp/DPBase.h
new file mode 100644
index 0000000..e74f91d
--- /dev/null
+++ b/media/libeffects/dynamicsproc/dsp/DPBase.h

@@ -0,0 +1,362 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef DPBASE_H_
+#define DPBASE_H_
+
+
+#include <stdint.h>
+#include <cmath>
+#include <vector>
+#include <android/log.h>
+
+namespace dp_fx {
+
+#define DP_DEFAULT_BAND_ENABLED false
+#define DP_DEFAULT_BAND_CUTOFF_FREQUENCY_HZ 1000
+#define DP_DEFAULT_ATTACK_TIME_MS 50
+#define DP_DEFAULT_RELEASE_TIME_MS 120
+#define DP_DEFAULT_RATIO 2
+#define DP_DEFAULT_THRESHOLD_DB -30
+#define DP_DEFAULT_KNEE_WIDTH_DB 0
+#define DP_DEFAULT_NOISE_GATE_THRESHOLD_DB -90
+#define DP_DEFAULT_EXPANDER_RATIO 1
+#define DP_DEFAULT_GAIN_DB 0
+#define DP_DEFAULT_STAGE_INUSE false
+#define DP_DEFAULT_STAGE_ENABLED false
+#define DP_DEFAULT_LINK_GROUP 0
+
+
+
+class DPStage {
+public:
+    DPStage();
+    ~DPStage() = default;
+    void init(bool inUse, bool enabled);
+    bool isInUse() const {
+        return mInUse;
+    }
+    bool isEnabled() const {
+        return mEnabled;
+    }
+    void setEnabled(bool enabled) {
+        mEnabled = enabled;
+    }
+private:
+    bool mInUse;
+    bool mEnabled;
+};
+
+class DPBandStage : public DPStage {
+public:
+    DPBandStage();
+    ~DPBandStage() = default;
+    void init(bool inUse, bool enabled, int bandCount);
+    uint32_t getBandCount() const {
+        return mBandCount;
+    }
+    void setBandCount(uint32_t bandCount) {
+        mBandCount = bandCount;
+    }
+private:
+    uint32_t mBandCount;
+};
+
+class DPBandBase {
+public:
+    DPBandBase();
+    ~DPBandBase() = default;
+    void init(bool enabled, float cutoffFrequency);
+    bool isEnabled() const {
+        return mEnabled;
+    }
+    void setEnabled(bool enabled) {
+        mEnabled = enabled;
+    }
+    float getCutoffFrequency() const {
+        return mCutoofFrequencyHz;
+    }
+    void setCutoffFrequency(float cutoffFrequency) {
+        mCutoofFrequencyHz = cutoffFrequency;
+    }
+private:
+    bool mEnabled;
+    float mCutoofFrequencyHz;
+};
+
+class DPEqBand : public DPBandBase {
+public:
+    DPEqBand();
+    ~DPEqBand() = default;
+    void init(bool enabled, float cutoffFrequency, float gain);
+    float getGain() const;
+    void setGain(float gain);
+private:
+    float mGainDb;
+};
+
+class DPMbcBand : public DPBandBase {
+public:
+    DPMbcBand();
+    ~DPMbcBand() = default;
+    void init(bool enabled, float cutoffFrequency, float attackTime, float releaseTime,
+            float ratio, float threshold, float kneeWidth, float noiseGateThreshold,
+            float expanderRatio, float preGain, float postGain);
+    float getAttackTime() const {
+        return mAttackTimeMs;
+    }
+    void setAttackTime(float attackTime) {
+        mAttackTimeMs = attackTime;
+    }
+    float getReleaseTime() const {
+        return mReleaseTimeMs;
+    }
+    void setReleaseTime(float releaseTime) {
+        mReleaseTimeMs = releaseTime;
+    }
+    float getRatio() const {
+        return mRatio;
+    }
+    void setRatio(float ratio) {
+        mRatio = ratio;
+    }
+    float getThreshold() const {
+        return mThresholdDb;
+    }
+    void setThreshold(float threshold) {
+        mThresholdDb = threshold;
+    }
+    float getKneeWidth() const {
+        return mKneeWidthDb;
+    }
+    void setKneeWidth(float kneeWidth) {
+        mKneeWidthDb = kneeWidth;
+    }
+    float getNoiseGateThreshold() const {
+        return mNoiseGateThresholdDb;
+    }
+    void setNoiseGateThreshold(float noiseGateThreshold) {
+        mNoiseGateThresholdDb = noiseGateThreshold;
+    }
+    float getExpanderRatio() const {
+        return mExpanderRatio;
+    }
+    void setExpanderRatio(float expanderRatio) {
+        mExpanderRatio = expanderRatio;
+    }
+    float getPreGain() const {
+        return mPreGainDb;
+    }
+    void setPreGain(float preGain) {
+        mPreGainDb = preGain;
+    }
+    float getPostGain() const {
+        return mPostGainDb;
+    }
+    void setPostGain(float postGain) {
+        mPostGainDb = postGain;
+    }
+private:
+    float mAttackTimeMs;
+    float mReleaseTimeMs;
+    float mRatio;
+    float mThresholdDb;
+    float mKneeWidthDb;
+    float mNoiseGateThresholdDb;
+    float mExpanderRatio;
+    float mPreGainDb;
+    float mPostGainDb;
+};
+
+class DPEq : public DPBandStage {
+public:
+    DPEq();
+    ~DPEq() = default;
+    void init(bool inUse, bool enabled, uint32_t bandCount);
+    DPEqBand * getBand(uint32_t band);
+    void setBand(uint32_t band, DPEqBand &src);
+private:
+    std::vector<DPEqBand> mBands;
+};
+
+class DPMbc : public DPBandStage {
+public:
+    DPMbc();
+    ~DPMbc() = default;
+    void init(bool inUse, bool enabled, uint32_t bandCount);
+    DPMbcBand * getBand(uint32_t band);
+    void setBand(uint32_t band, DPMbcBand &src);
+private:
+    std::vector<DPMbcBand> mBands;
+};
+
+class DPLimiter : public DPStage {
+public:
+    DPLimiter();
+    ~DPLimiter() = default;
+    void init(bool inUse, bool enabled, uint32_t linkGroup, float attackTime, float releaseTime,
+            float ratio, float threshold, float postGain);
+    uint32_t getLinkGroup() const {
+        return mLinkGroup;
+    }
+    void setLinkGroup(uint32_t linkGroup) {
+        mLinkGroup = linkGroup;
+    }
+    float getAttackTime() const {
+        return mAttackTimeMs;
+    }
+    void setAttackTime(float attackTime) {
+        mAttackTimeMs = attackTime;
+    }
+    float getReleaseTime() const {
+        return mReleaseTimeMs;
+    }
+    void setReleaseTime(float releaseTime) {
+        mReleaseTimeMs = releaseTime;
+    }
+    float getRatio() const {
+        return mRatio;
+    }
+    void setRatio(float ratio) {
+        mRatio = ratio;
+    }
+    float getThreshold() const {
+        return mThresholdDb;
+    }
+    void setThreshold(float threshold) {
+        mThresholdDb = threshold;
+    }
+    float getPostGain() const {
+        return mPostGainDb;
+    }
+    void setPostGain(float postGain) {
+        mPostGainDb = postGain;
+    }
+private:
+    uint32_t mLinkGroup;
+    float mAttackTimeMs;
+    float mReleaseTimeMs;
+    float mRatio;
+    float mThresholdDb;
+    float mPostGainDb;
+};
+
+class DPChannel {
+public:
+    DPChannel();
+    ~DPChannel() = default;
+    void init(float inputGain, bool preEqInUse, uint32_t preEqBandCount,
+            bool mbcInUse, uint32_t mbcBandCount, bool postEqInUse, uint32_t postEqBandCount,
+            bool limiterInUse);
+
+    float getInputGain() const {
+        if (!mInitialized) {
+            return 0;
+        }
+        return mInputGainDb;
+    }
+    void setInputGain(float gain) {
+        mInputGainDb = gain;
+    }
+
+    float getOutputGain() const {
+        if (!mInitialized) {
+            return 0;
+        }
+        return mOutputGainDb;
+    }
+    void setOutputGain(float gain) {
+        mOutputGainDb = gain;
+    }
+
+    DPEq* getPreEq();
+    DPMbc* getMbc();
+    DPEq* getPostEq();
+    DPLimiter *getLimiter();
+    void setLimiter(DPLimiter &limiter);
+
+private:
+    bool mInitialized;
+    float mInputGainDb;
+    float mOutputGainDb;
+
+    DPEq mPreEq;
+    DPMbc mMbc;
+    DPEq mPostEq;
+    DPLimiter mLimiter;
+
+    bool mPreEqInUse;
+    bool mMbcInUse;
+    bool mPostEqInUse;
+    bool mLimiterInUse;
+};
+
+class DPBase {
+public:
+    DPBase();
+    virtual ~DPBase() = default;
+
+    void init(uint32_t channelCount, bool preEqInUse, uint32_t preEqBandCount,
+            bool mbcInUse, uint32_t mbcBandCount, bool postEqInUse, uint32_t postEqBandCount,
+            bool limiterInUse);
+    virtual size_t processSamples(const float *in, float *out, size_t samples) = 0;
+    virtual void reset() = 0;
+
+    DPChannel* getChannel(uint32_t channelIndex);
+    uint32_t getChannelCount() const {
+        return mChannelCount;
+    }
+    uint32_t getPreEqBandCount() const {
+        return mPreEqBandCount;
+    }
+    uint32_t getMbcBandCount() const {
+        return mMbcBandCount;
+    }
+    uint32_t getPostEqBandCount() const {
+        return mPostEqBandCount;
+    }
+    bool isPreEQInUse() const {
+        return mPreEqInUse;
+    }
+    bool isMbcInUse() const {
+        return mMbcInUse;
+    }
+    bool isPostEqInUse() const {
+        return mPostEqInUse;
+    }
+    bool isLimiterInUse() const {
+        return mLimiterInUse;
+    }
+
+private:
+    bool mInitialized;
+    //general
+    uint32_t mChannelCount;
+    bool mPreEqInUse;
+    uint32_t mPreEqBandCount;
+    bool mMbcInUse;
+    uint32_t mMbcBandCount;
+    bool mPostEqInUse;
+    uint32_t mPostEqBandCount;
+    bool mLimiterInUse;
+
+    std::vector<DPChannel> mChannel;
+};
+
+} //namespace dp_fx
+
+
+#endif  // DPBASE_H_

diff --git a/media/libeffects/dynamicsproc/dsp/DPFrequency.cpp b/media/libeffects/dynamicsproc/dsp/DPFrequency.cpp
new file mode 100644
index 0000000..d06fd70
--- /dev/null
+++ b/media/libeffects/dynamicsproc/dsp/DPFrequency.cpp

@@ -0,0 +1,675 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "DPFrequency"
+//#define LOG_NDEBUG 0
+
+#include <log/log.h>
+#include "DPFrequency.h"
+#include <algorithm>
+
+namespace dp_fx {
+
+using Eigen::MatrixXd;
+#define MAX_BLOCKSIZE 16384 //For this implementation
+#define MIN_BLOCKSIZE 8
+
+#define CIRCULAR_BUFFER_UPSAMPLE 4  //4 times buffer size
+
+static constexpr float MIN_ENVELOPE = 1e-6f; //-120 dB
+//helper functionS
+static inline bool isPowerOf2(unsigned long n) {
+    return (n & (n - 1)) == 0;
+}
+static constexpr float EPSILON = 0.0000001f;
+
+static inline bool isZero(float f) {
+    return fabs(f) <= EPSILON;
+}
+
+template <class T>
+bool compareEquality(T a, T b) {
+    return (a == b);
+}
+
+template <> bool compareEquality<float>(float a, float b) {
+    return isZero(a - b);
+}
+
+//TODO: avoid using macro for estimating change and assignment.
+#define IS_CHANGED(c, a, b) { c |= !compareEquality(a,b); \
+    (a) = (b); }
+
+//ChannelBuffers helper
+void ChannelBuffer::initBuffers(unsigned int blockSize, unsigned int overlapSize,
+        unsigned int halfFftSize, unsigned int samplingRate, DPBase &dpBase) {
+    ALOGV("ChannelBuffer::initBuffers blockSize %d, overlap %d, halfFft %d",
+            blockSize, overlapSize, halfFftSize);
+
+    mSamplingRate = samplingRate;
+    mBlockSize = blockSize;
+
+    cBInput.resize(mBlockSize * CIRCULAR_BUFFER_UPSAMPLE);
+    cBOutput.resize(mBlockSize * CIRCULAR_BUFFER_UPSAMPLE);
+
+    //fill input with half block size...
+    for (unsigned int k = 0; k < mBlockSize/2; k++) {
+        cBInput.write(0);
+    }
+
+    //temp vectors
+    input.resize(mBlockSize);
+    output.resize(mBlockSize);
+    outTail.resize(overlapSize);
+
+    //module vectors
+    mPreEqFactorVector.resize(halfFftSize, 1.0);
+    mPostEqFactorVector.resize(halfFftSize, 1.0);
+
+    mPreEqBands.resize(dpBase.getPreEqBandCount());
+    mMbcBands.resize(dpBase.getMbcBandCount());
+    mPostEqBands.resize(dpBase.getPostEqBandCount());
+    ALOGV("mPreEqBands %zu, mMbcBands %zu, mPostEqBands %zu",mPreEqBands.size(),
+            mMbcBands.size(), mPostEqBands.size());
+
+    DPChannel *pChannel = dpBase.getChannel(0);
+    if (pChannel != nullptr) {
+        mPreEqInUse = pChannel->getPreEq()->isInUse();
+        mMbcInUse = pChannel->getMbc()->isInUse();
+        mPostEqInUse = pChannel->getPostEq()->isInUse();
+        mLimiterInUse = pChannel->getLimiter()->isInUse();
+    }
+
+    mLimiterParams.linkGroup = -1; //no group.
+}
+
+void ChannelBuffer::computeBinStartStop(BandParams &bp, size_t binStart) {
+
+    bp.binStart = binStart;
+    bp.binStop = (int)(0.5 + bp.freqCutoffHz * mBlockSize / mSamplingRate);
+}
+
+//== LinkedLimiters Helper
+void LinkedLimiters::reset() {
+    mGroupsMap.clear();
+}
+
+void LinkedLimiters::update(int32_t group, int index) {
+    mGroupsMap[group].push_back(index);
+}
+
+void LinkedLimiters::remove(int index) {
+    //check all groups and if index is found, remove it.
+    //if group is empty afterwards, remove it.
+    for (auto it = mGroupsMap.begin(); it != mGroupsMap.end(); ) {
+        for (auto itIndex = it->second.begin(); itIndex != it->second.end(); ) {
+            if (*itIndex == index) {
+                itIndex = it->second.erase(itIndex);
+            } else {
+                ++itIndex;
+            }
+        }
+        if (it->second.size() == 0) {
+            it = mGroupsMap.erase(it);
+        } else {
+            ++it;
+        }
+    }
+}
+
+//== DPFrequency
+void DPFrequency::reset() {
+}
+
+size_t DPFrequency::getMinBockSize() {
+    return MIN_BLOCKSIZE;
+}
+
+size_t DPFrequency::getMaxBockSize() {
+    return MAX_BLOCKSIZE;
+}
+
+void DPFrequency::configure(size_t blockSize, size_t overlapSize,
+        size_t samplingRate) {
+    ALOGV("configure");
+    mBlockSize = blockSize;
+    if (mBlockSize > MAX_BLOCKSIZE) {
+        mBlockSize = MAX_BLOCKSIZE;
+    } else if (mBlockSize < MIN_BLOCKSIZE) {
+        mBlockSize = MIN_BLOCKSIZE;
+    } else {
+        if (!isPowerOf2(blockSize)) {
+            //find next highest power of 2.
+            mBlockSize = 1 << (32 - __builtin_clz(blockSize));
+        }
+    }
+
+    mHalfFFTSize = 1 + mBlockSize / 2; //including Nyquist bin
+    mOverlapSize = std::min(overlapSize, mBlockSize/2);
+
+    int channelcount = getChannelCount();
+    mSamplingRate = samplingRate;
+    mChannelBuffers.resize(channelcount);
+    for (int ch = 0; ch < channelcount; ch++) {
+        mChannelBuffers[ch].initBuffers(mBlockSize, mOverlapSize, mHalfFFTSize,
+                mSamplingRate, *this);
+    }
+
+    //effective number of frames processed per second
+    mBlocksPerSecond = (float)mSamplingRate / (mBlockSize - mOverlapSize);
+
+    fill_window(mVWindow, RDSP_WINDOW_HANNING_FLAT_TOP, mBlockSize, mOverlapSize);
+
+    //compute window rms for energy compensation
+    mWindowRms = 0;
+    for (size_t i = 0; i < mVWindow.size(); i++) {
+        mWindowRms += mVWindow[i] * mVWindow[i];
+    }
+
+    //Making sure window rms is not zero.
+    mWindowRms = std::max(sqrt(mWindowRms / mVWindow.size()), MIN_ENVELOPE);
+}
+
+void DPFrequency::updateParameters(ChannelBuffer &cb, int channelIndex) {
+    DPChannel *pChannel = getChannel(channelIndex);
+
+    if (pChannel == nullptr) {
+        ALOGE("Error: updateParameters null DPChannel %d", channelIndex);
+        return;
+    }
+
+    //===Input Gain and preEq
+    {
+        bool changed = false;
+        IS_CHANGED(changed, cb.inputGainDb, pChannel->getInputGain());
+        //===EqPre
+        if (cb.mPreEqInUse) {
+            DPEq *pPreEq = pChannel->getPreEq();
+            if (pPreEq == nullptr) {
+                ALOGE("Error: updateParameters null PreEq for channel: %d", channelIndex);
+                return;
+            }
+            IS_CHANGED(changed, cb.mPreEqEnabled, pPreEq->isEnabled());
+            if (cb.mPreEqEnabled) {
+                for (unsigned int b = 0; b < getPreEqBandCount(); b++) {
+                    DPEqBand *pEqBand = pPreEq->getBand(b);
+                    if (pEqBand == nullptr) {
+                        ALOGE("Error: updateParameters null PreEqBand for band %d", b);
+                        return; //failed.
+                    }
+                    ChannelBuffer::EqBandParams *pEqBandParams = &cb.mPreEqBands[b];
+                    IS_CHANGED(changed, pEqBandParams->enabled, pEqBand->isEnabled());
+                    IS_CHANGED(changed, pEqBandParams->freqCutoffHz,
+                            pEqBand->getCutoffFrequency());
+                    IS_CHANGED(changed, pEqBandParams->gainDb, pEqBand->getGain());
+                }
+            }
+        }
+
+        if (changed) {
+            float inputGainFactor = dBtoLinear(cb.inputGainDb);
+            if (cb.mPreEqInUse && cb.mPreEqEnabled) {
+                ALOGV("preEq changed, recomputing! channel %d", channelIndex);
+                size_t binNext = 0;
+                for (unsigned int b = 0; b < getPreEqBandCount(); b++) {
+                    ChannelBuffer::EqBandParams *pEqBandParams = &cb.mPreEqBands[b];
+
+                    //frequency translation
+                    cb.computeBinStartStop(*pEqBandParams, binNext);
+                    binNext = pEqBandParams->binStop + 1;
+                    float factor = dBtoLinear(pEqBandParams->gainDb);
+                    if (!pEqBandParams->enabled) {
+                        factor = inputGainFactor;
+                    }
+                    for (size_t k = pEqBandParams->binStart;
+                            k <= pEqBandParams->binStop && k < mHalfFFTSize; k++) {
+                        cb.mPreEqFactorVector[k] = factor * inputGainFactor;
+                    }
+                }
+            } else {
+                ALOGV("only input gain changed, recomputing!");
+                //populate PreEq factor with input gain factor.
+                for (size_t k = 0; k < mHalfFFTSize; k++) {
+                    cb.mPreEqFactorVector[k] = inputGainFactor;
+                }
+            }
+        }
+    } //inputGain and preEq
+
+    //===EqPost
+    if (cb.mPostEqInUse) {
+        bool changed = false;
+
+        DPEq *pPostEq = pChannel->getPostEq();
+        if (pPostEq == nullptr) {
+            ALOGE("Error: updateParameters null postEq for channel: %d", channelIndex);
+            return; //failed.
+        }
+        IS_CHANGED(changed, cb.mPostEqEnabled, pPostEq->isEnabled());
+        if (cb.mPostEqEnabled) {
+            for (unsigned int b = 0; b < getPostEqBandCount(); b++) {
+                DPEqBand *pEqBand = pPostEq->getBand(b);
+                if (pEqBand == nullptr) {
+                    ALOGE("Error: updateParameters PostEqBand NULL for band %d", b);
+                    return; //failed.
+                }
+                ChannelBuffer::EqBandParams *pEqBandParams = &cb.mPostEqBands[b];
+                IS_CHANGED(changed, pEqBandParams->enabled, pEqBand->isEnabled());
+                IS_CHANGED(changed, pEqBandParams->freqCutoffHz,
+                        pEqBand->getCutoffFrequency());
+                IS_CHANGED(changed, pEqBandParams->gainDb, pEqBand->getGain());
+            }
+            if (changed) {
+                ALOGV("postEq changed, recomputing! channel %d", channelIndex);
+                size_t binNext = 0;
+                for (unsigned int b = 0; b < getPostEqBandCount(); b++) {
+                    ChannelBuffer::EqBandParams *pEqBandParams = &cb.mPostEqBands[b];
+
+                    //frequency translation
+                    cb.computeBinStartStop(*pEqBandParams, binNext);
+                    binNext = pEqBandParams->binStop + 1;
+                    float factor = dBtoLinear(pEqBandParams->gainDb);
+                    if (!pEqBandParams->enabled) {
+                        factor = 1.0;
+                    }
+                    for (size_t k = pEqBandParams->binStart;
+                            k <= pEqBandParams->binStop && k < mHalfFFTSize; k++) {
+                        cb.mPostEqFactorVector[k] = factor;
+                    }
+                }
+            }
+        } //enabled
+    }
+
+    //===MBC
+    if (cb.mMbcInUse) {
+        DPMbc *pMbc = pChannel->getMbc();
+        if (pMbc == nullptr) {
+            ALOGE("Error: updateParameters Mbc NULL for channel: %d", channelIndex);
+            return;
+        }
+        cb.mMbcEnabled = pMbc->isEnabled();
+        if (cb.mMbcEnabled) {
+            bool changed = false;
+            for (unsigned int b = 0; b < getMbcBandCount(); b++) {
+                DPMbcBand *pMbcBand = pMbc->getBand(b);
+                if (pMbcBand == nullptr) {
+                    ALOGE("Error: updateParameters MbcBand NULL for band %d", b);
+                    return; //failed.
+                }
+                ChannelBuffer::MbcBandParams *pMbcBandParams = &cb.mMbcBands[b];
+                pMbcBandParams->enabled = pMbcBand->isEnabled();
+                IS_CHANGED(changed, pMbcBandParams->freqCutoffHz,
+                        pMbcBand->getCutoffFrequency());
+
+                pMbcBandParams->gainPreDb = pMbcBand->getPreGain();
+                pMbcBandParams->gainPostDb = pMbcBand->getPostGain();
+                pMbcBandParams->attackTimeMs = pMbcBand->getAttackTime();
+                pMbcBandParams->releaseTimeMs = pMbcBand->getReleaseTime();
+                pMbcBandParams->ratio = pMbcBand->getRatio();
+                pMbcBandParams->thresholdDb = pMbcBand->getThreshold();
+                pMbcBandParams->kneeWidthDb = pMbcBand->getKneeWidth();
+                pMbcBandParams->noiseGateThresholdDb = pMbcBand->getNoiseGateThreshold();
+                pMbcBandParams->expanderRatio = pMbcBand->getExpanderRatio();
+
+            }
+
+            if (changed) {
+                ALOGV("mbc changed, recomputing! channel %d", channelIndex);
+                size_t binNext= 0;
+                for (unsigned int b = 0; b < getMbcBandCount(); b++) {
+                    ChannelBuffer::MbcBandParams *pMbcBandParams = &cb.mMbcBands[b];
+
+                    pMbcBandParams->previousEnvelope = 0;
+
+                    //frequency translation
+                    cb.computeBinStartStop(*pMbcBandParams, binNext);
+                    binNext = pMbcBandParams->binStop + 1;
+                }
+            }
+        }
+    }
+
+    //===Limiter
+    if (cb.mLimiterInUse) {
+        bool changed = false;
+        DPLimiter *pLimiter = pChannel->getLimiter();
+        if (pLimiter == nullptr) {
+            ALOGE("Error: updateParameters Limiter NULL for channel: %d", channelIndex);
+            return;
+        }
+        cb.mLimiterEnabled = pLimiter->isEnabled();
+        if (cb.mLimiterEnabled) {
+            IS_CHANGED(changed, cb.mLimiterParams.linkGroup ,
+                    (int32_t)pLimiter->getLinkGroup());
+            cb.mLimiterParams.attackTimeMs = pLimiter->getAttackTime();
+            cb.mLimiterParams.releaseTimeMs = pLimiter->getReleaseTime();
+            cb.mLimiterParams.ratio = pLimiter->getRatio();
+            cb.mLimiterParams.thresholdDb = pLimiter->getThreshold();
+            cb.mLimiterParams.postGainDb = pLimiter->getPostGain();
+        }
+
+        if (changed) {
+            ALOGV("limiter changed, recomputing linkGroups for %d", channelIndex);
+            mLinkedLimiters.remove(channelIndex); //in case it was already there.
+            mLinkedLimiters.update(cb.mLimiterParams.linkGroup, channelIndex);
+        }
+    }
+
+    //=== Output Gain
+    cb.outputGainDb = pChannel->getOutputGain();
+}
+
+size_t DPFrequency::processSamples(const float *in, float *out, size_t samples) {
+       const float *pIn = in;
+       float *pOut = out;
+
+       int channelCount = mChannelBuffers.size();
+       if (channelCount < 1) {
+           ALOGW("warning: no Channels ready for processing");
+           return 0;
+       }
+
+       //**Check if parameters have changed and update
+       for (int ch = 0; ch < channelCount; ch++) {
+           updateParameters(mChannelBuffers[ch], ch);
+       }
+
+       //**separate into channels
+       for (size_t k = 0; k < samples; k += channelCount) {
+           for (int ch = 0; ch < channelCount; ch++) {
+               mChannelBuffers[ch].cBInput.write(*pIn++);
+           }
+       }
+
+       //**process all channelBuffers
+       processChannelBuffers(mChannelBuffers);
+
+       //** estimate how much data is available in ALL channels
+       size_t available = mChannelBuffers[0].cBOutput.availableToRead();
+       for (int ch = 1; ch < channelCount; ch++) {
+           available = std::min(available, mChannelBuffers[ch].cBOutput.availableToRead());
+       }
+
+       //** make sure to output just what the buffer can handle
+       if (available > samples/channelCount) {
+           available = samples/channelCount;
+       }
+
+       //**Prepend zeroes if necessary
+       size_t fill = samples - (channelCount * available);
+       for (size_t k = 0; k < fill; k++) {
+           *pOut++ = 0;
+       }
+
+       //**interleave channels
+       for (size_t k = 0; k < available; k++) {
+           for (int ch = 0; ch < channelCount; ch++) {
+               *pOut++ = mChannelBuffers[ch].cBOutput.read();
+           }
+       }
+
+       return samples;
+}
+
+size_t DPFrequency::processChannelBuffers(CBufferVector &channelBuffers) {
+    const int channelCount = channelBuffers.size();
+    size_t processedSamples = 0;
+    size_t processFrames = mBlockSize - mOverlapSize;
+
+    size_t available = channelBuffers[0].cBInput.availableToRead();
+    for (int ch = 1; ch < channelCount; ch++) {
+        available = std::min(available, channelBuffers[ch].cBInput.availableToRead());
+    }
+
+    while (available >= processFrames) {
+        //First pass
+        for (int ch = 0; ch < channelCount; ch++) {
+            ChannelBuffer * pCb = &channelBuffers[ch];
+            //move tail of previous
+            std::copy(pCb->input.begin() + processFrames,
+                    pCb->input.end(),
+                    pCb->input.begin());
+
+            //read new available data
+            for (unsigned int k = 0; k < processFrames; k++) {
+                pCb->input[mOverlapSize + k] = pCb->cBInput.read();
+            }
+            //first stages: fft, preEq, mbc, postEq and start of Limiter
+            processedSamples += processFirstStages(*pCb);
+        }
+
+        //**compute linked limiters and update levels if needed
+        processLinkedLimiters(channelBuffers);
+
+        //final pass.
+        for (int ch = 0; ch < channelCount; ch++) {
+            ChannelBuffer * pCb = &channelBuffers[ch];
+
+            //linked limiter and ifft
+            processLastStages(*pCb);
+
+            //mix tail (and capture new tail
+            for (unsigned int k = 0; k < mOverlapSize; k++) {
+                pCb->output[k] += pCb->outTail[k];
+                pCb->outTail[k] = pCb->output[processFrames + k]; //new tail
+            }
+
+            //output data
+            for (unsigned int k = 0; k < processFrames; k++) {
+                pCb->cBOutput.write(pCb->output[k]);
+            }
+        }
+        available -= processFrames;
+    }
+    return processedSamples;
+}
+size_t DPFrequency::processFirstStages(ChannelBuffer &cb) {
+
+    //##apply window
+    Eigen::Map<Eigen::VectorXf> eWindow(&mVWindow[0], mVWindow.size());
+    Eigen::Map<Eigen::VectorXf> eInput(&cb.input[0], cb.input.size());
+
+    Eigen::VectorXf eWin = eInput.cwiseProduct(eWindow); //apply window
+
+    //##fft
+    //Note: we are using eigen with the default scaling, which ensures that
+    //  IFFT( FFT(x) ) = x.
+    // TODO: optimize by using the noscale option, and compensate with dB scale offsets
+    mFftServer.fwd(cb.complexTemp, eWin);
+
+    size_t cSize = cb.complexTemp.size();
+    size_t maxBin = std::min(cSize/2, mHalfFFTSize);
+
+    //== EqPre (always runs)
+    for (size_t k = 0; k < maxBin; k++) {
+        cb.complexTemp[k] *= cb.mPreEqFactorVector[k];
+    }
+
+    //== MBC
+    if (cb.mMbcInUse && cb.mMbcEnabled) {
+        for (size_t band = 0; band < cb.mMbcBands.size(); band++) {
+            ChannelBuffer::MbcBandParams *pMbcBandParams = &cb.mMbcBands[band];
+            float fEnergySum = 0;
+
+            //apply pre gain.
+            float preGainFactor = dBtoLinear(pMbcBandParams->gainPreDb);
+            float preGainSquared = preGainFactor * preGainFactor;
+
+            for (size_t k = pMbcBandParams->binStart; k <= pMbcBandParams->binStop; k++) {
+                fEnergySum += std::norm(cb.complexTemp[k]) * preGainSquared; //mag squared
+            }
+
+            //Eigen FFT is full spectrum, even if the source was real data.
+            // Each half spectrum has half the energy. This is taken into account with the * 2
+            // factor in the energy computations.
+            // energy = sqrt(sum_components_squared) number_points
+            // in here, the fEnergySum is duplicated to account for the second half spectrum,
+            // and the windowRms is used to normalize by the expected energy reduction
+            // caused by the window used (expected for steady state signals)
+            fEnergySum = sqrt(fEnergySum * 2) / (mBlockSize * mWindowRms);
+
+            // updates computed per frame advance.
+            float fTheta = 0.0;
+            float fFAttSec = pMbcBandParams->attackTimeMs / 1000; //in seconds
+            float fFRelSec = pMbcBandParams->releaseTimeMs / 1000; //in seconds
+
+            if (fEnergySum > pMbcBandParams->previousEnvelope) {
+                fTheta = exp(-1.0 / (fFAttSec * mBlocksPerSecond));
+            } else {
+                fTheta = exp(-1.0 / (fFRelSec * mBlocksPerSecond));
+            }
+
+            float fEnv = (1.0 - fTheta) * fEnergySum + fTheta * pMbcBandParams->previousEnvelope;
+            //preserve for next iteration
+            pMbcBandParams->previousEnvelope = fEnv;
+
+            if (fEnv < MIN_ENVELOPE) {
+                fEnv = MIN_ENVELOPE;
+            }
+            const float envDb = linearToDb(fEnv);
+            float newLevelDb = envDb;
+            //using shorter variables for code clarity
+            const float thresholdDb = pMbcBandParams->thresholdDb;
+            const float ratio = pMbcBandParams->ratio;
+            const float kneeWidthDbHalf = pMbcBandParams->kneeWidthDb / 2;
+            const float noiseGateThresholdDb = pMbcBandParams->noiseGateThresholdDb;
+            const float expanderRatio = pMbcBandParams->expanderRatio;
+
+            //find segment
+            if (envDb > thresholdDb + kneeWidthDbHalf) {
+                //compression segment
+                newLevelDb = envDb + ((1 / ratio) - 1) * (envDb - thresholdDb);
+            } else if (envDb > thresholdDb - kneeWidthDbHalf) {
+                //knee-compression segment
+                float temp = (envDb - thresholdDb + kneeWidthDbHalf);
+                newLevelDb = envDb + ((1 / ratio) - 1) *
+                        temp * temp / (kneeWidthDbHalf * 4);
+            } else if (envDb < noiseGateThresholdDb) {
+                //expander segment
+                newLevelDb = noiseGateThresholdDb -
+                        expanderRatio * (noiseGateThresholdDb - envDb);
+            }
+
+            float newFactor = dBtoLinear(newLevelDb - envDb);
+
+            //apply post gain.
+            newFactor *= dBtoLinear(pMbcBandParams->gainPostDb);
+
+            //apply to this band
+            for (size_t k = pMbcBandParams->binStart; k <= pMbcBandParams->binStop; k++) {
+                cb.complexTemp[k] *= newFactor;
+            }
+
+        } //end per band process
+
+    } //end MBC
+
+    //== EqPost
+    if (cb.mPostEqInUse && cb.mPostEqEnabled) {
+        for (size_t k = 0; k < maxBin; k++) {
+            cb.complexTemp[k] *= cb.mPostEqFactorVector[k];
+        }
+    }
+
+    //== Limiter. First Pass
+    if (cb.mLimiterInUse && cb.mLimiterEnabled) {
+        float fEnergySum = 0;
+        for (size_t k = 0; k < maxBin; k++) {
+            fEnergySum += std::norm(cb.complexTemp[k]);
+        }
+
+        //see explanation above for energy computation logic
+        fEnergySum = sqrt(fEnergySum * 2) / (mBlockSize * mWindowRms);
+        float fTheta = 0.0;
+        float fFAttSec = cb.mLimiterParams.attackTimeMs / 1000; //in seconds
+        float fFRelSec = cb.mLimiterParams.releaseTimeMs / 1000; //in seconds
+
+        if (fEnergySum > cb.mLimiterParams.previousEnvelope) {
+            fTheta = exp(-1.0 / (fFAttSec * mBlocksPerSecond));
+        } else {
+            fTheta = exp(-1.0 / (fFRelSec * mBlocksPerSecond));
+        }
+
+        float fEnv = (1.0 - fTheta) * fEnergySum + fTheta * cb.mLimiterParams.previousEnvelope;
+        //preserve for next iteration
+        cb.mLimiterParams.previousEnvelope = fEnv;
+
+        const float envDb = linearToDb(fEnv);
+        float newFactorDb = 0;
+        //using shorter variables for code clarity
+        const float thresholdDb = cb.mLimiterParams.thresholdDb;
+        const float ratio = cb.mLimiterParams.ratio;
+
+        if (envDb > thresholdDb) {
+            //limiter segment
+            newFactorDb = ((1 / ratio) - 1) * (envDb - thresholdDb);
+        }
+
+        float newFactor = dBtoLinear(newFactorDb);
+
+        cb.mLimiterParams.newFactor = newFactor;
+
+    } //end Limiter
+    return mBlockSize;
+}
+
+void DPFrequency::processLinkedLimiters(CBufferVector &channelBuffers) {
+
+    const int channelCount = channelBuffers.size();
+    for (auto &groupPair : mLinkedLimiters.mGroupsMap) {
+        float minFactor = 1.0;
+        //estimate minfactor for all linked
+        for(int index : groupPair.second) {
+            if (index >= 0 && index < channelCount) {
+                minFactor = std::min(channelBuffers[index].mLimiterParams.newFactor, minFactor);
+            }
+        }
+        //apply minFactor
+        for(int index : groupPair.second) {
+            if (index >= 0 && index < channelCount) {
+                channelBuffers[index].mLimiterParams.linkFactor = minFactor;
+            }
+        }
+    }
+}
+
+size_t DPFrequency::processLastStages(ChannelBuffer &cb) {
+
+    float outputGainFactor = dBtoLinear(cb.outputGainDb);
+    //== Limiter. last Pass
+    if (cb.mLimiterInUse && cb.mLimiterEnabled) {
+        //compute factor, with post-gain
+        float factor = cb.mLimiterParams.linkFactor * dBtoLinear(cb.mLimiterParams.postGainDb);
+        outputGainFactor *= factor;
+    }
+
+    //apply to all if != 1.0
+    if (!compareEquality(outputGainFactor, 1.0f)) {
+        size_t cSize = cb.complexTemp.size();
+        size_t maxBin = std::min(cSize/2, mHalfFFTSize);
+        for (size_t k = 0; k < maxBin; k++) {
+            cb.complexTemp[k] *= outputGainFactor;
+        }
+    }
+
+    //##ifft directly to output.
+    Eigen::Map<Eigen::VectorXf> eOutput(&cb.output[0], cb.output.size());
+    mFftServer.inv(eOutput, cb.complexTemp);
+    return mBlockSize;
+}
+
+} //namespace dp_fx

diff --git a/media/libeffects/dynamicsproc/dsp/DPFrequency.h b/media/libeffects/dynamicsproc/dsp/DPFrequency.h
new file mode 100644
index 0000000..be8771d
--- /dev/null
+++ b/media/libeffects/dynamicsproc/dsp/DPFrequency.h

@@ -0,0 +1,160 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#ifndef DPFREQUENCY_H_
+#define DPFREQUENCY_H_
+
+#include <Eigen/Dense>
+#include <unsupported/Eigen/FFT>
+
+#include "RDsp.h"
+#include "SHCircularBuffer.h"
+
+#include "DPBase.h"
+
+
+namespace dp_fx {
+
+using FXBuffer = SHCircularBuffer<float>;
+
+class ChannelBuffer {
+public:
+    FXBuffer cBInput;   // Circular Buffer input
+    FXBuffer cBOutput;  // Circular Buffer output
+    FloatVec input;     // time domain temp vector for input
+    FloatVec output;    // time domain temp vector for output
+    FloatVec outTail;   // time domain temp vector for output tail (for overlap-add method)
+
+    Eigen::VectorXcf complexTemp; // complex temp vector for frequency domain operations
+
+    //Current parameters
+    float inputGainDb;
+    float outputGainDb;
+    struct BandParams {
+        bool enabled;
+        float freqCutoffHz;
+        size_t binStart;
+        size_t binStop;
+    };
+    struct EqBandParams : public BandParams {
+        float gainDb;
+    };
+    struct MbcBandParams : public BandParams {
+        float gainPreDb;
+        float gainPostDb;
+        float attackTimeMs;
+        float releaseTimeMs;
+        float ratio;
+        float thresholdDb;
+        float kneeWidthDb;
+        float noiseGateThresholdDb;
+        float expanderRatio;
+
+        //Historic values
+        float previousEnvelope;
+    };
+    struct LimiterParams {
+        int32_t linkGroup;
+        float attackTimeMs;
+        float releaseTimeMs;
+        float ratio;
+        float thresholdDb;
+        float postGainDb;
+
+        //Historic values
+        float previousEnvelope;
+        float newFactor;
+        float linkFactor;
+    };
+
+    bool mPreEqInUse;
+    bool mPreEqEnabled;
+    std::vector<EqBandParams> mPreEqBands;
+
+    bool mMbcInUse;
+    bool mMbcEnabled;
+    std::vector<MbcBandParams> mMbcBands;
+
+    bool mPostEqInUse;
+    bool mPostEqEnabled;
+    std::vector<EqBandParams> mPostEqBands;
+
+    bool mLimiterInUse;
+    bool mLimiterEnabled;
+    LimiterParams mLimiterParams;
+    FloatVec mPreEqFactorVector; // temp pre-computed vector to shape spectrum at preEQ stage
+    FloatVec mPostEqFactorVector; // temp pre-computed vector to shape spectrum at postEQ stage
+
+    void initBuffers(unsigned int blockSize, unsigned int overlapSize, unsigned int halfFftSize,
+            unsigned int samplingRate, DPBase &dpBase);
+    void computeBinStartStop(BandParams &bp, size_t binStart);
+private:
+    unsigned int mSamplingRate;
+    unsigned int mBlockSize;
+
+};
+
+using CBufferVector = std::vector<ChannelBuffer>;
+
+using GroupsMap = std::map<int32_t, IntVec>;
+
+class LinkedLimiters {
+public:
+    void reset();
+    void update(int32_t group, int index);
+    void remove(int index);
+    GroupsMap mGroupsMap;
+};
+
+class DPFrequency : public DPBase {
+public:
+    virtual size_t processSamples(const float *in, float *out, size_t samples);
+    virtual void reset();
+    void configure(size_t blockSize, size_t overlapSize, size_t samplingRate);
+    static size_t getMinBockSize();
+    static size_t getMaxBockSize();
+
+private:
+    void updateParameters(ChannelBuffer &cb, int channelIndex);
+    size_t processMono(ChannelBuffer &cb);
+    size_t processOneVector(FloatVec &output, FloatVec &input, ChannelBuffer &cb);
+
+    size_t processChannelBuffers(CBufferVector &channelBuffers);
+    size_t processFirstStages(ChannelBuffer &cb);
+    size_t processLastStages(ChannelBuffer &cb);
+    void processLinkedLimiters(CBufferVector &channelBuffers);
+
+    size_t mBlockSize;
+    size_t mHalfFFTSize;
+    size_t mOverlapSize;
+    size_t mSamplingRate;
+
+    float mBlocksPerSecond;
+
+    CBufferVector mChannelBuffers;
+
+    LinkedLimiters mLinkedLimiters;
+
+    //dsp
+    FloatVec mVWindow;  //window class.
+    float mWindowRms;
+    Eigen::FFT<float> mFftServer;
+};
+
+} //namespace dp_fx
+
+#endif  // DPFREQUENCY_H_

diff --git a/media/libeffects/dynamicsproc/dsp/RDsp.h b/media/libeffects/dynamicsproc/dsp/RDsp.h
new file mode 100644
index 0000000..cfa1305
--- /dev/null
+++ b/media/libeffects/dynamicsproc/dsp/RDsp.h

@@ -0,0 +1,175 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef RDSP_H
+#define RDSP_H
+
+#include <complex>
+#include <log/log.h>
+#include <vector>
+#include <map>
+using FloatVec = std::vector<float>;
+using IntVec = std::vector<int>;
+using ComplexVec  = std::vector<std::complex<float>>;
+
+// =======
+// Helper Functions
+// =======
+template <class T>
+static T dBtoLinear(T valueDb) {
+    return pow (10, valueDb / 20.0);
+}
+
+template <class T>
+static T linearToDb(T value) {
+    return 20 * log10(value);
+}
+
+// =======
+// DSP window creation
+// =======
+
+#define TWOPI (M_PI * 2)
+
+enum rdsp_window_type {
+    RDSP_WINDOW_RECTANGULAR,
+    RDSP_WINDOW_TRIANGULAR,
+    RDSP_WINDOW_TRIANGULAR_FLAT_TOP,
+    RDSP_WINDOW_HAMMING,
+    RDSP_WINDOW_HAMMING_FLAT_TOP,
+    RDSP_WINDOW_HANNING,
+    RDSP_WINDOW_HANNING_FLAT_TOP,
+};
+
+template <typename T>
+static void fillRectangular(T &v) {
+    const size_t size = v.size();
+    for (size_t i = 0; i < size; i++) {
+        v[i] = 1.0;
+    }
+} //rectangular
+
+template <typename T>
+static void fillTriangular(T &v, size_t overlap) {
+    const size_t size = v.size();
+    //ramp up
+    size_t i = 0;
+    if (overlap > 0) {
+        for (; i < overlap; i++) {
+            v[i] = (2.0 * i + 1) / (2 * overlap);
+        }
+    }
+
+    //flat top
+    for (; i < size - overlap; i++) {
+        v[i] = 1.0;
+    }
+
+    //ramp down
+    if (overlap > 0) {
+        for (; i < size; i++) {
+            v[i] = (2.0 * (size - i) - 1) / (2 * overlap);
+        }
+    }
+} //triangular
+
+template <typename T>
+static void fillHamming(T &v, size_t overlap) {
+    const size_t size = v.size();
+    const size_t twoOverlap = 2 * overlap;
+    size_t i = 0;
+    if (overlap > 0) {
+        for (; i < overlap; i++) {
+            v[i] = 0.54 - 0.46 * cos(TWOPI * i /(twoOverlap - 1));
+        }
+    }
+
+    //flat top
+    for (; i < size - overlap; i++) {
+        v[i] = 1.0;
+    }
+
+    //ramp down
+    if (overlap > 0) {
+        for (; i < size; i++) {
+            int k = i - ((int)size - 2 * overlap);
+            v[i] = 0.54 - 0.46 * cos(TWOPI * k / (twoOverlap - 1));
+        }
+    }
+} //hamming
+
+template <typename T>
+static void fillHanning(T &v, size_t overlap) {
+    const size_t size = v.size();
+    const size_t twoOverlap = 2 * overlap;
+    //ramp up
+    size_t i = 0;
+    if (overlap > 0) {
+        for (; i < overlap; i++) {
+            v[i] = 0.5 * (1.0 - cos(TWOPI * i / (twoOverlap - 1)));
+        }
+    }
+
+    //flat top
+    for (; i < size - overlap; i++) {
+        v[i] = 1.0;
+    }
+
+    //ramp down
+    if (overlap > 0) {
+        for (; i < size; i++) {
+            int k = i - ((int)size - 2 * overlap);
+            v[i] = 0.5 * (1.0 - cos(TWOPI * k / (twoOverlap - 1)));
+        }
+    }
+}
+
+template <typename T>
+static void fill_window(T &v, int type, size_t size, size_t overlap) {
+    if (overlap > size / 2) {
+        overlap = size / 2;
+    }
+    v.resize(size);
+
+    switch (type) {
+    case RDSP_WINDOW_RECTANGULAR:
+        fillRectangular(v);
+        break;
+    case RDSP_WINDOW_TRIANGULAR:
+        fillTriangular(v, size / 2);
+        break;
+    case RDSP_WINDOW_TRIANGULAR_FLAT_TOP:
+        fillTriangular(v, overlap);
+        break;
+    case RDSP_WINDOW_HAMMING:
+        fillHamming(v, size / 2);
+        break;
+    case RDSP_WINDOW_HAMMING_FLAT_TOP:
+        fillHamming(v, overlap);
+        break;
+    case RDSP_WINDOW_HANNING:
+        fillHanning(v, size / 2);
+        break;
+    case RDSP_WINDOW_HANNING_FLAT_TOP:
+        fillHanning(v, overlap);
+        break;
+    default:
+        ALOGE("Error: unknown window type %d", type);
+    }
+}
+
+//};
+#endif //RDSP_H

diff --git a/media/libeffects/dynamicsproc/dsp/SHCircularBuffer.h b/media/libeffects/dynamicsproc/dsp/SHCircularBuffer.h
new file mode 100644
index 0000000..c139cd8
--- /dev/null
+++ b/media/libeffects/dynamicsproc/dsp/SHCircularBuffer.h

@@ -0,0 +1,81 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SHCIRCULARBUFFER_H
+#define SHCIRCULARBUFFER_H
+
+#include <log/log.h>
+#include <vector>
+
+template <class T>
+class SHCircularBuffer {
+
+public:
+    SHCircularBuffer() : mReadIndex(0), mWriteIndex(0), mReadAvailable(0) {
+    }
+
+    explicit SHCircularBuffer(size_t maxSize) {
+        resize(maxSize);
+    }
+    void resize(size_t maxSize) {
+        mBuffer.resize(maxSize);
+        mReadIndex = 0;
+        mWriteIndex = 0;
+        mReadAvailable = 0;
+    }
+    inline void write(T value) {
+        if (availableToWrite()) {
+            mBuffer[mWriteIndex++] = value;
+            if (mWriteIndex >= getSize()) {
+                mWriteIndex = 0;
+            }
+            mReadAvailable++;
+        } else {
+            ALOGE("Error: SHCircularBuffer no space to write. allocated size %zu ", getSize());
+        }
+    }
+    inline T read() {
+        T value = T();
+        if (availableToRead()) {
+            value = mBuffer[mReadIndex++];
+            if (mReadIndex >= getSize()) {
+                mReadIndex = 0;
+            }
+            mReadAvailable--;
+        } else {
+            ALOGW("Warning: SHCircularBuffer no data available to read. Default value returned");
+        }
+        return value;
+    }
+    inline size_t availableToRead() const {
+        return mReadAvailable;
+    }
+    inline size_t availableToWrite() const {
+        return getSize() - mReadAvailable;
+    }
+    inline size_t getSize() const {
+        return mBuffer.size();
+    }
+
+private:
+    std::vector<T> mBuffer;
+    size_t mReadIndex;
+    size_t mWriteIndex;
+    size_t mReadAvailable;
+};
+
+
+#endif //SHCIRCULARBUFFER_H

diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c
index cd0e765..c1ce513 100644
--- a/media/libeffects/factory/EffectsFactory.c
+++ b/media/libeffects/factory/EffectsFactory.c

@@ -261,7 +261,6 @@
     effect_descriptor_t *d = NULL;
     effect_handle_t itfe;
     effect_entry_t *fx;
-    int found = 0;
     int ret;
 
     if (uuid == NULL || pHandle == NULL) {
@@ -428,8 +427,6 @@
 /////////////////////////////////////////////////
 
 int init() {
-    int hdl;
-
     if (gInitDone) {
         return 0;
     }
@@ -552,7 +549,6 @@
     list_elem_t *e = gLibraryList;
     lib_entry_t *l = NULL;
     effect_descriptor_t *d = NULL;
-    int found = 0;
     int ret = 0;
 
     dprintf(fd, "Libraries loaded:\n");

diff --git a/media/libeffects/factory/EffectsXmlConfigLoader.cpp b/media/libeffects/factory/EffectsXmlConfigLoader.cpp
index 438b787..052a88b 100644
--- a/media/libeffects/factory/EffectsXmlConfigLoader.cpp
+++ b/media/libeffects/factory/EffectsXmlConfigLoader.cpp

@@ -327,7 +327,8 @@
                                            &gSkippedEffects, &gSubEffectList);
 
     ALOGE_IF(result.nbSkippedElement != 0, "%zu errors during loading of configuration: %s",
-             result.nbSkippedElement, path ?: effectsConfig::DEFAULT_PATH);
+             result.nbSkippedElement,
+             result.configPath.empty() ? "No config file found" : result.configPath.c_str());
 
     return result.nbSkippedElement;
 }

diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h b/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h
index f32ed30..4ecaf14 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h

@@ -534,246 +534,246 @@
 
  /* Coefficients for centre frequency 55Hz */
 #define HPF_Fs8000_Fc55_A0                        0.958849f
-#define HPF_Fs8000_Fc55_A1                        -1.917698f
+#define HPF_Fs8000_Fc55_A1                        (-1.917698f)
 #define HPF_Fs8000_Fc55_A2                        0.958849f
-#define HPF_Fs8000_Fc55_B1                        -1.939001f
+#define HPF_Fs8000_Fc55_B1                        (-1.939001f)
 #define HPF_Fs8000_Fc55_B2                        0.940807f
 #define HPF_Fs11025_Fc55_A0                       0.966909f
-#define HPF_Fs11025_Fc55_A1                       -1.933818f
+#define HPF_Fs11025_Fc55_A1                       (-1.933818f)
 #define HPF_Fs11025_Fc55_A2                       0.966909f
-#define HPF_Fs11025_Fc55_B1                       -1.955732f
+#define HPF_Fs11025_Fc55_B1                       (-1.955732f)
 #define HPF_Fs11025_Fc55_B2                       0.956690f
 #define HPF_Fs12000_Fc55_A0                       0.968650f
-#define HPF_Fs12000_Fc55_A1                       -1.937300f
+#define HPF_Fs12000_Fc55_A1                       (-1.937300f)
 #define HPF_Fs12000_Fc55_A2                       0.968650f
-#define HPF_Fs12000_Fc55_B1                       -1.959327f
+#define HPF_Fs12000_Fc55_B1                       (-1.959327f)
 #define HPF_Fs12000_Fc55_B2                       0.960138f
 #define HPF_Fs16000_Fc55_A0                       0.973588f
-#define HPF_Fs16000_Fc55_A1                       -1.947176f
+#define HPF_Fs16000_Fc55_A1                       (-1.947176f)
 #define HPF_Fs16000_Fc55_A2                       0.973588f
-#define HPF_Fs16000_Fc55_B1                       -1.969494f
+#define HPF_Fs16000_Fc55_B1                       (-1.969494f)
 #define HPF_Fs16000_Fc55_B2                       0.969952f
 #define HPF_Fs22050_Fc55_A0                       0.977671f
-#define HPF_Fs22050_Fc55_A1                       -1.955343f
+#define HPF_Fs22050_Fc55_A1                       (-1.955343f)
 #define HPF_Fs22050_Fc55_A2                       0.977671f
-#define HPF_Fs22050_Fc55_B1                       -1.977863f
+#define HPF_Fs22050_Fc55_B1                       (-1.977863f)
 #define HPF_Fs22050_Fc55_B2                       0.978105f
 #define HPF_Fs24000_Fc55_A0                       0.978551f
-#define HPF_Fs24000_Fc55_A1                       -1.957102f
+#define HPF_Fs24000_Fc55_A1                       (-1.957102f)
 #define HPF_Fs24000_Fc55_A2                       0.978551f
-#define HPF_Fs24000_Fc55_B1                       -1.979662f
+#define HPF_Fs24000_Fc55_B1                       (-1.979662f)
 #define HPF_Fs24000_Fc55_B2                       0.979866f
 #define HPF_Fs32000_Fc55_A0                       0.981042f
-#define HPF_Fs32000_Fc55_A1                       -1.962084f
+#define HPF_Fs32000_Fc55_A1                       (-1.962084f)
 #define HPF_Fs32000_Fc55_A2                       0.981042f
-#define HPF_Fs32000_Fc55_B1                       -1.984746f
+#define HPF_Fs32000_Fc55_B1                       (-1.984746f)
 #define HPF_Fs32000_Fc55_B2                       0.984861f
 #define HPF_Fs44100_Fc55_A0                       0.983097f
-#define HPF_Fs44100_Fc55_A1                       -1.966194f
+#define HPF_Fs44100_Fc55_A1                       (-1.966194f)
 #define HPF_Fs44100_Fc55_A2                       0.983097f
-#define HPF_Fs44100_Fc55_B1                       -1.988931f
+#define HPF_Fs44100_Fc55_B1                       (-1.988931f)
 #define HPF_Fs44100_Fc55_B2                       0.988992f
 #define HPF_Fs48000_Fc55_A0                       0.983539f
-#define HPF_Fs48000_Fc55_A1                       -1.967079f
+#define HPF_Fs48000_Fc55_A1                       (-1.967079f)
 #define HPF_Fs48000_Fc55_A2                       0.983539f
-#define HPF_Fs48000_Fc55_B1                       -1.989831f
+#define HPF_Fs48000_Fc55_B1                       (-1.989831f)
 #define HPF_Fs48000_Fc55_B2                       0.989882f
 
 #ifdef HIGHER_FS
 #define HPF_Fs96000_Fc55_A0                       0.986040f
-#define HPF_Fs96000_Fc55_A1                       -1.972080f
+#define HPF_Fs96000_Fc55_A1                       (-1.972080f)
 #define HPF_Fs96000_Fc55_A2                       0.986040f
-#define HPF_Fs96000_Fc55_B1                       -1.994915f
+#define HPF_Fs96000_Fc55_B1                       (-1.994915f)
 #define HPF_Fs96000_Fc55_B2                       0.994928f
 
 #define HPF_Fs192000_Fc55_A0                      0.987294f
-#define HPF_Fs192000_Fc55_A1                      -1.974588f
+#define HPF_Fs192000_Fc55_A1                      (-1.974588f)
 #define HPF_Fs192000_Fc55_A2                      0.987294f
-#define HPF_Fs192000_Fc55_B1                      -1.997458f
+#define HPF_Fs192000_Fc55_B1                      (-1.997458f)
 #define HPF_Fs192000_Fc55_B2                      0.997461f
 #endif
 
 
  /* Coefficients for centre frequency 66Hz */
 #define HPF_Fs8000_Fc66_A0                        0.953016f
-#define HPF_Fs8000_Fc66_A1                        -1.906032f
+#define HPF_Fs8000_Fc66_A1                        (-1.906032f)
 #define HPF_Fs8000_Fc66_A2                        0.953016f
-#define HPF_Fs8000_Fc66_B1                        -1.926810f
+#define HPF_Fs8000_Fc66_B1                        (-1.926810f)
 #define HPF_Fs8000_Fc66_B2                        0.929396f
 #define HPF_Fs11025_Fc66_A0                       0.962638f
-#define HPF_Fs11025_Fc66_A1                       -1.925275f
+#define HPF_Fs11025_Fc66_A1                       (-1.925275f)
 #define HPF_Fs11025_Fc66_A2                       0.962638f
-#define HPF_Fs11025_Fc66_B1                       -1.946881f
+#define HPF_Fs11025_Fc66_B1                       (-1.946881f)
 #define HPF_Fs11025_Fc66_B2                       0.948256f
 #define HPF_Fs12000_Fc66_A0                       0.964718f
-#define HPF_Fs12000_Fc66_A1                       -1.929435f
+#define HPF_Fs12000_Fc66_A1                       (-1.929435f)
 #define HPF_Fs12000_Fc66_A2                       0.964718f
-#define HPF_Fs12000_Fc66_B1                       -1.951196f
+#define HPF_Fs12000_Fc66_B1                       (-1.951196f)
 #define HPF_Fs12000_Fc66_B2                       0.952359f
 #define HPF_Fs16000_Fc66_A0                       0.970622f
-#define HPF_Fs16000_Fc66_A1                       -1.941244f
+#define HPF_Fs16000_Fc66_A1                       (-1.941244f)
 #define HPF_Fs16000_Fc66_A2                       0.970622f
-#define HPF_Fs16000_Fc66_B1                       -1.963394f
+#define HPF_Fs16000_Fc66_B1                       (-1.963394f)
 #define HPF_Fs16000_Fc66_B2                       0.964052f
 #define HPF_Fs22050_Fc66_A0                       0.975509f
-#define HPF_Fs22050_Fc66_A1                       -1.951019f
+#define HPF_Fs22050_Fc66_A1                       (-1.951019f)
 #define HPF_Fs22050_Fc66_A2                       0.975509f
-#define HPF_Fs22050_Fc66_B1                       -1.973436f
+#define HPF_Fs22050_Fc66_B1                       (-1.973436f)
 #define HPF_Fs22050_Fc66_B2                       0.973784f
 #define HPF_Fs24000_Fc66_A0                       0.976563f
-#define HPF_Fs24000_Fc66_A1                       -1.953125f
+#define HPF_Fs24000_Fc66_A1                       (-1.953125f)
 #define HPF_Fs24000_Fc66_A2                       0.976563f
-#define HPF_Fs24000_Fc66_B1                       -1.975594f
+#define HPF_Fs24000_Fc66_B1                       (-1.975594f)
 #define HPF_Fs24000_Fc66_B2                       0.975889f
 #define HPF_Fs32000_Fc66_A0                       0.979547f
-#define HPF_Fs32000_Fc66_A1                       -1.959093f
+#define HPF_Fs32000_Fc66_A1                       (-1.959093f)
 #define HPF_Fs32000_Fc66_A2                       0.979547f
-#define HPF_Fs32000_Fc66_B1                       -1.981695f
+#define HPF_Fs32000_Fc66_B1                       (-1.981695f)
 #define HPF_Fs32000_Fc66_B2                       0.981861f
 #define HPF_Fs44100_Fc66_A0                       0.982010f
-#define HPF_Fs44100_Fc66_A1                       -1.964019f
+#define HPF_Fs44100_Fc66_A1                       (-1.964019f)
 #define HPF_Fs44100_Fc66_A2                       0.982010f
-#define HPF_Fs44100_Fc66_B1                       -1.986718f
+#define HPF_Fs44100_Fc66_B1                       (-1.986718f)
 #define HPF_Fs44100_Fc66_B2                       0.986805f
 #define HPF_Fs48000_Fc66_A0                       0.982540f
-#define HPF_Fs48000_Fc66_A1                       -1.965079f
+#define HPF_Fs48000_Fc66_A1                       (-1.965079f)
 #define HPF_Fs48000_Fc66_A2                       0.982540f
-#define HPF_Fs48000_Fc66_B1                       -1.987797f
+#define HPF_Fs48000_Fc66_B1                       (-1.987797f)
 #define HPF_Fs48000_Fc66_B2                       0.987871f
 
 #ifdef HIGHER_FS
 #define HPF_Fs96000_Fc66_A0                       0.985539f
-#define HPF_Fs96000_Fc66_A1                       -1.971077f
+#define HPF_Fs96000_Fc66_A1                       (-1.971077f)
 #define HPF_Fs96000_Fc66_A2                       0.985539f
-#define HPF_Fs96000_Fc66_B1                       -1.993898f
+#define HPF_Fs96000_Fc66_B1                       (-1.993898f)
 #define HPF_Fs96000_Fc66_B2                       0.993917f
 
 #define HPF_Fs192000_Fc66_A0                      0.987043f
-#define HPF_Fs192000_Fc66_A1                      -1.974086f
+#define HPF_Fs192000_Fc66_A1                      (-1.974086f)
 #define HPF_Fs192000_Fc66_A2                      0.987043f
-#define HPF_Fs192000_Fc66_B1                      -1.996949f
+#define HPF_Fs192000_Fc66_B1                      (-1.996949f)
 #define HPF_Fs192000_Fc66_B2                      0.996954f
 #endif
 
 /* Coefficients for centre frequency 78Hz */
 #define HPF_Fs8000_Fc78_A0                        0.946693f
-#define HPF_Fs8000_Fc78_A1                        -1.893387f
+#define HPF_Fs8000_Fc78_A1                        (-1.893387f)
 #define HPF_Fs8000_Fc78_A2                        0.946693f
-#define HPF_Fs8000_Fc78_B1                        -1.913517f
+#define HPF_Fs8000_Fc78_B1                        (-1.913517f)
 #define HPF_Fs8000_Fc78_B2                        0.917105f
 #define HPF_Fs11025_Fc78_A0                       0.957999f
-#define HPF_Fs11025_Fc78_A1                       -1.915998f
+#define HPF_Fs11025_Fc78_A1                       (-1.915998f)
 #define HPF_Fs11025_Fc78_A2                       0.957999f
-#define HPF_Fs11025_Fc78_B1                       -1.937229f
+#define HPF_Fs11025_Fc78_B1                       (-1.937229f)
 #define HPF_Fs11025_Fc78_B2                       0.939140f
 #define HPF_Fs12000_Fc78_A0                       0.960446f
-#define HPF_Fs12000_Fc78_A1                       -1.920892f
+#define HPF_Fs12000_Fc78_A1                       (-1.920892f)
 #define HPF_Fs12000_Fc78_A2                       0.960446f
-#define HPF_Fs12000_Fc78_B1                       -1.942326f
+#define HPF_Fs12000_Fc78_B1                       (-1.942326f)
 #define HPF_Fs12000_Fc78_B2                       0.943944f
 #define HPF_Fs16000_Fc78_A0                       0.967397f
-#define HPF_Fs16000_Fc78_A1                       -1.934794f
+#define HPF_Fs16000_Fc78_A1                       (-1.934794f)
 #define HPF_Fs16000_Fc78_A2                       0.967397f
-#define HPF_Fs16000_Fc78_B1                       -1.956740f
+#define HPF_Fs16000_Fc78_B1                       (-1.956740f)
 #define HPF_Fs16000_Fc78_B2                       0.957656f
 #define HPF_Fs22050_Fc78_A0                       0.973156f
-#define HPF_Fs22050_Fc78_A1                       -1.946313f
+#define HPF_Fs22050_Fc78_A1                       (-1.946313f)
 #define HPF_Fs22050_Fc78_A2                       0.973156f
-#define HPF_Fs22050_Fc78_B1                       -1.968607f
+#define HPF_Fs22050_Fc78_B1                       (-1.968607f)
 #define HPF_Fs22050_Fc78_B2                       0.969092f
 #define HPF_Fs24000_Fc78_A0                       0.974398f
-#define HPF_Fs24000_Fc78_A1                       -1.948797f
+#define HPF_Fs24000_Fc78_A1                       (-1.948797f)
 #define HPF_Fs24000_Fc78_A2                       0.974398f
-#define HPF_Fs24000_Fc78_B1                       -1.971157f
+#define HPF_Fs24000_Fc78_B1                       (-1.971157f)
 #define HPF_Fs24000_Fc78_B2                       0.971568f
 #define HPF_Fs32000_Fc78_A0                       0.977918f
-#define HPF_Fs32000_Fc78_A1                       -1.955836f
+#define HPF_Fs32000_Fc78_A1                       (-1.955836f)
 #define HPF_Fs32000_Fc78_A2                       0.977918f
-#define HPF_Fs32000_Fc78_B1                       -1.978367f
+#define HPF_Fs32000_Fc78_B1                       (-1.978367f)
 #define HPF_Fs32000_Fc78_B2                       0.978599f
 #define HPF_Fs44100_Fc78_A0                       0.980824f
-#define HPF_Fs44100_Fc78_A1                       -1.961649f
+#define HPF_Fs44100_Fc78_A1                       (-1.961649f)
 #define HPF_Fs44100_Fc78_A2                       0.980824f
-#define HPF_Fs44100_Fc78_B1                       -1.984303f
+#define HPF_Fs44100_Fc78_B1                       (-1.984303f)
 #define HPF_Fs44100_Fc78_B2                       0.984425f
 #define HPF_Fs48000_Fc78_A0                       0.981450f
-#define HPF_Fs48000_Fc78_A1                       -1.962900f
+#define HPF_Fs48000_Fc78_A1                       (-1.962900f)
 #define HPF_Fs48000_Fc78_A2                       0.981450f
-#define HPF_Fs48000_Fc78_B1                       -1.985578f
+#define HPF_Fs48000_Fc78_B1                       (-1.985578f)
 #define HPF_Fs48000_Fc78_B2                       0.985681f
 
 #ifdef HIGHER_FS
 #define HPF_Fs96000_Fc78_A0                       0.984992f
-#define HPF_Fs96000_Fc78_A1                       -1.969984f
+#define HPF_Fs96000_Fc78_A1                       (-1.969984f)
 #define HPF_Fs96000_Fc78_A2                       0.984992f
-#define HPF_Fs96000_Fc78_B1                       -1.992789f
+#define HPF_Fs96000_Fc78_B1                       (-1.992789f)
 #define HPF_Fs96000_Fc78_B2                       0.992815f
 
 #define HPF_Fs192000_Fc78_A0                      0.986769f
-#define HPF_Fs192000_Fc78_A1                      -1.973539f
+#define HPF_Fs192000_Fc78_A1                      (-1.973539f)
 #define HPF_Fs192000_Fc78_A2                      0.986769f
-#define HPF_Fs192000_Fc78_B1                      -1.996394f
+#define HPF_Fs192000_Fc78_B1                      (-1.996394f)
 #define HPF_Fs192000_Fc78_B2                      0.996401f
 #endif
 
 /* Coefficients for centre frequency 90Hz */
 #define HPF_Fs8000_Fc90_A0                       0.940412f
-#define HPF_Fs8000_Fc90_A1                       -1.880825f
+#define HPF_Fs8000_Fc90_A1                       (-1.880825f)
 #define HPF_Fs8000_Fc90_A2                       0.940412f
-#define HPF_Fs8000_Fc90_B1                       -1.900231f
+#define HPF_Fs8000_Fc90_B1                       (-1.900231f)
 #define HPF_Fs8000_Fc90_B2                       0.904977f
 #define HPF_Fs11025_Fc90_A0                      0.953383f
-#define HPF_Fs11025_Fc90_A1                      -1.906766f
+#define HPF_Fs11025_Fc90_A1                      (-1.906766f)
 #define HPF_Fs11025_Fc90_A2                      0.953383f
-#define HPF_Fs11025_Fc90_B1                      -1.927579f
+#define HPF_Fs11025_Fc90_B1                      (-1.927579f)
 #define HPF_Fs11025_Fc90_B2                      0.930111f
 #define HPF_Fs12000_Fc90_A0                      0.956193f
-#define HPF_Fs12000_Fc90_A1                      -1.912387f
+#define HPF_Fs12000_Fc90_A1                      (-1.912387f)
 #define HPF_Fs12000_Fc90_A2                      0.956193f
-#define HPF_Fs12000_Fc90_B1                      -1.933459f
+#define HPF_Fs12000_Fc90_B1                      (-1.933459f)
 #define HPF_Fs12000_Fc90_B2                      0.935603f
 #define HPF_Fs16000_Fc90_A0                      0.964183f
-#define HPF_Fs16000_Fc90_A1                      -1.928365f
+#define HPF_Fs16000_Fc90_A1                      (-1.928365f)
 #define HPF_Fs16000_Fc90_A2                      0.964183f
-#define HPF_Fs16000_Fc90_B1                      -1.950087f
+#define HPF_Fs16000_Fc90_B1                      (-1.950087f)
 #define HPF_Fs16000_Fc90_B2                      0.951303f
 #define HPF_Fs22050_Fc90_A0                      0.970809f
-#define HPF_Fs22050_Fc90_A1                      -1.941618f
+#define HPF_Fs22050_Fc90_A1                      (-1.941618f)
 #define HPF_Fs22050_Fc90_A2                      0.970809f
-#define HPF_Fs22050_Fc90_B1                      -1.963778f
+#define HPF_Fs22050_Fc90_B1                      (-1.963778f)
 #define HPF_Fs22050_Fc90_B2                      0.964423f
 #define HPF_Fs24000_Fc90_A0                      0.972239f
-#define HPF_Fs24000_Fc90_A1                      -1.944477f
+#define HPF_Fs24000_Fc90_A1                      (-1.944477f)
 #define HPF_Fs24000_Fc90_A2                      0.972239f
-#define HPF_Fs24000_Fc90_B1                      -1.966721f
+#define HPF_Fs24000_Fc90_B1                      (-1.966721f)
 #define HPF_Fs24000_Fc90_B2                      0.967266f
 #define HPF_Fs32000_Fc90_A0                      0.976292f
-#define HPF_Fs32000_Fc90_A1                      -1.952584f
+#define HPF_Fs32000_Fc90_A1                      (-1.952584f)
 #define HPF_Fs32000_Fc90_A2                      0.976292f
-#define HPF_Fs32000_Fc90_B1                      -1.975040f
+#define HPF_Fs32000_Fc90_B1                      (-1.975040f)
 #define HPF_Fs32000_Fc90_B2                      0.975347f
 #define HPF_Fs44100_Fc90_A0                      0.979641f
-#define HPF_Fs44100_Fc90_A1                      -1.959282f
+#define HPF_Fs44100_Fc90_A1                      (-1.959282f)
 #define HPF_Fs44100_Fc90_A2                      0.979641f
-#define HPF_Fs44100_Fc90_B1                      -1.981888f
+#define HPF_Fs44100_Fc90_B1                      (-1.981888f)
 #define HPF_Fs44100_Fc90_B2                      0.982050f
 #define HPF_Fs48000_Fc90_A0                      0.980362f
-#define HPF_Fs48000_Fc90_A1                      -1.960724f
+#define HPF_Fs48000_Fc90_A1                      (-1.960724f)
 #define HPF_Fs48000_Fc90_A2                      0.980362f
-#define HPF_Fs48000_Fc90_B1                      -1.983359f
+#define HPF_Fs48000_Fc90_B1                      (-1.983359f)
 #define HPF_Fs48000_Fc90_B2                      0.983497f
 
 #ifdef HIGHER_FS
 #define HPF_Fs96000_Fc90_A0                       0.984446f
-#define HPF_Fs96000_Fc90_A1                       -1.968892f
+#define HPF_Fs96000_Fc90_A1                       (-1.968892f)
 #define HPF_Fs96000_Fc90_A2                       0.984446f
-#define HPF_Fs96000_Fc90_B1                       -1.991680f
+#define HPF_Fs96000_Fc90_B1                       (-1.991680f)
 #define HPF_Fs96000_Fc90_B2                       0.991714f
 
 #define HPF_Fs192000_Fc90_A0                      0.986496f
-#define HPF_Fs192000_Fc90_A1                      -1.972992f
+#define HPF_Fs192000_Fc90_A1                      (-1.972992f)
 #define HPF_Fs192000_Fc90_A2                      0.986496f
-#define HPF_Fs192000_Fc90_B1                      -1.995840f
+#define HPF_Fs192000_Fc90_B1                      (-1.995840f)
 #define HPF_Fs192000_Fc90_B2                      0.995848f
 #endif
 
@@ -786,244 +786,244 @@
 /* Coefficients for centre frequency 55Hz */
 #define BPF_Fs8000_Fc55_A0                       0.009197f
 #define BPF_Fs8000_Fc55_A1                       0.000000f
-#define BPF_Fs8000_Fc55_A2                       -0.009197f
-#define BPF_Fs8000_Fc55_B1                       -1.979545f
+#define BPF_Fs8000_Fc55_A2                       (-0.009197f)
+#define BPF_Fs8000_Fc55_B1                       (-1.979545f)
 #define BPF_Fs8000_Fc55_B2                       0.981393f
 #define BPF_Fs11025_Fc55_A0                      0.006691f
 #define BPF_Fs11025_Fc55_A1                      0.000000f
-#define BPF_Fs11025_Fc55_A2                      -0.006691f
-#define BPF_Fs11025_Fc55_B1                      -1.985488f
+#define BPF_Fs11025_Fc55_A2                      (-0.006691f)
+#define BPF_Fs11025_Fc55_B1                      (-1.985488f)
 #define BPF_Fs11025_Fc55_B2                      0.986464f
 #define BPF_Fs12000_Fc55_A0                      0.006150f
 #define BPF_Fs12000_Fc55_A1                      0.000000f
-#define BPF_Fs12000_Fc55_A2                      -0.006150f
-#define BPF_Fs12000_Fc55_B1                      -1.986733f
+#define BPF_Fs12000_Fc55_A2                      (-0.006150f)
+#define BPF_Fs12000_Fc55_B1                      (-1.986733f)
 #define BPF_Fs12000_Fc55_B2                      0.987557f
 #define BPF_Fs16000_Fc55_A0                      0.004620f
 #define BPF_Fs16000_Fc55_A1                      0.000000f
-#define BPF_Fs16000_Fc55_A2                      -0.004620f
-#define BPF_Fs16000_Fc55_B1                      -1.990189f
+#define BPF_Fs16000_Fc55_A2                      (-0.004620f)
+#define BPF_Fs16000_Fc55_B1                      (-1.990189f)
 #define BPF_Fs16000_Fc55_B2                      0.990653f
 #define BPF_Fs22050_Fc55_A0                      0.003357f
 #define BPF_Fs22050_Fc55_A1                      0.000000f
-#define BPF_Fs22050_Fc55_A2                      -0.003357f
-#define BPF_Fs22050_Fc55_B1                      -1.992964f
+#define BPF_Fs22050_Fc55_A2                      (-0.003357f)
+#define BPF_Fs22050_Fc55_B1                      (-1.992964f)
 #define BPF_Fs22050_Fc55_B2                      0.993209f
 #define BPF_Fs24000_Fc55_A0                      0.003085f
 #define BPF_Fs24000_Fc55_A1                      0.000000f
-#define BPF_Fs24000_Fc55_A2                      -0.003085f
-#define BPF_Fs24000_Fc55_B1                      -1.993552f
+#define BPF_Fs24000_Fc55_A2                      (-0.003085f)
+#define BPF_Fs24000_Fc55_B1                      (-1.993552f)
 #define BPF_Fs24000_Fc55_B2                      0.993759f
 #define BPF_Fs32000_Fc55_A0                      0.002315f
 #define BPF_Fs32000_Fc55_A1                      0.000000f
-#define BPF_Fs32000_Fc55_A2                      -0.002315f
-#define BPF_Fs32000_Fc55_B1                      -1.995199f
+#define BPF_Fs32000_Fc55_A2                      (-0.002315f)
+#define BPF_Fs32000_Fc55_B1                      (-1.995199f)
 #define BPF_Fs32000_Fc55_B2                      0.995316f
 #define BPF_Fs44100_Fc55_A0                      0.001681f
 #define BPF_Fs44100_Fc55_A1                      0.000000f
-#define BPF_Fs44100_Fc55_A2                      -0.001681f
-#define BPF_Fs44100_Fc55_B1                      -1.996537f
+#define BPF_Fs44100_Fc55_A2                      (-0.001681f)
+#define BPF_Fs44100_Fc55_B1                      (-1.996537f)
 #define BPF_Fs44100_Fc55_B2                      0.996599f
 #define BPF_Fs48000_Fc55_A0                      0.001545f
 #define BPF_Fs48000_Fc55_A1                      0.000000f
-#define BPF_Fs48000_Fc55_A2                      -0.001545f
-#define BPF_Fs48000_Fc55_B1                      -1.996823f
+#define BPF_Fs48000_Fc55_A2                      (-0.001545f)
+#define BPF_Fs48000_Fc55_B1                      (-1.996823f)
 #define BPF_Fs48000_Fc55_B2                      0.996875f
 
 #ifdef HIGHER_FS
 #define BPF_Fs96000_Fc55_A0                      0.000762f
 #define BPF_Fs96000_Fc55_A1                      0.000000f
-#define BPF_Fs96000_Fc55_A2                      -0.000762f
-#define BPF_Fs96000_Fc55_B1                      -1.998461f
+#define BPF_Fs96000_Fc55_A2                      (-0.000762f)
+#define BPF_Fs96000_Fc55_B1                      (-1.998461f)
 #define BPF_Fs96000_Fc55_B2                      0.998477f
 
 #define BPF_Fs192000_Fc55_A0                     0.000381f
 #define BPF_Fs192000_Fc55_A1                     0.000000f
-#define BPF_Fs192000_Fc55_A2                     -0.000381f
-#define BPF_Fs192000_Fc55_B1                     -1.999234f
+#define BPF_Fs192000_Fc55_A2                     (-0.000381f)
+#define BPF_Fs192000_Fc55_B1                     (-1.999234f)
 #define BPF_Fs192000_Fc55_B2                     0.999238f
 #endif
 
 /* Coefficients for centre frequency 66Hz */
 #define BPF_Fs8000_Fc66_A0                      0.012648f
 #define BPF_Fs8000_Fc66_A1                      0.000000f
-#define BPF_Fs8000_Fc66_A2                      -0.012648f
-#define BPF_Fs8000_Fc66_B1                      -1.971760f
+#define BPF_Fs8000_Fc66_A2                      (-0.012648f)
+#define BPF_Fs8000_Fc66_B1                      (-1.971760f)
 #define BPF_Fs8000_Fc66_B2                      0.974412f
 #define BPF_Fs11025_Fc66_A0                     0.009209f
 #define BPF_Fs11025_Fc66_A1                     0.000000f
-#define BPF_Fs11025_Fc66_A2                     -0.009209f
-#define BPF_Fs11025_Fc66_B1                     -1.979966f
+#define BPF_Fs11025_Fc66_A2                     (-0.009209f)
+#define BPF_Fs11025_Fc66_B1                     (-1.979966f)
 #define BPF_Fs11025_Fc66_B2                     0.981368f
 #define BPF_Fs12000_Fc66_A0                     0.008468f
 #define BPF_Fs12000_Fc66_A1                     0.000000f
-#define BPF_Fs12000_Fc66_A2                     -0.008468f
-#define BPF_Fs12000_Fc66_B1                     -1.981685f
+#define BPF_Fs12000_Fc66_A2                     (-0.008468f)
+#define BPF_Fs12000_Fc66_B1                     (-1.981685f)
 #define BPF_Fs12000_Fc66_B2                     0.982869f
 #define BPF_Fs16000_Fc66_A0                     0.006364f
 #define BPF_Fs16000_Fc66_A1                     0.000000f
-#define BPF_Fs16000_Fc66_A2                     -0.006364f
-#define BPF_Fs16000_Fc66_B1                     -1.986457f
+#define BPF_Fs16000_Fc66_A2                     (-0.006364f)
+#define BPF_Fs16000_Fc66_B1                     (-1.986457f)
 #define BPF_Fs16000_Fc66_B2                     0.987124f
 #define BPF_Fs22050_Fc66_A0                     0.004626f
 #define BPF_Fs22050_Fc66_A1                     0.000000f
-#define BPF_Fs22050_Fc66_A2                     -0.004626f
-#define BPF_Fs22050_Fc66_B1                     -1.990288f
+#define BPF_Fs22050_Fc66_A2                     (-0.004626f)
+#define BPF_Fs22050_Fc66_B1                     (-1.990288f)
 #define BPF_Fs22050_Fc66_B2                     0.990641f
 #define BPF_Fs24000_Fc66_A0                     0.004252f
 #define BPF_Fs24000_Fc66_A1                     0.000000f
-#define BPF_Fs24000_Fc66_A2                     -0.004252f
-#define BPF_Fs24000_Fc66_B1                     -1.991100f
+#define BPF_Fs24000_Fc66_A2                     (-0.004252f)
+#define BPF_Fs24000_Fc66_B1                     (-1.991100f)
 #define BPF_Fs24000_Fc66_B2                     0.991398f
 #define BPF_Fs32000_Fc66_A0                     0.003192f
 #define BPF_Fs32000_Fc66_A1                     0.000000f
-#define BPF_Fs32000_Fc66_A2                     -0.003192f
-#define BPF_Fs32000_Fc66_B1                     -1.993374f
+#define BPF_Fs32000_Fc66_A2                     (-0.003192f)
+#define BPF_Fs32000_Fc66_B1                     (-1.993374f)
 #define BPF_Fs32000_Fc66_B2                     0.993541f
 #define BPF_Fs44100_Fc66_A0                     0.002318f
 #define BPF_Fs44100_Fc66_A1                     0.000000f
-#define BPF_Fs44100_Fc66_A2                     -0.002318f
-#define BPF_Fs44100_Fc66_B1                     -1.995221f
+#define BPF_Fs44100_Fc66_A2                     (-0.002318f)
+#define BPF_Fs44100_Fc66_B1                     (-1.995221f)
 #define BPF_Fs44100_Fc66_B2                     0.995309f
 #define BPF_Fs48000_Fc66_A0                     0.002131f
 #define BPF_Fs48000_Fc66_A1                     0.000000f
-#define BPF_Fs48000_Fc66_A2                     -0.002131f
-#define BPF_Fs48000_Fc66_B1                     -1.995615f
+#define BPF_Fs48000_Fc66_A2                     (-0.002131f)
+#define BPF_Fs48000_Fc66_B1                     (-1.995615f)
 #define BPF_Fs48000_Fc66_B2                     0.995690f
 
 #ifdef HIGHER_FS
 #define BPF_Fs96000_Fc66_A0                     0.001055f
 #define BPF_Fs96000_Fc66_A1                     0.000000f
-#define BPF_Fs96000_Fc66_A2                     -0.001055f
-#define BPF_Fs96000_Fc66_B1                     -1.997868f
+#define BPF_Fs96000_Fc66_A2                     (-0.001055f)
+#define BPF_Fs96000_Fc66_B1                     (-1.997868f)
 #define BPF_Fs96000_Fc66_B2                     0.997891f
 
 #define BPF_Fs192000_Fc66_A0                    0.000528f
 #define BPF_Fs192000_Fc66_A1                    0.000000f
-#define BPF_Fs192000_Fc66_A2                   -0.000528f
-#define BPF_Fs192000_Fc66_B1                   -1.998939f
+#define BPF_Fs192000_Fc66_A2                   (-0.000528f)
+#define BPF_Fs192000_Fc66_B1                   (-1.998939f)
 #define BPF_Fs192000_Fc66_B2                    0.998945f
 #endif
 
 /* Coefficients for centre frequency 78Hz */
 #define BPF_Fs8000_Fc78_A0                      0.018572f
 #define BPF_Fs8000_Fc78_A1                      0.000000f
-#define BPF_Fs8000_Fc78_A2                      -0.018572f
-#define BPF_Fs8000_Fc78_B1                      -1.958745f
+#define BPF_Fs8000_Fc78_A2                      (-0.018572f)
+#define BPF_Fs8000_Fc78_B1                      (-1.958745f)
 #define BPF_Fs8000_Fc78_B2                      0.962427f
 #define BPF_Fs11025_Fc78_A0                     0.013545f
 #define BPF_Fs11025_Fc78_A1                     0.000000f
-#define BPF_Fs11025_Fc78_A2                     -0.013545f
-#define BPF_Fs11025_Fc78_B1                     -1.970647f
+#define BPF_Fs11025_Fc78_A2                     (-0.013545f)
+#define BPF_Fs11025_Fc78_B1                     (-1.970647f)
 #define BPF_Fs11025_Fc78_B2                     0.972596f
 #define BPF_Fs12000_Fc78_A0                     0.012458f
 #define BPF_Fs12000_Fc78_A1                     0.000000f
-#define BPF_Fs12000_Fc78_A2                     -0.012458f
-#define BPF_Fs12000_Fc78_B1                     -1.973148f
+#define BPF_Fs12000_Fc78_A2                     (-0.012458f)
+#define BPF_Fs12000_Fc78_B1                     (-1.973148f)
 #define BPF_Fs12000_Fc78_B2                     0.974795f
 #define BPF_Fs16000_Fc78_A0                     0.009373f
 #define BPF_Fs16000_Fc78_A1                     0.000000f
-#define BPF_Fs16000_Fc78_A2                     -0.009373f
-#define BPF_Fs16000_Fc78_B1                     -1.980108f
+#define BPF_Fs16000_Fc78_A2                     (-0.009373f)
+#define BPF_Fs16000_Fc78_B1                     (-1.980108f)
 #define BPF_Fs16000_Fc78_B2                     0.981037f
 #define BPF_Fs22050_Fc78_A0                     0.006819f
 #define BPF_Fs22050_Fc78_A1                     0.000000f
-#define BPF_Fs22050_Fc78_A2                     -0.006819f
-#define BPF_Fs22050_Fc78_B1                     -1.985714f
+#define BPF_Fs22050_Fc78_A2                     (-0.006819f)
+#define BPF_Fs22050_Fc78_B1                     (-1.985714f)
 #define BPF_Fs22050_Fc78_B2                     0.986204f
 #define BPF_Fs24000_Fc78_A0                     0.006268f
 #define BPF_Fs24000_Fc78_A1                     0.000000f
-#define BPF_Fs24000_Fc78_A2                     -0.006268f
-#define BPF_Fs24000_Fc78_B1                     -1.986904f
+#define BPF_Fs24000_Fc78_A2                     (-0.006268f)
+#define BPF_Fs24000_Fc78_B1                     (-1.986904f)
 #define BPF_Fs24000_Fc78_B2                     0.987318f
 #define BPF_Fs32000_Fc78_A0                     0.004709f
 #define BPF_Fs32000_Fc78_A1                     0.000000f
-#define BPF_Fs32000_Fc78_A2                     -0.004709f
-#define BPF_Fs32000_Fc78_B1                     -1.990240f
+#define BPF_Fs32000_Fc78_A2                     (-0.004709f)
+#define BPF_Fs32000_Fc78_B1                     (-1.990240f)
 #define BPF_Fs32000_Fc78_B2                     0.990473f
 #define BPF_Fs44100_Fc78_A0                     0.003421f
 #define BPF_Fs44100_Fc78_A1                     0.000000f
-#define BPF_Fs44100_Fc78_A2                     -0.003421f
-#define BPF_Fs44100_Fc78_B1                     -1.992955f
+#define BPF_Fs44100_Fc78_A2                     (-0.003421f)
+#define BPF_Fs44100_Fc78_B1                     (-1.992955f)
 #define BPF_Fs44100_Fc78_B2                     0.993078f
 #define BPF_Fs48000_Fc78_A0                     0.003144f
 #define BPF_Fs48000_Fc78_A1                     0.000000f
-#define BPF_Fs48000_Fc78_A2                     -0.003144f
-#define BPF_Fs48000_Fc78_B1                     -1.993535f
+#define BPF_Fs48000_Fc78_A2                     (-0.003144f)
+#define BPF_Fs48000_Fc78_B1                     (-1.993535f)
 #define BPF_Fs48000_Fc78_B2                     0.993639f
 
 #ifdef HIGHER_FS
 #define BPF_Fs96000_Fc78_A0                     0.001555f
 #define BPF_Fs96000_Fc78_A1                     0.000000f
-#define BPF_Fs96000_Fc78_A2                    -0.0015555f
-#define BPF_Fs96000_Fc78_B1                    -1.996860f
+#define BPF_Fs96000_Fc78_A2                    (-0.0015555f)
+#define BPF_Fs96000_Fc78_B1                    (-1.996860f)
 #define BPF_Fs96000_Fc78_B2                     0.996891f
 
 #define BPF_Fs192000_Fc78_A0                    0.000778f
 #define BPF_Fs192000_Fc78_A1                    0.000000f
-#define BPF_Fs192000_Fc78_A2                   -0.000778f
-#define BPF_Fs192000_Fc78_B1                   -1.998437f
+#define BPF_Fs192000_Fc78_A2                   (-0.000778f)
+#define BPF_Fs192000_Fc78_B1                   (-1.998437f)
 #define BPF_Fs192000_Fc78_B2                    0.998444f
 #endif
 
 /* Coefficients for centre frequency 90Hz */
 #define BPF_Fs8000_Fc90_A0                       0.022760f
 #define BPF_Fs8000_Fc90_A1                       0.000000f
-#define BPF_Fs8000_Fc90_A2                       -0.022760f
-#define BPF_Fs8000_Fc90_B1                       -1.949073f
+#define BPF_Fs8000_Fc90_A2                       (-0.022760f)
+#define BPF_Fs8000_Fc90_B1                       (-1.949073f)
 #define BPF_Fs8000_Fc90_B2                       0.953953f
 #define BPF_Fs11025_Fc90_A0                      0.016619f
 #define BPF_Fs11025_Fc90_A1                      0.000000f
-#define BPF_Fs11025_Fc90_A2                      -0.016619f
-#define BPF_Fs11025_Fc90_B1                      -1.963791f
+#define BPF_Fs11025_Fc90_A2                      (-0.016619f)
+#define BPF_Fs11025_Fc90_B1                      (-1.963791f)
 #define BPF_Fs11025_Fc90_B2                      0.966377f
 #define BPF_Fs12000_Fc90_A0                      0.015289f
 #define BPF_Fs12000_Fc90_A1                      0.000000f
-#define BPF_Fs12000_Fc90_A2                      -0.015289f
-#define BPF_Fs12000_Fc90_B1                      -1.966882f
+#define BPF_Fs12000_Fc90_A2                      (-0.015289f)
+#define BPF_Fs12000_Fc90_B1                      (-1.966882f)
 #define BPF_Fs12000_Fc90_B2                      0.969067f
 #define BPF_Fs16000_Fc90_A0                      0.011511f
 #define BPF_Fs16000_Fc90_A1                      0.000000f
-#define BPF_Fs16000_Fc90_A2                      -0.011511f
-#define BPF_Fs16000_Fc90_B1                      -1.975477f
+#define BPF_Fs16000_Fc90_A2                      (-0.011511f)
+#define BPF_Fs16000_Fc90_B1                      (-1.975477f)
 #define BPF_Fs16000_Fc90_B2                      0.976711f
 #define BPF_Fs22050_Fc90_A0                      0.008379f
 #define BPF_Fs22050_Fc90_A1                      0.000000f
-#define BPF_Fs22050_Fc90_A2                      -0.008379f
-#define BPF_Fs22050_Fc90_B1                      -1.982395f
+#define BPF_Fs22050_Fc90_A2                      (-0.008379f)
+#define BPF_Fs22050_Fc90_B1                      (-1.982395f)
 #define BPF_Fs22050_Fc90_B2                      0.983047f
 #define BPF_Fs24000_Fc90_A0                      0.007704f
 #define BPF_Fs24000_Fc90_A1                      0.000000f
-#define BPF_Fs24000_Fc90_A2                      -0.007704f
-#define BPF_Fs24000_Fc90_B1                      -1.983863f
+#define BPF_Fs24000_Fc90_A2                      (-0.007704f)
+#define BPF_Fs24000_Fc90_B1                      (-1.983863f)
 #define BPF_Fs24000_Fc90_B2                      0.984414f
 #define BPF_Fs32000_Fc90_A0                      0.005789f
 #define BPF_Fs32000_Fc90_A1                      0.000000f
-#define BPF_Fs32000_Fc90_A2                      -0.005789f
-#define BPF_Fs32000_Fc90_B1                      -1.987977f
+#define BPF_Fs32000_Fc90_A2                      (-0.005789f)
+#define BPF_Fs32000_Fc90_B1                      (-1.987977f)
 #define BPF_Fs32000_Fc90_B2                      0.988288f
 #define BPF_Fs44100_Fc90_A0                      0.004207f
 #define BPF_Fs44100_Fc90_A1                      0.000000f
-#define BPF_Fs44100_Fc90_A2                      -0.004207f
-#define BPF_Fs44100_Fc90_B1                      -1.991324f
+#define BPF_Fs44100_Fc90_A2                      (-0.004207f)
+#define BPF_Fs44100_Fc90_B1                      (-1.991324f)
 #define BPF_Fs44100_Fc90_B2                      0.991488f
 #define BPF_Fs48000_Fc90_A0                      0.003867f
 #define BPF_Fs48000_Fc90_A1                      0.000000f
-#define BPF_Fs48000_Fc90_A2                      -0.003867f
-#define BPF_Fs48000_Fc90_B1                      -1.992038f
+#define BPF_Fs48000_Fc90_A2                      (-0.003867f)
+#define BPF_Fs48000_Fc90_B1                      (-1.992038f)
 #define BPF_Fs48000_Fc90_B2                      0.992177f
 
 #ifdef HIGHER_FS
 #define BPF_Fs96000_Fc90_A0                      0.001913f
 #define BPF_Fs96000_Fc90_A1                      0.000000f
-#define BPF_Fs96000_Fc90_A2                     -0.001913f
-#define BPF_Fs96000_Fc90_B1                     -1.996134f
+#define BPF_Fs96000_Fc90_A2                     (-0.001913f)
+#define BPF_Fs96000_Fc90_B1                     (-1.996134f)
 #define BPF_Fs96000_Fc90_B2                      0.996174f
 
 #define BPF_Fs192000_Fc90_A0                     0.000958f
 #define BPF_Fs192000_Fc90_A1                     0.000000f
-#define BPF_Fs192000_Fc90_A2                    -0.000958f
-#define BPF_Fs192000_Fc90_B1                    -1.998075f
+#define BPF_Fs192000_Fc90_A2                    (-0.000958f)
+#define BPF_Fs192000_Fc90_B1                    (-1.998075f)
 #define BPF_Fs192000_Fc90_B2                     0.998085f
 #endif
 

diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h b/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h
index 353560c..8c04847 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h

@@ -69,55 +69,55 @@
 #define HPF_Fs22050_Gain6_B2                            0.000000
                                                                     /* Gain =  7.000000 dB */
 #define HPF_Fs22050_Gain7_A0                            1.390177
-#define HPF_Fs22050_Gain7_A1                            -0.020144
+#define HPF_Fs22050_Gain7_A1                            (-0.020144)
 #define HPF_Fs22050_Gain7_A2                            0.000000
 #define HPF_Fs22050_Gain7_B1                            0.370033
 #define HPF_Fs22050_Gain7_B2                            0.000000
                                                                     /* Gain =  8.000000 dB */
 #define HPF_Fs22050_Gain8_A0                            1.476219
-#define HPF_Fs22050_Gain8_A1                            -0.106187
+#define HPF_Fs22050_Gain8_A1                            (-0.106187)
 #define HPF_Fs22050_Gain8_A2                            0.000000
 #define HPF_Fs22050_Gain8_B1                            0.370033
 #define HPF_Fs22050_Gain8_B2                            0.000000
                                                                     /* Gain =  9.000000 dB */
 #define HPF_Fs22050_Gain9_A0                            1.572761
-#define HPF_Fs22050_Gain9_A1                            -0.202728
+#define HPF_Fs22050_Gain9_A1                            (-0.202728)
 #define HPF_Fs22050_Gain9_A2                            0.000000
 #define HPF_Fs22050_Gain9_B1                            0.370033
 #define HPF_Fs22050_Gain9_B2                            0.000000
                                                                     /* Gain =  10.000000 dB */
 #define HPF_Fs22050_Gain10_A0                           1.681082
-#define HPF_Fs22050_Gain10_A1                           -0.311049
+#define HPF_Fs22050_Gain10_A1                           (-0.311049)
 #define HPF_Fs22050_Gain10_A2                           0.000000
 #define HPF_Fs22050_Gain10_B1                           0.370033
 #define HPF_Fs22050_Gain10_B2                           0.000000
                                                                     /* Gain =  11.000000 dB */
 #define HPF_Fs22050_Gain11_A0                           1.802620
-#define HPF_Fs22050_Gain11_A1                           -0.432588
+#define HPF_Fs22050_Gain11_A1                           (-0.432588)
 #define HPF_Fs22050_Gain11_A2                           0.000000
 #define HPF_Fs22050_Gain11_B1                           0.370033
 #define HPF_Fs22050_Gain11_B2                           0.000000
                                                                     /* Gain =  12.000000 dB */
 #define HPF_Fs22050_Gain12_A0                           1.938989
-#define HPF_Fs22050_Gain12_A1                           -0.568956
+#define HPF_Fs22050_Gain12_A1                           (-0.568956)
 #define HPF_Fs22050_Gain12_A2                           0.000000
 #define HPF_Fs22050_Gain12_B1                           0.370033
 #define HPF_Fs22050_Gain12_B2                           0.000000
                                                                     /* Gain =  13.000000 dB */
 #define HPF_Fs22050_Gain13_A0                           2.091997
-#define HPF_Fs22050_Gain13_A1                           -0.721964
+#define HPF_Fs22050_Gain13_A1                           (-0.721964)
 #define HPF_Fs22050_Gain13_A2                           0.000000
 #define HPF_Fs22050_Gain13_B1                           0.370033
 #define HPF_Fs22050_Gain13_B2                           0.000000
                                                                     /* Gain =  14.000000 dB */
 #define HPF_Fs22050_Gain14_A0                           2.263674
-#define HPF_Fs22050_Gain14_A1                           -0.893641
+#define HPF_Fs22050_Gain14_A1                           (-0.893641)
 #define HPF_Fs22050_Gain14_A2                           0.000000
 #define HPF_Fs22050_Gain14_B1                           0.370033
 #define HPF_Fs22050_Gain14_B2                           0.000000
                                                                     /* Gain =  15.000000 dB */
 #define HPF_Fs22050_Gain15_A0                           2.456300
-#define HPF_Fs22050_Gain15_A1                           -1.086267
+#define HPF_Fs22050_Gain15_A1                           (-1.086267)
 #define HPF_Fs22050_Gain15_A2                           0.000000
 #define HPF_Fs22050_Gain15_B1                           0.370033
 #define HPF_Fs22050_Gain15_B2                           0.000000
@@ -148,342 +148,342 @@
 #define HPF_Fs24000_Gain4_B2                            0.000000
                                                                     /* Gain =  5.000000 dB */
 #define HPF_Fs24000_Gain5_A0                            1.284870
-#define HPF_Fs24000_Gain5_A1                            -0.016921
+#define HPF_Fs24000_Gain5_A1                            (-0.016921)
 #define HPF_Fs24000_Gain5_A2                            0.000000
 #define HPF_Fs24000_Gain5_B1                            0.267949
 #define HPF_Fs24000_Gain5_B2                            0.000000
                                                                     /* Gain =  6.000000 dB */
 #define HPF_Fs24000_Gain6_A0                           1.364291
-#define HPF_Fs24000_Gain6_A1                           -0.096342
+#define HPF_Fs24000_Gain6_A1                           (-0.096342)
 #define HPF_Fs24000_Gain6_A2                           0.000000
 #define HPF_Fs24000_Gain6_B1                           0.267949
 #define HPF_Fs24000_Gain6_B2                           0.000000
                                                                     /* Gain =  7.000000 dB */
 #define HPF_Fs24000_Gain7_A0                            1.453403
-#define HPF_Fs24000_Gain7_A1                            -0.185454
+#define HPF_Fs24000_Gain7_A1                            (-0.185454)
 #define HPF_Fs24000_Gain7_A2                            0.000000
 #define HPF_Fs24000_Gain7_B1                            0.267949
 #define HPF_Fs24000_Gain7_B2                            0.000000
                                                                     /* Gain =  8.000000 dB */
 #define HPF_Fs24000_Gain8_A0                            1.553389
-#define HPF_Fs24000_Gain8_A1                            -0.285440
+#define HPF_Fs24000_Gain8_A1                            (-0.285440)
 #define HPF_Fs24000_Gain8_A2                            0.000000
 #define HPF_Fs24000_Gain8_B1                            0.267949
 #define HPF_Fs24000_Gain8_B2                            0.000000
                                                                     /* Gain =  9.000000 dB */
 #define HPF_Fs24000_Gain9_A0                            1.665574
-#define HPF_Fs24000_Gain9_A1                            -0.397625
+#define HPF_Fs24000_Gain9_A1                            (-0.397625)
 #define HPF_Fs24000_Gain9_A2                            0.000000
 #define HPF_Fs24000_Gain9_B1                            0.267949
 #define HPF_Fs24000_Gain9_B2                            0.000000
                                                                     /* Gain =  10.000000 dB */
 #define HPF_Fs24000_Gain10_A0                           1.791449
-#define HPF_Fs24000_Gain10_A1                           -0.523499
+#define HPF_Fs24000_Gain10_A1                           (-0.523499)
 #define HPF_Fs24000_Gain10_A2                           0.000000
 #define HPF_Fs24000_Gain10_B1                           0.267949
 #define HPF_Fs24000_Gain10_B2                           0.000000
                                                                     /* Gain =  11.000000 dB */
 #define HPF_Fs24000_Gain11_A0                           1.932682
-#define HPF_Fs24000_Gain11_A1                           -0.664733
+#define HPF_Fs24000_Gain11_A1                           (-0.664733)
 #define HPF_Fs24000_Gain11_A2                           0.000000
 #define HPF_Fs24000_Gain11_B1                           0.267949
 #define HPF_Fs24000_Gain11_B2                           0.000000
                                                                     /* Gain =  12.000000 dB */
 #define HPF_Fs24000_Gain12_A0                           2.091148
-#define HPF_Fs24000_Gain12_A1                           -0.823199
+#define HPF_Fs24000_Gain12_A1                           (-0.823199)
 #define HPF_Fs24000_Gain12_A2                           0.000000
 #define HPF_Fs24000_Gain12_B1                           0.267949
 #define HPF_Fs24000_Gain12_B2                           0.000000
                                                                     /* Gain =  13.000000 dB */
 #define HPF_Fs24000_Gain13_A0                           2.268950
-#define HPF_Fs24000_Gain13_A1                           -1.001001
+#define HPF_Fs24000_Gain13_A1                           (-1.001001)
 #define HPF_Fs24000_Gain13_A2                           0.000000
 #define HPF_Fs24000_Gain13_B1                           0.267949
 #define HPF_Fs24000_Gain13_B2                           0.000000
                                                                     /* Gain =  14.000000 dB */
 #define HPF_Fs24000_Gain14_A0                           2.468447
-#define HPF_Fs24000_Gain14_A1                           -1.200498
+#define HPF_Fs24000_Gain14_A1                           (-1.200498)
 #define HPF_Fs24000_Gain14_A2                           0.000000
 #define HPF_Fs24000_Gain14_B1                           0.267949
 #define HPF_Fs24000_Gain14_B2                           0.000000
                                                                     /* Gain =  15.000000 dB */
 #define HPF_Fs24000_Gain15_A0                           2.692287
-#define HPF_Fs24000_Gain15_A1                           -1.424338
+#define HPF_Fs24000_Gain15_A1                           (-1.424338)
 #define HPF_Fs24000_Gain15_A2                           0.000000
 #define HPF_Fs24000_Gain15_B1                           0.267949
 #define HPF_Fs24000_Gain15_B2                           0.000000
 /* Coefficients for sample rate 32000Hz */
                                                                     /* Gain =  1.000000 dB */
 #define HPF_Fs32000_Gain1_A0                            1.061009
-#define HPF_Fs32000_Gain1_A1                            -0.061009
+#define HPF_Fs32000_Gain1_A1                            (-0.061009)
 #define HPF_Fs32000_Gain1_A2                            0.000000
-#define HPF_Fs32000_Gain1_B1                            -0.000000
+#define HPF_Fs32000_Gain1_B1                            (-0.000000)
 #define HPF_Fs32000_Gain1_B2                            0.000000
                                                                     /* Gain =  2.000000 dB */
 #define HPF_Fs32000_Gain2_A0                             1.129463
-#define HPF_Fs32000_Gain2_A1                             -0.129463
+#define HPF_Fs32000_Gain2_A1                             (-0.129463)
 #define HPF_Fs32000_Gain2_A2                             0.000000
-#define HPF_Fs32000_Gain2_B1                             -0.000000
+#define HPF_Fs32000_Gain2_B1                             (-0.000000)
 #define HPF_Fs32000_Gain2_B2                             0.000000
                                                                     /* Gain =  3.000000 dB */
 #define HPF_Fs32000_Gain3_A0                             1.206267
-#define HPF_Fs32000_Gain3_A1                             -0.206267
+#define HPF_Fs32000_Gain3_A1                             (-0.206267)
 #define HPF_Fs32000_Gain3_A2                             0.000000
-#define HPF_Fs32000_Gain3_B1                             -0.000000
+#define HPF_Fs32000_Gain3_B1                             (-0.000000)
 #define HPF_Fs32000_Gain3_B2                             0.000000
                                                                     /* Gain =  4.000000 dB */
 #define HPF_Fs32000_Gain4_A0                            1.292447
-#define HPF_Fs32000_Gain4_A1                            -0.292447
+#define HPF_Fs32000_Gain4_A1                            (-0.292447)
 #define HPF_Fs32000_Gain4_A2                            0.000000
-#define HPF_Fs32000_Gain4_B1                            -0.000000
+#define HPF_Fs32000_Gain4_B1                            (-0.000000)
 #define HPF_Fs32000_Gain4_B2                            0.000000
                                                                     /* Gain =  5.000000 dB */
 #define HPF_Fs32000_Gain5_A0                            1.389140
-#define HPF_Fs32000_Gain5_A1                            -0.389140
+#define HPF_Fs32000_Gain5_A1                            (-0.389140)
 #define HPF_Fs32000_Gain5_A2                            0.000000
-#define HPF_Fs32000_Gain5_B1                            -0.000000
+#define HPF_Fs32000_Gain5_B1                            (-0.000000)
 #define HPF_Fs32000_Gain5_B2                            0.000000
                                                                     /* Gain =  6.000000 dB */
 #define HPF_Fs32000_Gain6_A0                             1.497631
-#define HPF_Fs32000_Gain6_A1                             -0.497631
+#define HPF_Fs32000_Gain6_A1                             (-0.497631)
 #define HPF_Fs32000_Gain6_A2                             0.000000
-#define HPF_Fs32000_Gain6_B1                             -0.000000
+#define HPF_Fs32000_Gain6_B1                             (-0.000000)
 #define HPF_Fs32000_Gain6_B2                             0.000000
                                                                     /* Gain =  7.000000 dB */
 #define HPF_Fs32000_Gain7_A0                             1.619361
-#define HPF_Fs32000_Gain7_A1                             -0.619361
+#define HPF_Fs32000_Gain7_A1                             (-0.619361)
 #define HPF_Fs32000_Gain7_A2                             0.000000
-#define HPF_Fs32000_Gain7_B1                             -0.000000
+#define HPF_Fs32000_Gain7_B1                             (-0.000000)
 #define HPF_Fs32000_Gain7_B2                             0.000000
                                                                     /* Gain =  8.000000 dB */
 #define HPF_Fs32000_Gain8_A0                             1.755943
-#define HPF_Fs32000_Gain8_A1                             -0.755943
+#define HPF_Fs32000_Gain8_A1                             (-0.755943)
 #define HPF_Fs32000_Gain8_A2                             0.000000
-#define HPF_Fs32000_Gain8_B1                             -0.000000
+#define HPF_Fs32000_Gain8_B1                             (-0.000000)
 #define HPF_Fs32000_Gain8_B2                             0.000000
                                                                     /* Gain =  9.000000 dB */
 #define HPF_Fs32000_Gain9_A0                             1.909191
-#define HPF_Fs32000_Gain9_A1                             -0.909191
+#define HPF_Fs32000_Gain9_A1                             (-0.909191)
 #define HPF_Fs32000_Gain9_A2                             0.000000
-#define HPF_Fs32000_Gain9_B1                             -0.000000
+#define HPF_Fs32000_Gain9_B1                             (-0.000000)
 #define HPF_Fs32000_Gain9_B2                             0.000000
                                                                     /* Gain =  10.000000 dB */
 #define HPF_Fs32000_Gain10_A0                            2.081139
-#define HPF_Fs32000_Gain10_A1                            -1.081139
+#define HPF_Fs32000_Gain10_A1                            (-1.081139)
 #define HPF_Fs32000_Gain10_A2                            0.000000
-#define HPF_Fs32000_Gain10_B1                            -0.000000
+#define HPF_Fs32000_Gain10_B1                            (-0.000000)
 #define HPF_Fs32000_Gain10_B2                            0.000000
                                                                     /* Gain =  11.000000 dB */
 #define HPF_Fs32000_Gain11_A0                           2.274067
-#define HPF_Fs32000_Gain11_A1                           -1.274067
+#define HPF_Fs32000_Gain11_A1                           (-1.274067)
 #define HPF_Fs32000_Gain11_A2                           0.000000
-#define HPF_Fs32000_Gain11_B1                           -0.000000
+#define HPF_Fs32000_Gain11_B1                           (-0.000000)
 #define HPF_Fs32000_Gain11_B2                           0.000000
                                                                     /* Gain =  12.000000 dB */
 #define HPF_Fs32000_Gain12_A0                          2.490536
-#define HPF_Fs32000_Gain12_A1                          -1.490536
+#define HPF_Fs32000_Gain12_A1                          (-1.490536)
 #define HPF_Fs32000_Gain12_A2                          0.000000
-#define HPF_Fs32000_Gain12_B1                          -0.000000
+#define HPF_Fs32000_Gain12_B1                          (-0.000000)
 #define HPF_Fs32000_Gain12_B2                          0.000000
                                                                     /* Gain =  13.000000 dB */
 #define HPF_Fs32000_Gain13_A0                           2.733418
-#define HPF_Fs32000_Gain13_A1                           -1.733418
+#define HPF_Fs32000_Gain13_A1                           (-1.733418)
 #define HPF_Fs32000_Gain13_A2                           0.000000
-#define HPF_Fs32000_Gain13_B1                           -0.000000
+#define HPF_Fs32000_Gain13_B1                           (-0.000000)
 #define HPF_Fs32000_Gain13_B2                           0.000000
                                                                     /* Gain =  14.000000 dB */
 #define HPF_Fs32000_Gain14_A0                           3.005936
-#define HPF_Fs32000_Gain14_A1                           -2.005936
+#define HPF_Fs32000_Gain14_A1                           (-2.005936)
 #define HPF_Fs32000_Gain14_A2                           0.000000
-#define HPF_Fs32000_Gain14_B1                           -0.000000
+#define HPF_Fs32000_Gain14_B1                           (-0.000000)
 #define HPF_Fs32000_Gain14_B2                           0.000000
                                                                     /* Gain =  15.000000 dB */
 #define HPF_Fs32000_Gain15_A0                          3.311707
-#define HPF_Fs32000_Gain15_A1                          -2.311707
+#define HPF_Fs32000_Gain15_A1                          (-2.311707)
 #define HPF_Fs32000_Gain15_A2                          0.000000
-#define HPF_Fs32000_Gain15_B1                          -0.000000
+#define HPF_Fs32000_Gain15_B1                          (-0.000000)
 #define HPF_Fs32000_Gain15_B2                          0.000000
 /* Coefficients for sample rate 44100Hz */
                                                                     /* Gain =  1.000000 dB */
 #define HPF_Fs44100_Gain1_A0                            1.074364
-#define HPF_Fs44100_Gain1_A1                            -0.293257
+#define HPF_Fs44100_Gain1_A1                            (-0.293257)
 #define HPF_Fs44100_Gain1_A2                            0.000000
-#define HPF_Fs44100_Gain1_B1                            -0.218894
+#define HPF_Fs44100_Gain1_B1                            (-0.218894)
 #define HPF_Fs44100_Gain1_B2                            0.000000
                                                                     /* Gain =  2.000000 dB */
 #define HPF_Fs44100_Gain2_A0                            1.157801
-#define HPF_Fs44100_Gain2_A1                            -0.376695
+#define HPF_Fs44100_Gain2_A1                            (-0.376695)
 #define HPF_Fs44100_Gain2_A2                            0.000000
-#define HPF_Fs44100_Gain2_B1                            -0.218894
+#define HPF_Fs44100_Gain2_B1                            (-0.218894)
 #define HPF_Fs44100_Gain2_B2                            0.000000
                                                                     /* Gain =  3.000000 dB */
 #define HPF_Fs44100_Gain3_A0                           1.251420
-#define HPF_Fs44100_Gain3_A1                           -0.470313
+#define HPF_Fs44100_Gain3_A1                           (-0.470313)
 #define HPF_Fs44100_Gain3_A2                           0.000000
-#define HPF_Fs44100_Gain3_B1                           -0.218894
+#define HPF_Fs44100_Gain3_B1                           (-0.218894)
 #define HPF_Fs44100_Gain3_B2                           0.000000
                                                                     /* Gain =  4.000000 dB */
 #define HPF_Fs44100_Gain4_A0                            1.356461
-#define HPF_Fs44100_Gain4_A1                            -0.575355
+#define HPF_Fs44100_Gain4_A1                            (-0.575355)
 #define HPF_Fs44100_Gain4_A2                            0.000000
-#define HPF_Fs44100_Gain4_B1                            -0.218894
+#define HPF_Fs44100_Gain4_B1                            (-0.218894)
 #define HPF_Fs44100_Gain4_B2                            0.000000
                                                                     /* Gain =  5.000000 dB */
 #define HPF_Fs44100_Gain5_A0                            1.474320
-#define HPF_Fs44100_Gain5_A1                            -0.693213
+#define HPF_Fs44100_Gain5_A1                            (-0.693213)
 #define HPF_Fs44100_Gain5_A2                            0.000000
-#define HPF_Fs44100_Gain5_B1                            -0.218894
+#define HPF_Fs44100_Gain5_B1                            (-0.218894)
 #define HPF_Fs44100_Gain5_B2                            0.000000
                                                                     /* Gain =  6.000000 dB */
 #define HPF_Fs44100_Gain6_A0                           1.606559
-#define HPF_Fs44100_Gain6_A1                           -0.825453
+#define HPF_Fs44100_Gain6_A1                           (-0.825453)
 #define HPF_Fs44100_Gain6_A2                           0.000000
-#define HPF_Fs44100_Gain6_B1                           -0.218894
+#define HPF_Fs44100_Gain6_B1                           (-0.218894)
 #define HPF_Fs44100_Gain6_B2                           0.000000
                                                                     /* Gain =  7.000000 dB */
 #define HPF_Fs44100_Gain7_A0                           1.754935
-#define HPF_Fs44100_Gain7_A1                           -0.973828
+#define HPF_Fs44100_Gain7_A1                           (-0.973828)
 #define HPF_Fs44100_Gain7_A2                           0.000000
-#define HPF_Fs44100_Gain7_B1                           -0.218894
+#define HPF_Fs44100_Gain7_B1                           (-0.218894)
 #define HPF_Fs44100_Gain7_B2                           0.000000
                                                                     /* Gain =  8.000000 dB */
 #define HPF_Fs44100_Gain8_A0                            1.921414
-#define HPF_Fs44100_Gain8_A1                            -1.140308
+#define HPF_Fs44100_Gain8_A1                            (-1.140308)
 #define HPF_Fs44100_Gain8_A2                            0.000000
-#define HPF_Fs44100_Gain8_B1                            -0.218894
+#define HPF_Fs44100_Gain8_B1                            (-0.218894)
 #define HPF_Fs44100_Gain8_B2                            0.000000
                                                                     /* Gain =  9.000000 dB */
 #define HPF_Fs44100_Gain9_A0                            2.108208
-#define HPF_Fs44100_Gain9_A1                            -1.327101
+#define HPF_Fs44100_Gain9_A1                            (-1.327101)
 #define HPF_Fs44100_Gain9_A2                            0.000000
-#define HPF_Fs44100_Gain9_B1                            -0.218894
+#define HPF_Fs44100_Gain9_B1                            (-0.218894)
 #define HPF_Fs44100_Gain9_B2                            0.000000
                                                                     /* Gain =  10.000000 dB */
 #define HPF_Fs44100_Gain10_A0                          2.317793
-#define HPF_Fs44100_Gain10_A1                          -1.536687
+#define HPF_Fs44100_Gain10_A1                          (-1.536687)
 #define HPF_Fs44100_Gain10_A2                          0.000000
-#define HPF_Fs44100_Gain10_B1                          -0.218894
+#define HPF_Fs44100_Gain10_B1                          (-0.218894)
 #define HPF_Fs44100_Gain10_B2                          0.000000
                                                                     /* Gain =  11.000000 dB */
 #define HPF_Fs44100_Gain11_A0                          2.552952
-#define HPF_Fs44100_Gain11_A1                          -1.771846
+#define HPF_Fs44100_Gain11_A1                          (-1.771846)
 #define HPF_Fs44100_Gain11_A2                          0.000000
-#define HPF_Fs44100_Gain11_B1                          -0.218894
+#define HPF_Fs44100_Gain11_B1                          (-0.218894)
 #define HPF_Fs44100_Gain11_B2                          0.000000
                                                                     /* Gain =  12.000000 dB */
 #define HPF_Fs44100_Gain12_A0                          2.816805
-#define HPF_Fs44100_Gain12_A1                          -2.035698
+#define HPF_Fs44100_Gain12_A1                          (-2.035698)
 #define HPF_Fs44100_Gain12_A2                          0.000000
-#define HPF_Fs44100_Gain12_B1                          -0.218894
+#define HPF_Fs44100_Gain12_B1                          (-0.218894)
 #define HPF_Fs44100_Gain12_B2                          0.000000
                                                                     /* Gain =  13.000000 dB */
 #define HPF_Fs44100_Gain13_A0                           3.112852
-#define HPF_Fs44100_Gain13_A1                           -2.331746
+#define HPF_Fs44100_Gain13_A1                           (-2.331746)
 #define HPF_Fs44100_Gain13_A2                           0.000000
-#define HPF_Fs44100_Gain13_B1                           -0.218894
+#define HPF_Fs44100_Gain13_B1                           (-0.218894)
 #define HPF_Fs44100_Gain13_B2                           0.000000
                                                                     /* Gain =  14.000000 dB */
 #define HPF_Fs44100_Gain14_A0                          3.445023
-#define HPF_Fs44100_Gain14_A1                          -2.663916
+#define HPF_Fs44100_Gain14_A1                          (-2.663916)
 #define HPF_Fs44100_Gain14_A2                          0.000000
-#define HPF_Fs44100_Gain14_B1                          -0.218894
+#define HPF_Fs44100_Gain14_B1                          (-0.218894)
 #define HPF_Fs44100_Gain14_B2                          0.000000
                                                                     /* Gain =  15.000000 dB */
 #define HPF_Fs44100_Gain15_A0                          3.817724
-#define HPF_Fs44100_Gain15_A1                          -3.036618
+#define HPF_Fs44100_Gain15_A1                          (-3.036618)
 #define HPF_Fs44100_Gain15_A2                          0.000000
-#define HPF_Fs44100_Gain15_B1                          -0.218894
+#define HPF_Fs44100_Gain15_B1                          (-0.218894)
 #define HPF_Fs44100_Gain15_B2                          0.000000
 /* Coefficients for sample rate 48000Hz */
                                                                     /* Gain =  1.000000 dB */
 #define HPF_Fs48000_Gain1_A0                          1.077357
-#define HPF_Fs48000_Gain1_A1                          -0.345306
+#define HPF_Fs48000_Gain1_A1                          (-0.345306)
 #define HPF_Fs48000_Gain1_A2                          0.000000
-#define HPF_Fs48000_Gain1_B1                          -0.267949
+#define HPF_Fs48000_Gain1_B1                          (-0.267949)
 #define HPF_Fs48000_Gain1_B2                          0.000000
                                                                     /* Gain =  2.000000 dB */
 #define HPF_Fs48000_Gain2_A0                          1.164152
-#define HPF_Fs48000_Gain2_A1                          -0.432101
+#define HPF_Fs48000_Gain2_A1                          (-0.432101)
 #define HPF_Fs48000_Gain2_A2                          0.000000
-#define HPF_Fs48000_Gain2_B1                          -0.267949
+#define HPF_Fs48000_Gain2_B1                          (-0.267949)
 #define HPF_Fs48000_Gain2_B2                          0.000000
                                                                     /* Gain =  3.000000 dB */
 #define HPF_Fs48000_Gain3_A0                          1.261538
-#define HPF_Fs48000_Gain3_A1                          -0.529488
+#define HPF_Fs48000_Gain3_A1                          (-0.529488)
 #define HPF_Fs48000_Gain3_A2                          0.000000
-#define HPF_Fs48000_Gain3_B1                          -0.267949
+#define HPF_Fs48000_Gain3_B1                          (-0.267949)
 #define HPF_Fs48000_Gain3_B2                          0.000000
                                                                     /* Gain =  4.000000 dB */
 #define HPF_Fs48000_Gain4_A0                           1.370807
-#define HPF_Fs48000_Gain4_A1                           -0.638757
+#define HPF_Fs48000_Gain4_A1                           (-0.638757)
 #define HPF_Fs48000_Gain4_A2                           0.000000
-#define HPF_Fs48000_Gain4_B1                           -0.267949
+#define HPF_Fs48000_Gain4_B1                           (-0.267949)
 #define HPF_Fs48000_Gain4_B2                           0.000000
                                                                     /* Gain =  5.000000 dB */
 #define HPF_Fs48000_Gain5_A0                           1.493409
-#define HPF_Fs48000_Gain5_A1                           -0.761359
+#define HPF_Fs48000_Gain5_A1                           (-0.761359)
 #define HPF_Fs48000_Gain5_A2                           0.000000
-#define HPF_Fs48000_Gain5_B1                           -0.267949
+#define HPF_Fs48000_Gain5_B1                           (-0.267949)
 #define HPF_Fs48000_Gain5_B2                           0.000000
                                                                     /* Gain =  6.000000 dB */
 #define HPF_Fs48000_Gain6_A0                            1.630971
-#define HPF_Fs48000_Gain6_A1                            -0.898920
+#define HPF_Fs48000_Gain6_A1                            (-0.898920)
 #define HPF_Fs48000_Gain6_A2                            0.000000
-#define HPF_Fs48000_Gain6_B1                            -0.267949
+#define HPF_Fs48000_Gain6_B1                            (-0.267949)
 #define HPF_Fs48000_Gain6_B2                            0.000000
                                                                     /* Gain =  7.000000 dB */
 #define HPF_Fs48000_Gain7_A0                            1.785318
-#define HPF_Fs48000_Gain7_A1                            -1.053267
+#define HPF_Fs48000_Gain7_A1                            (-1.053267)
 #define HPF_Fs48000_Gain7_A2                            0.000000
-#define HPF_Fs48000_Gain7_B1                            -0.267949
+#define HPF_Fs48000_Gain7_B1                            (-0.267949)
 #define HPF_Fs48000_Gain7_B2                            0.000000
                                                                     /* Gain =  8.000000 dB */
 #define HPF_Fs48000_Gain8_A0                           1.958498
-#define HPF_Fs48000_Gain8_A1                           -1.226447
+#define HPF_Fs48000_Gain8_A1                           (-1.226447)
 #define HPF_Fs48000_Gain8_A2                           0.000000
-#define HPF_Fs48000_Gain8_B1                           -0.267949
+#define HPF_Fs48000_Gain8_B1                           (-0.267949)
 #define HPF_Fs48000_Gain8_B2                           0.000000
                                                                     /* Gain =  9.000000 dB */
 #define HPF_Fs48000_Gain9_A0                          2.152809
-#define HPF_Fs48000_Gain9_A1                          -1.420758
+#define HPF_Fs48000_Gain9_A1                          (-1.420758)
 #define HPF_Fs48000_Gain9_A2                          0.000000
-#define HPF_Fs48000_Gain9_B1                          -0.267949
+#define HPF_Fs48000_Gain9_B1                          (-0.267949)
 #define HPF_Fs48000_Gain9_B2                          0.000000
                                                                     /* Gain =  10.000000 dB */
 #define HPF_Fs48000_Gain10_A0                         2.370829
-#define HPF_Fs48000_Gain10_A1                         -1.638778
+#define HPF_Fs48000_Gain10_A1                         (-1.638778)
 #define HPF_Fs48000_Gain10_A2                         0.000000
-#define HPF_Fs48000_Gain10_B1                         -0.267949
+#define HPF_Fs48000_Gain10_B1                         (-0.267949)
 #define HPF_Fs48000_Gain10_B2                         0.000000
                                                                     /* Gain =  11.000000 dB */
 #define HPF_Fs48000_Gain11_A0                          2.615452
-#define HPF_Fs48000_Gain11_A1                          -1.883401
+#define HPF_Fs48000_Gain11_A1                          (-1.883401)
 #define HPF_Fs48000_Gain11_A2                          0.000000
-#define HPF_Fs48000_Gain11_B1                          -0.267949
+#define HPF_Fs48000_Gain11_B1                          (-0.267949)
 #define HPF_Fs48000_Gain11_B2                          0.000000
                                                                     /* Gain =  12.000000 dB */
 #define HPF_Fs48000_Gain12_A0                          2.889924
-#define HPF_Fs48000_Gain12_A1                          -2.157873
+#define HPF_Fs48000_Gain12_A1                          (-2.157873)
 #define HPF_Fs48000_Gain12_A2                          0.000000
-#define HPF_Fs48000_Gain12_B1                          -0.267949
+#define HPF_Fs48000_Gain12_B1                          (-0.267949)
 #define HPF_Fs48000_Gain12_B2                          0.000000
                                                                     /* Gain =  13.000000 dB */
 #define HPF_Fs48000_Gain13_A0                           3.197886
-#define HPF_Fs48000_Gain13_A1                           -2.465835
+#define HPF_Fs48000_Gain13_A1                           (-2.465835)
 #define HPF_Fs48000_Gain13_A2                           0.000000
-#define HPF_Fs48000_Gain13_B1                           -0.267949
+#define HPF_Fs48000_Gain13_B1                           (-0.267949)
 #define HPF_Fs48000_Gain13_B2                           0.000000
                                                                     /* Gain =  14.000000 dB */
 #define HPF_Fs48000_Gain14_A0                          3.543425
-#define HPF_Fs48000_Gain14_A1                          -2.811374
+#define HPF_Fs48000_Gain14_A1                          (-2.811374)
 #define HPF_Fs48000_Gain14_A2                          0.000000
-#define HPF_Fs48000_Gain14_B1                          -0.267949
+#define HPF_Fs48000_Gain14_B1                          (-0.267949)
 #define HPF_Fs48000_Gain14_B2                          0.000000
                                                                     /* Gain =  15.000000 dB */
 #define HPF_Fs48000_Gain15_A0                         3.931127
-#define HPF_Fs48000_Gain15_A1                         -3.199076
+#define HPF_Fs48000_Gain15_A1                         (-3.199076)
 #define HPF_Fs48000_Gain15_A2                         0.000000
-#define HPF_Fs48000_Gain15_B1                         -0.267949
+#define HPF_Fs48000_Gain15_B1                         (-0.267949)
 #define HPF_Fs48000_Gain15_B2                         0.000000
 
 #ifdef HIGHER_FS
@@ -491,185 +491,185 @@
 /* Coefficients for sample rate 96000Hz */
                                                                  /* Gain =  1.000000 dB */
 #define HPF_Fs96000_Gain1_A0                          1.096233
-#define HPF_Fs96000_Gain1_A1                          -0.673583
+#define HPF_Fs96000_Gain1_A1                          (-0.673583)
 #define HPF_Fs96000_Gain1_A2                          0.000000
-#define HPF_Fs96000_Gain1_B1                          -0.577350
+#define HPF_Fs96000_Gain1_B1                          (-0.577350)
 #define HPF_Fs96000_Gain1_B2                          0.000000
                                                                  /* Gain =  2.000000 dB */
 #define HPF_Fs96000_Gain2_A0                          1.204208
-#define HPF_Fs96000_Gain2_A1                          -0.781558
+#define HPF_Fs96000_Gain2_A1                          (-0.781558)
 #define HPF_Fs96000_Gain2_A2                          0.000000
-#define HPF_Fs96000_Gain2_B1                          -0.577350
+#define HPF_Fs96000_Gain2_B1                          (-0.577350)
 #define HPF_Fs96000_Gain2_B2                          0.000000
                                                                  /* Gain =  3.000000 dB */
 #define HPF_Fs96000_Gain3_A0                          1.325358
-#define HPF_Fs96000_Gain3_A1                          -0.902708
+#define HPF_Fs96000_Gain3_A1                          (-0.902708)
 #define HPF_Fs96000_Gain3_A2                          0.000000
-#define HPF_Fs96000_Gain3_B1                          -0.577350
+#define HPF_Fs96000_Gain3_B1                          (-0.577350)
 #define HPF_Fs96000_Gain3_B2                          0.000000
                                                                  /* Gain =  4.000000 dB */
 #define HPF_Fs96000_Gain4_A0                           1.461291
-#define HPF_Fs96000_Gain4_A1                           -1.038641
+#define HPF_Fs96000_Gain4_A1                           (-1.038641)
 #define HPF_Fs96000_Gain4_A2                           0.000000
-#define HPF_Fs96000_Gain4_B1                           -0.577350
+#define HPF_Fs96000_Gain4_B1                           (-0.577350)
 #define HPF_Fs96000_Gain4_B2                           0.000000
                                                                  /* Gain =  5.000000 dB */
 #define HPF_Fs96000_Gain5_A0                           1.613810
-#define HPF_Fs96000_Gain5_A1                           -1.191160
+#define HPF_Fs96000_Gain5_A1                           (-1.191160)
 #define HPF_Fs96000_Gain5_A2                           0.000000
-#define HPF_Fs96000_Gain5_B1                           -0.577350
+#define HPF_Fs96000_Gain5_B1                           (-0.577350)
 #define HPF_Fs96000_Gain5_B2                           0.000000
                                                                  /* Gain =  6.000000 dB */
 #define HPF_Fs96000_Gain6_A0                            1.784939
-#define HPF_Fs96000_Gain6_A1                            -1.362289
+#define HPF_Fs96000_Gain6_A1                            (-1.362289)
 #define HPF_Fs96000_Gain6_A2                            0.000000
-#define HPF_Fs96000_Gain6_B1                            -0.577350
+#define HPF_Fs96000_Gain6_B1                            (-0.577350)
 #define HPF_Fs96000_Gain6_B2                            0.000000
                                                                 /* Gain =  7.000000 dB */
 #define HPF_Fs96000_Gain7_A0                            1.976949
-#define HPF_Fs96000_Gain7_A1                            -1.554299
+#define HPF_Fs96000_Gain7_A1                            (-1.554299)
 #define HPF_Fs96000_Gain7_A2                            0.000000
-#define HPF_Fs96000_Gain7_B1                            -0.577350
+#define HPF_Fs96000_Gain7_B1                            (-0.577350)
 #define HPF_Fs96000_Gain7_B2                            0.000000
                                                                  /* Gain =  8.000000 dB */
 #define HPF_Fs96000_Gain8_A0                           2.192387
-#define HPF_Fs96000_Gain8_A1                           -1.769738
+#define HPF_Fs96000_Gain8_A1                           (-1.769738)
 #define HPF_Fs96000_Gain8_A2                           0.000000
-#define HPF_Fs96000_Gain8_B1                           -0.577350
+#define HPF_Fs96000_Gain8_B1                           (-0.577350)
 #define HPF_Fs96000_Gain8_B2                           0.000000
                                                                 /* Gain =  9.000000 dB */
 #define HPF_Fs96000_Gain9_A0                          2.434113
-#define HPF_Fs96000_Gain9_A1                          -2.011464
+#define HPF_Fs96000_Gain9_A1                          (-2.011464)
 #define HPF_Fs96000_Gain9_A2                          0.000000
-#define HPF_Fs96000_Gain9_B1                          -0.577350
+#define HPF_Fs96000_Gain9_B1                          (-0.577350)
 #define HPF_Fs96000_Gain9_B2                          0.000000
                                                                /* Gain =  10.000000 dB */
 #define HPF_Fs96000_Gain10_A0                        2.705335
-#define HPF_Fs96000_Gain10_A1                        -2.282685
+#define HPF_Fs96000_Gain10_A1                        (-2.282685)
 #define HPF_Fs96000_Gain10_A2                         0.000000
-#define HPF_Fs96000_Gain10_B1                         -0.577350
+#define HPF_Fs96000_Gain10_B1                         (-0.577350)
 #define HPF_Fs96000_Gain10_B2                         0.000000
                                                               /* Gain =  11.000000 dB */
 #define HPF_Fs96000_Gain11_A0                          3.009650
-#define HPF_Fs96000_Gain11_A1                          -2.587000
+#define HPF_Fs96000_Gain11_A1                          (-2.587000)
 #define HPF_Fs96000_Gain11_A2                          0.000000
-#define HPF_Fs96000_Gain11_B1                          -0.577350
+#define HPF_Fs96000_Gain11_B1                          (-0.577350)
 #define HPF_Fs96000_Gain11_B2                          0.000000
                                                                   /* Gain =  12.000000 dB */
 #define HPF_Fs96000_Gain12_A0                          3.351097
-#define HPF_Fs96000_Gain12_A1                          -2.928447
+#define HPF_Fs96000_Gain12_A1                          (-2.928447)
 #define HPF_Fs96000_Gain12_A2                          0.000000
-#define HPF_Fs96000_Gain12_B1                          -0.577350
+#define HPF_Fs96000_Gain12_B1                          (-0.577350)
 #define HPF_Fs96000_Gain12_B2                          0.000000
                                                                 /* Gain =  13.000000 dB */
 #define HPF_Fs96000_Gain13_A0                           3.734207
-#define HPF_Fs96000_Gain13_A1                           -3.311558
+#define HPF_Fs96000_Gain13_A1                           (-3.311558)
 #define HPF_Fs96000_Gain13_A2                           0.000000
-#define HPF_Fs96000_Gain13_B1                           -0.577350
+#define HPF_Fs96000_Gain13_B1                           (-0.577350)
 #define HPF_Fs96000_Gain13_B2                           0.000000
                                                                  /* Gain =  14.000000 dB */
 #define HPF_Fs96000_Gain14_A0                         4.164064
-#define HPF_Fs96000_Gain14_A1                         -3.741414
+#define HPF_Fs96000_Gain14_A1                         (-3.741414)
 #define HPF_Fs96000_Gain14_A2                          0.000000
-#define HPF_Fs96000_Gain14_B1                          -0.577350
+#define HPF_Fs96000_Gain14_B1                          (-0.577350)
 #define HPF_Fs96000_Gain14_B2                          0.000000
                                                                  /* Gain =  15.000000 dB */
 #define HPF_Fs96000_Gain15_A0                         4.646371
-#define HPF_Fs96000_Gain15_A1                         -4.223721
+#define HPF_Fs96000_Gain15_A1                         (-4.223721)
 #define HPF_Fs96000_Gain15_A2                         0.000000
-#define HPF_Fs96000_Gain15_B1                         -0.577350
+#define HPF_Fs96000_Gain15_B1                         (-0.577350)
 #define HPF_Fs96000_Gain15_B2                         0.000000
 
 /* Coefficients for sample rate 192000Hz */
                                                                   /* Gain =  1.000000 dB */
 #define HPF_Fs192000_Gain1_A0                          1.107823
-#define HPF_Fs192000_Gain1_A1                          -0.875150
+#define HPF_Fs192000_Gain1_A1                          (-0.875150)
 #define HPF_Fs192000_Gain1_A2                          0.000000
-#define HPF_Fs192000_Gain1_B1                          -0.767327
+#define HPF_Fs192000_Gain1_B1                          (-0.767327)
 #define HPF_Fs192000_Gain1_B2                          0.000000
                                                                   /* Gain =  2.000000 dB */
 #define HPF_Fs192000_Gain2_A0                          1.228803
-#define HPF_Fs192000_Gain2_A1                          -0.996130
+#define HPF_Fs192000_Gain2_A1                          (-0.996130)
 #define HPF_Fs192000_Gain2_A2                          0.000000
-#define HPF_Fs192000_Gain2_B1                          -0.767327
+#define HPF_Fs192000_Gain2_B1                          (-0.767327)
 #define HPF_Fs192000_Gain2_B2                          0.000000
                                                                    /* Gain =  3.000000 dB */
 #define HPF_Fs192000_Gain3_A0                          1.364544
-#define HPF_Fs192000_Gain3_A1                          -1.131871
+#define HPF_Fs192000_Gain3_A1                          (-1.131871)
 #define HPF_Fs192000_Gain3_A2                          0.000000
-#define HPF_Fs192000_Gain3_B1                          -0.767327
+#define HPF_Fs192000_Gain3_B1                          (-0.767327)
 #define HPF_Fs192000_Gain3_B2                          0.000000
                                                                    /* Gain =  4.000000 dB */
 #define HPF_Fs192000_Gain4_A0                          1.516849
-#define HPF_Fs192000_Gain4_A1                          -1.284176
+#define HPF_Fs192000_Gain4_A1                          (-1.284176)
 #define HPF_Fs192000_Gain4_A2                           0.000000
-#define HPF_Fs192000_Gain4_B1                           -0.767327
+#define HPF_Fs192000_Gain4_B1                           (-0.767327)
 #define HPF_Fs192000_Gain4_B2                           0.000000
                                                                    /* Gain =  5.000000 dB */
 #define HPF_Fs192000_Gain5_A0                           1.687737
-#define HPF_Fs192000_Gain5_A1                           -1.455064
+#define HPF_Fs192000_Gain5_A1                           (-1.455064)
 #define HPF_Fs192000_Gain5_A2                           0.000000
-#define HPF_Fs192000_Gain5_B1                           -0.767327
+#define HPF_Fs192000_Gain5_B1                           (-0.767327)
 #define HPF_Fs192000_Gain5_B2                           0.000000
                                                                    /* Gain =  6.000000 dB */
 #define HPF_Fs192000_Gain6_A0                            1.879477
-#define HPF_Fs192000_Gain6_A1                            -1.646804
+#define HPF_Fs192000_Gain6_A1                            (-1.646804)
 #define HPF_Fs192000_Gain6_A2                            0.000000
-#define HPF_Fs192000_Gain6_B1                            -0.767327
+#define HPF_Fs192000_Gain6_B1                            (-0.767327)
 #define HPF_Fs192000_Gain6_B2                            0.000000
                                                                  /* Gain =  7.000000 dB */
 #define HPF_Fs192000_Gain7_A0                            2.094613
-#define HPF_Fs192000_Gain7_A1                            -1.861940
+#define HPF_Fs192000_Gain7_A1                            (-1.861940)
 #define HPF_Fs192000_Gain7_A2                            0.000000
-#define HPF_Fs192000_Gain7_B1                            -0.767327
+#define HPF_Fs192000_Gain7_B1                            (-0.767327)
 #define HPF_Fs192000_Gain7_B2                            0.000000
                                                                    /* Gain =  8.000000 dB */
 #define HPF_Fs192000_Gain8_A0                           2.335999
-#define HPF_Fs192000_Gain8_A1                           -2.103326
+#define HPF_Fs192000_Gain8_A1                           (-2.103326)
 #define HPF_Fs192000_Gain8_A2                           0.000000
-#define HPF_Fs192000_Gain8_B1                           -0.767327
+#define HPF_Fs192000_Gain8_B1                           (-0.767327)
 #define HPF_Fs192000_Gain8_B2                           0.000000
                                                                    /* Gain =  9.000000 dB */
 #define HPF_Fs192000_Gain9_A0                          2.606839
-#define HPF_Fs192000_Gain9_A1                          -2.374166
+#define HPF_Fs192000_Gain9_A1                          (-2.374166)
 #define HPF_Fs192000_Gain9_A2                          0.000000
-#define HPF_Fs192000_Gain9_B1                          -0.767327
+#define HPF_Fs192000_Gain9_B1                          (-0.767327)
 #define HPF_Fs192000_Gain9_B2                          0.000000
                                                                  /* Gain =  10.000000 dB */
 #define HPF_Fs192000_Gain10_A0                        2.910726
-#define HPF_Fs192000_Gain10_A1                        -2.678053
+#define HPF_Fs192000_Gain10_A1                        (-2.678053)
 #define HPF_Fs192000_Gain10_A2                         0.000000
-#define HPF_Fs192000_Gain10_B1                         -0.767327
+#define HPF_Fs192000_Gain10_B1                         (-0.767327)
 #define HPF_Fs192000_Gain10_B2                         0.000000
                                                                   /* Gain =  11.000000 dB */
 #define HPF_Fs192000_Gain11_A0                          3.251693
-#define HPF_Fs192000_Gain11_A1                          -3.019020
+#define HPF_Fs192000_Gain11_A1                          (-3.019020)
 #define HPF_Fs192000_Gain11_A2                          0.000000
-#define HPF_Fs192000_Gain11_B1                          -0.767327
+#define HPF_Fs192000_Gain11_B1                          (-0.767327)
 #define HPF_Fs192000_Gain11_B2                          0.000000
                                                                   /* Gain =  12.000000 dB */
 #define HPF_Fs192000_Gain12_A0                          3.634264
-#define HPF_Fs192000_Gain12_A1                          -3.401591
+#define HPF_Fs192000_Gain12_A1                          (-3.401591)
 #define HPF_Fs192000_Gain12_A2                          0.000000
-#define HPF_Fs192000_Gain12_B1                          -0.767327
+#define HPF_Fs192000_Gain12_B1                          (-0.767327)
 #define HPF_Fs192000_Gain12_B2                          0.000000
                                                                 /* Gain =  13.000000 dB */
 #define HPF_Fs192000_Gain13_A0                           4.063516
-#define HPF_Fs192000_Gain13_A1                           -3.830843
+#define HPF_Fs192000_Gain13_A1                           (-3.830843)
 #define HPF_Fs192000_Gain13_A2                           0.000000
-#define HPF_Fs192000_Gain13_B1                           -0.767327
+#define HPF_Fs192000_Gain13_B1                           (-0.767327)
 #define HPF_Fs192000_Gain13_B2                           0.000000
                                                                 /* Gain =  14.000000 dB */
 #define HPF_Fs192000_Gain14_A0                          4.545145
-#define HPF_Fs192000_Gain14_A1                          -4.312472
+#define HPF_Fs192000_Gain14_A1                          (-4.312472)
 #define HPF_Fs192000_Gain14_A2                          0.000000
-#define HPF_Fs192000_Gain14_B1                          -0.767327
+#define HPF_Fs192000_Gain14_B1                          (-0.767327)
 #define HPF_Fs192000_Gain14_B2                          0.000000
                                                                   /* Gain =  15.000000 dB */
 #define HPF_Fs192000_Gain15_A0                         5.085542
-#define HPF_Fs192000_Gain15_A1                         -4.852868
+#define HPF_Fs192000_Gain15_A1                         (-4.852868)
 #define HPF_Fs192000_Gain15_A2                         0.000000
-#define HPF_Fs192000_Gain15_B1                         -0.767327
+#define HPF_Fs192000_Gain15_B1                         (-0.767327)
 #define HPF_Fs192000_Gain15_B2                         0.000000
 
 #endif

diff --git a/media/libeffects/lvm/lib/Common/lib/LVM_Types.h b/media/libeffects/lvm/lib/Common/lib/LVM_Types.h
index cb15b60..ea16072 100644
--- a/media/libeffects/lvm/lib/Common/lib/LVM_Types.h
+++ b/media/libeffects/lvm/lib/Common/lib/LVM_Types.h

@@ -44,9 +44,6 @@
 
 #define LVM_MAXINT_8            127                 /* Maximum positive integer size */
 #define LVM_MAXINT_16           32767
-#ifdef BUILD_FLOAT
-#define LVM_MAXFLOAT            1.0f
-#endif
 #define LVM_MAXINT_32           2147483647
 #define LVM_MAXENUM             2147483647
 
@@ -99,8 +96,32 @@
 typedef     uint32_t            LVM_UINT32;         /* Unsigned 32-bit word */
 
 #ifdef BUILD_FLOAT
-typedef     float               LVM_FLOAT;          /* single precission floating point*/
-#endif
+
+#define LVM_MAXFLOAT            1.f
+
+typedef     float               LVM_FLOAT;          /* single precision floating point */
+
+// If NATIVE_FLOAT_BUFFER is defined, we expose effects as floating point format;
+// otherwise we expose as integer 16 bit and translate to float for the effect libraries.
+// Hence, NATIVE_FLOAT_BUFFER should only be enabled under BUILD_FLOAT compilation.
+
+#define NATIVE_FLOAT_BUFFER
+
+#endif // BUILD_FLOAT
+
+// Select whether we expose int16_t or float buffers.
+#ifdef NATIVE_FLOAT_BUFFER
+
+#define    EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_FLOAT
+typedef     float               effect_buffer_t;
+
+#else // NATIVE_FLOAT_BUFFER
+
+#define    EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_16_BIT
+typedef     int16_t             effect_buffer_t;
+
+#endif // NATIVE_FLOAT_BUFFER
+
 /****************************************************************************************/
 /*                                                                                      */
 /*  Standard Enumerated types                                                           */

diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h
index f0deb6c..42ea46f 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h

@@ -26,21 +26,21 @@
 /*                                                                                  */
 /************************************************************************************/
 #ifdef BUILD_FLOAT
-#define LVEQNB_Gain_Neg15_dB                             -0.822172f
-#define LVEQNB_Gain_Neg14_dB                             -0.800474f
-#define LVEQNB_Gain_Neg13_dB                             -0.776128f
-#define LVEQNB_Gain_Neg12_dB                             -0.748811f
-#define LVEQNB_Gain_Neg11_dB                             -0.718162f
-#define LVEQNB_Gain_Neg10_dB                             -0.683772f
-#define LVEQNB_Gain_Neg9_dB                              -0.645187f
-#define LVEQNB_Gain_Neg8_dB                              -0.601893f
-#define LVEQNB_Gain_Neg7_dB                              -0.553316f
-#define LVEQNB_Gain_Neg6_dB                              -0.498813f
-#define LVEQNB_Gain_Neg5_dB                              -0.437659f
-#define LVEQNB_Gain_Neg4_dB                              -0.369043f
-#define LVEQNB_Gain_Neg3_dB                              -0.292054f
-#define LVEQNB_Gain_Neg2_dB                              -0.205672f
-#define LVEQNB_Gain_Neg1_dB                              -0.108749f
+#define LVEQNB_Gain_Neg15_dB                             (-0.822172f)
+#define LVEQNB_Gain_Neg14_dB                             (-0.800474f)
+#define LVEQNB_Gain_Neg13_dB                             (-0.776128f)
+#define LVEQNB_Gain_Neg12_dB                             (-0.748811f)
+#define LVEQNB_Gain_Neg11_dB                             (-0.718162f)
+#define LVEQNB_Gain_Neg10_dB                             (-0.683772f)
+#define LVEQNB_Gain_Neg9_dB                              (-0.645187f)
+#define LVEQNB_Gain_Neg8_dB                              (-0.601893f)
+#define LVEQNB_Gain_Neg7_dB                              (-0.553316f)
+#define LVEQNB_Gain_Neg6_dB                              (-0.498813f)
+#define LVEQNB_Gain_Neg5_dB                              (-0.437659f)
+#define LVEQNB_Gain_Neg4_dB                              (-0.369043f)
+#define LVEQNB_Gain_Neg3_dB                              (-0.292054f)
+#define LVEQNB_Gain_Neg2_dB                              (-0.205672f)
+#define LVEQNB_Gain_Neg1_dB                              (-0.108749f)
 #define LVEQNB_Gain_0_dB                                  0.000000f
 #define LVEQNB_Gain_1_dB                                  0.122018f
 #define LVEQNB_Gain_2_dB                                  0.258925f

diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.c b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.c
index c290aec..7b0f341 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.c
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.c

@@ -430,7 +430,15 @@
     }
 
 
-    if(bChange){
+    // During operating mode transition, there is a race condition where the mode
+    // is still LVEQNB_ON, but the effect is considered disabled in the upper layers.
+    // modeChange handles this special race condition.
+    const int /* bool */ modeChange = pParams->OperatingMode != OperatingModeSave
+            || (OperatingModeSave == LVEQNB_ON
+                    && pInstance->bInOperatingModeTransition
+                    && LVC_Mixer_GetTarget(&pInstance->BypassMixer.MixerStream[0]) == 0);
+
+    if (bChange || modeChange) {
 
         /*
          * If the sample rate has changed clear the history
@@ -462,8 +470,7 @@
             LVEQNB_SetCoefficients(pInstance);                  /* Instance pointer */
         }
 
-        if(pParams->OperatingMode != OperatingModeSave)
-        {
+        if (modeChange) {
             if(pParams->OperatingMode == LVEQNB_ON)
             {
 #ifdef BUILD_FLOAT
@@ -479,6 +486,8 @@
             else
             {
                 /* Stay on the ON operating mode until the transition is done */
+                // This may introduce a state race condition if the effect is enabled again
+                // while in transition.  This is fixed in the modeChange logic.
                 pInstance->Params.OperatingMode = LVEQNB_ON;
 #ifdef BUILD_FLOAT
                 LVC_Mixer_SetTarget(&pInstance->BypassMixer.MixerStream[0], 0.0f);

diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h
index 4f5221a..0c2fe53 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h

@@ -27,127 +27,127 @@
 #ifdef BUILD_FLOAT
 /* Stereo Enhancer coefficients for 8000 Hz sample rate, scaled with 0.161258 */
 #define CS_MIDDLE_8000_A0                           0.227720
-#define CS_MIDDLE_8000_A1                          -0.215125
+#define CS_MIDDLE_8000_A1                          (-0.215125)
 #define CS_MIDDLE_8000_A2                           0.000000
-#define CS_MIDDLE_8000_B1                          -0.921899
+#define CS_MIDDLE_8000_B1                          (-0.921899)
 #define CS_MIDDLE_8000_B2                           0.000000
 #define CS_MIDDLE_8000_SCALE                        15
 #define CS_SIDE_8000_A0                             0.611441
-#define CS_SIDE_8000_A1                            -0.380344
-#define CS_SIDE_8000_A2                            -0.231097
-#define CS_SIDE_8000_B1                            -0.622470
-#define CS_SIDE_8000_B2                            -0.130759
+#define CS_SIDE_8000_A1                            (-0.380344)
+#define CS_SIDE_8000_A2                            (-0.231097)
+#define CS_SIDE_8000_B1                            (-0.622470)
+#define CS_SIDE_8000_B2                            (-0.130759)
 #define CS_SIDE_8000_SCALE                         15
 
 /* Stereo Enhancer coefficients for 11025Hz sample rate, scaled with 0.162943 */
 #define CS_MIDDLE_11025_A0                       0.230838
-#define CS_MIDDLE_11025_A1                      -0.221559
+#define CS_MIDDLE_11025_A1                      (-0.221559)
 #define CS_MIDDLE_11025_A2                       0.000000
-#define CS_MIDDLE_11025_B1                      -0.943056
+#define CS_MIDDLE_11025_B1                      (-0.943056)
 #define CS_MIDDLE_11025_B2                       0.000000
 #define CS_MIDDLE_11025_SCALE                    15
 #define CS_SIDE_11025_A0                         0.557372
-#define CS_SIDE_11025_A1                        -0.391490
-#define CS_SIDE_11025_A2                        -0.165881
-#define CS_SIDE_11025_B1                        -0.880608
+#define CS_SIDE_11025_A1                        (-0.391490)
+#define CS_SIDE_11025_A2                        (-0.165881)
+#define CS_SIDE_11025_B1                        (-0.880608)
 #define CS_SIDE_11025_B2                         0.032397
 #define CS_SIDE_11025_SCALE                      15
 
 /* Stereo Enhancer coefficients for 12000Hz sample rate, scaled with 0.162191 */
 #define CS_MIDDLE_12000_A0                        0.229932
-#define CS_MIDDLE_12000_A1                       -0.221436
+#define CS_MIDDLE_12000_A1                       (-0.221436)
 #define CS_MIDDLE_12000_A2                        0.000000
-#define CS_MIDDLE_12000_B1                       -0.947616
+#define CS_MIDDLE_12000_B1                       (-0.947616)
 #define CS_MIDDLE_12000_B2                        0.000000
 #define CS_MIDDLE_12000_SCALE                        15
 #define CS_SIDE_12000_A0                         0.558398
-#define CS_SIDE_12000_A1                        -0.392211
-#define CS_SIDE_12000_A2                        -0.166187
-#define CS_SIDE_12000_B1                        -0.892550
+#define CS_SIDE_12000_A1                        (-0.392211)
+#define CS_SIDE_12000_A2                        (-0.166187)
+#define CS_SIDE_12000_B1                        (-0.892550)
 #define CS_SIDE_12000_B2                         0.032856
 #define CS_SIDE_12000_SCALE                          15
 
 /* Stereo Enhancer coefficients for 16000Hz sample rate, scaled with 0.162371 */
 #define CS_MIDDLE_16000_A0                       0.230638
-#define CS_MIDDLE_16000_A1                      -0.224232
+#define CS_MIDDLE_16000_A1                      (-0.224232)
 #define CS_MIDDLE_16000_A2                       0.000000
-#define CS_MIDDLE_16000_B1                      -0.960550
+#define CS_MIDDLE_16000_B1                      (-0.960550)
 #define CS_MIDDLE_16000_B2                       0.000000
 #define CS_MIDDLE_16000_SCALE                        15
 #define CS_SIDE_16000_A0                         0.499695
-#define CS_SIDE_16000_A1                        -0.355543
-#define CS_SIDE_16000_A2                        -0.144152
-#define CS_SIDE_16000_B1                        -1.050788
+#define CS_SIDE_16000_A1                        (-0.355543)
+#define CS_SIDE_16000_A2                        (-0.144152)
+#define CS_SIDE_16000_B1                        (-1.050788)
 #define CS_SIDE_16000_B2                         0.144104
 #define CS_SIDE_16000_SCALE                          14
 
 /* Stereo Enhancer coefficients for 22050Hz sample rate, scaled with 0.160781 */
 #define CS_MIDDLE_22050_A0                       0.228749
-#define CS_MIDDLE_22050_A1                      -0.224128
+#define CS_MIDDLE_22050_A1                      (-0.224128)
 #define CS_MIDDLE_22050_A2                       0.000000
-#define CS_MIDDLE_22050_B1                      -0.971262
+#define CS_MIDDLE_22050_B1                      (-0.971262)
 #define CS_MIDDLE_22050_B2                       0.000000
 #define CS_MIDDLE_22050_SCALE                        15
 #define CS_SIDE_22050_A0                          0.440112
-#define CS_SIDE_22050_A1                         -0.261096
-#define CS_SIDE_22050_A2                         -0.179016
-#define CS_SIDE_22050_B1                         -1.116786
+#define CS_SIDE_22050_A1                         (-0.261096)
+#define CS_SIDE_22050_A2                         (-0.179016)
+#define CS_SIDE_22050_B1                         (-1.116786)
 #define CS_SIDE_22050_B2                          0.182507
 #define CS_SIDE_22050_SCALE                          14
 
 /* Stereo Enhancer coefficients for 24000Hz sample rate, scaled with 0.161882 */
 #define CS_MIDDLE_24000_A0                         0.230395
-#define CS_MIDDLE_24000_A1                        -0.226117
+#define CS_MIDDLE_24000_A1                        (-0.226117)
 #define CS_MIDDLE_24000_A2                         0.000000
-#define CS_MIDDLE_24000_B1                        -0.973573
+#define CS_MIDDLE_24000_B1                        (-0.973573)
 #define CS_MIDDLE_24000_B2                         0.000000
 #define CS_MIDDLE_24000_SCALE                        15
 #define CS_SIDE_24000_A0                           0.414770
-#define CS_SIDE_24000_A1                          -0.287182
-#define CS_SIDE_24000_A2                          -0.127588
-#define CS_SIDE_24000_B1                          -1.229648
+#define CS_SIDE_24000_A1                          (-0.287182)
+#define CS_SIDE_24000_A2                          (-0.127588)
+#define CS_SIDE_24000_B1                          (-1.229648)
 #define CS_SIDE_24000_B2                           0.282177
 #define CS_SIDE_24000_SCALE                          14
 
 /* Stereo Enhancer coefficients for 32000Hz sample rate, scaled with 0.160322 */
 #define CS_MIDDLE_32000_A0                          0.228400
-#define CS_MIDDLE_32000_A1                         -0.225214
+#define CS_MIDDLE_32000_A1                         (-0.225214)
 #define CS_MIDDLE_32000_A2                          0.000000
-#define CS_MIDDLE_32000_B1                         -0.980126
+#define CS_MIDDLE_32000_B1                         (-0.980126)
 #define CS_MIDDLE_32000_B2                          0.000000
 #define CS_MIDDLE_32000_SCALE                        15
 #define CS_SIDE_32000_A0                            0.364579
-#define CS_SIDE_32000_A1                           -0.207355
-#define CS_SIDE_32000_A2                           -0.157224
-#define CS_SIDE_32000_B1                           -1.274231
+#define CS_SIDE_32000_A1                           (-0.207355)
+#define CS_SIDE_32000_A2                           (-0.157224)
+#define CS_SIDE_32000_B1                           (-1.274231)
 #define CS_SIDE_32000_B2                            0.312495
 #define CS_SIDE_32000_SCALE                          14
 
 /* Stereo Enhancer coefficients for 44100Hz sample rate, scaled with 0.163834 */
 #define CS_MIDDLE_44100_A0                     0.233593
-#define CS_MIDDLE_44100_A1                    -0.231225
+#define CS_MIDDLE_44100_A1                    (-0.231225)
 #define CS_MIDDLE_44100_A2                     0.000000
-#define CS_MIDDLE_44100_B1                    -0.985545
+#define CS_MIDDLE_44100_B1                    (-0.985545)
 #define CS_MIDDLE_44100_B2                     0.000000
 #define CS_MIDDLE_44100_SCALE                        15
 #define CS_SIDE_44100_A0                       0.284573
-#define CS_SIDE_44100_A1                      -0.258910
-#define CS_SIDE_44100_A2                      -0.025662
-#define CS_SIDE_44100_B1                      -1.572248
+#define CS_SIDE_44100_A1                      (-0.258910)
+#define CS_SIDE_44100_A2                      (-0.025662)
+#define CS_SIDE_44100_B1                      (-1.572248)
 #define CS_SIDE_44100_B2                       0.588399
 #define CS_SIDE_44100_SCALE                  14
 
 /* Stereo Enhancer coefficients for 48000Hz sample rate, scaled with 0.164402 */
 #define CS_MIDDLE_48000_A0                     0.234445
-#define CS_MIDDLE_48000_A1                    -0.232261
+#define CS_MIDDLE_48000_A1                    (-0.232261)
 #define CS_MIDDLE_48000_A2                     0.000000
-#define CS_MIDDLE_48000_B1                    -0.986713
+#define CS_MIDDLE_48000_B1                    (-0.986713)
 #define CS_MIDDLE_48000_B2                     0.000000
 #define CS_MIDDLE_48000_SCALE                        15
 #define CS_SIDE_48000_A0                     0.272606
-#define CS_SIDE_48000_A1                    -0.266952
-#define CS_SIDE_48000_A2                    -0.005654
-#define CS_SIDE_48000_B1                    -1.617141
+#define CS_SIDE_48000_A1                    (-0.266952)
+#define CS_SIDE_48000_A2                    (-0.005654)
+#define CS_SIDE_48000_B1                    (-1.617141)
 #define CS_SIDE_48000_B2                     0.630405
 #define CS_SIDE_48000_SCALE                          14
 
@@ -155,31 +155,31 @@
 /* Stereo Enhancer coefficients for 96000Hz sample rate, scaled with  0.165*/
 /* high pass filter with cutoff frequency 102.