am e5eda367: (-s ours) am afc7b857: am e849af87: Cherry-pick two BufferItem changes
* commit 'e5eda367571db5894024d0d71e3831484ef53141':
Cherry-pick two BufferItem changes
diff --git a/camera/Android.mk b/camera/Android.mk
index da5ac59..4c4700b 100644
--- a/camera/Android.mk
+++ b/camera/Android.mk
@@ -30,12 +30,10 @@
ICameraServiceListener.cpp \
ICameraRecordingProxy.cpp \
ICameraRecordingProxyListener.cpp \
- IProCameraUser.cpp \
- IProCameraCallbacks.cpp \
camera2/ICameraDeviceUser.cpp \
camera2/ICameraDeviceCallbacks.cpp \
camera2/CaptureRequest.cpp \
- ProCamera.cpp \
+ camera2/OutputConfiguration.cpp \
CameraBase.cpp \
CameraUtils.cpp \
VendorTagDescriptor.cpp
diff --git a/camera/CameraBase.cpp b/camera/CameraBase.cpp
index 65a1a47..5d50aa8 100644
--- a/camera/CameraBase.cpp
+++ b/camera/CameraBase.cpp
@@ -29,7 +29,6 @@
#include <camera/ICameraService.h>
// needed to instantiate
-#include <camera/ProCamera.h>
#include <camera/Camera.h>
#include <system/camera_metadata.h>
@@ -217,7 +216,6 @@
return cs->removeListener(listener);
}
-template class CameraBase<ProCamera>;
template class CameraBase<Camera>;
} // namespace android
diff --git a/camera/CameraMetadata.cpp b/camera/CameraMetadata.cpp
index 043437f..e216d26 100644
--- a/camera/CameraMetadata.cpp
+++ b/camera/CameraMetadata.cpp
@@ -74,7 +74,7 @@
clear();
}
-const camera_metadata_t* CameraMetadata::getAndLock() {
+const camera_metadata_t* CameraMetadata::getAndLock() const {
mLocked = true;
return mBuffer;
}
diff --git a/camera/CameraParameters.cpp b/camera/CameraParameters.cpp
index e5e4e90..68969cf 100644
--- a/camera/CameraParameters.cpp
+++ b/camera/CameraParameters.cpp
@@ -526,8 +526,12 @@
!strcmp(format, PIXEL_FORMAT_RGBA8888) ?
HAL_PIXEL_FORMAT_RGBA_8888 : // RGB8888
!strcmp(format, PIXEL_FORMAT_BAYER_RGGB) ?
- HAL_PIXEL_FORMAT_RAW_SENSOR : // Raw sensor data
+ HAL_PIXEL_FORMAT_RAW16 : // Raw sensor data
-1;
}
+bool CameraParameters::isEmpty() const {
+ return mMap.isEmpty();
+}
+
}; // namespace android
diff --git a/camera/ICameraService.cpp b/camera/ICameraService.cpp
index fc3e437..51a775b 100644
--- a/camera/ICameraService.cpp
+++ b/camera/ICameraService.cpp
@@ -2,16 +2,16 @@
**
** Copyright 2008, 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
+** 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
+** 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
+** 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.
*/
@@ -29,8 +29,6 @@
#include <camera/ICameraService.h>
#include <camera/ICameraServiceListener.h>
-#include <camera/IProCameraUser.h>
-#include <camera/IProCameraCallbacks.h>
#include <camera/ICamera.h>
#include <camera/ICameraClient.h>
#include <camera/camera2/ICameraDeviceUser.h>
@@ -209,26 +207,18 @@
return status;
}
- // connect to camera service (pro client)
- virtual status_t connectPro(const sp<IProCameraCallbacks>& cameraCb, int cameraId,
- const String16 &clientPackageName, int clientUid,
- /*out*/
- sp<IProCameraUser>& device)
+ virtual status_t setTorchMode(const String16& cameraId, bool enabled,
+ const sp<IBinder>& clientBinder)
{
Parcel data, reply;
data.writeInterfaceToken(ICameraService::getInterfaceDescriptor());
- data.writeStrongBinder(IInterface::asBinder(cameraCb));
- data.writeInt32(cameraId);
- data.writeString16(clientPackageName);
- data.writeInt32(clientUid);
- remote()->transact(BnCameraService::CONNECT_PRO, data, &reply);
+ data.writeString16(cameraId);
+ data.writeInt32(enabled ? 1 : 0);
+ data.writeStrongBinder(clientBinder);
+ remote()->transact(BnCameraService::SET_TORCH_MODE, data, &reply);
if (readExceptionCode(reply)) return -EPROTO;
- status_t status = reply.readInt32();
- if (reply.readInt32() != 0) {
- device = interface_cast<IProCameraUser>(reply.readStrongBinder());
- }
- return status;
+ return reply.readInt32();
}
// connect to camera service (android.hardware.camera2.CameraDevice)
@@ -312,6 +302,15 @@
status_t res = data.readInt32();
return res;
}
+
+ virtual void notifySystemEvent(int eventId, int arg0) {
+ Parcel data, reply;
+ data.writeInt32(eventId);
+ data.writeInt32(arg0);
+ remote()->transact(BnCameraService::NOTIFY_SYSTEM_EVENT, data, &reply,
+ IBinder::FLAG_ONEWAY);
+ }
+
};
IMPLEMENT_META_INTERFACE(CameraService, "android.hardware.ICameraService");
@@ -390,26 +389,6 @@
}
return NO_ERROR;
} break;
- case CONNECT_PRO: {
- CHECK_INTERFACE(ICameraService, data, reply);
- sp<IProCameraCallbacks> cameraClient =
- interface_cast<IProCameraCallbacks>(data.readStrongBinder());
- int32_t cameraId = data.readInt32();
- const String16 clientName = data.readString16();
- int32_t clientUid = data.readInt32();
- sp<IProCameraUser> camera;
- status_t status = connectPro(cameraClient, cameraId,
- clientName, clientUid, /*out*/camera);
- reply->writeNoException();
- reply->writeInt32(status);
- if (camera != NULL) {
- reply->writeInt32(1);
- reply->writeStrongBinder(IInterface::asBinder(camera));
- } else {
- reply->writeInt32(0);
- }
- return NO_ERROR;
- } break;
case CONNECT_DEVICE: {
CHECK_INTERFACE(ICameraService, data, reply);
sp<ICameraDeviceCallbacks> cameraClient =
@@ -490,6 +469,23 @@
}
return NO_ERROR;
} break;
+ case SET_TORCH_MODE: {
+ CHECK_INTERFACE(ICameraService, data, reply);
+ String16 cameraId = data.readString16();
+ bool enabled = data.readInt32() != 0 ? true : false;
+ const sp<IBinder> clientBinder = data.readStrongBinder();
+ status_t status = setTorchMode(cameraId, enabled, clientBinder);
+ reply->writeNoException();
+ reply->writeInt32(status);
+ return NO_ERROR;
+ } break;
+ case NOTIFY_SYSTEM_EVENT: {
+ CHECK_INTERFACE(ICameraService, data, reply);
+ int eventId = data.readInt32();
+ int arg0 = data.readInt32();
+ notifySystemEvent(eventId, arg0);
+ return NO_ERROR;
+ } break;
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/camera/ICameraServiceListener.cpp b/camera/ICameraServiceListener.cpp
index b2f1729..90a8bc2 100644
--- a/camera/ICameraServiceListener.cpp
+++ b/camera/ICameraServiceListener.cpp
@@ -29,6 +29,7 @@
namespace {
enum {
STATUS_CHANGED = IBinder::FIRST_CALL_TRANSACTION,
+ TORCH_STATUS_CHANGED,
};
}; // namespace anonymous
@@ -54,8 +55,21 @@
data,
&reply,
IBinder::FLAG_ONEWAY);
+ }
- reply.readExceptionCode();
+ virtual void onTorchStatusChanged(TorchStatus status, const String16 &cameraId)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(
+ ICameraServiceListener::getInterfaceDescriptor());
+
+ data.writeInt32(static_cast<int32_t>(status));
+ data.writeString16(cameraId);
+
+ remote()->transact(TORCH_STATUS_CHANGED,
+ data,
+ &reply,
+ IBinder::FLAG_ONEWAY);
}
};
@@ -75,7 +89,16 @@
int32_t cameraId = data.readInt32();
onStatusChanged(status, cameraId);
- reply->writeNoException();
+
+ return NO_ERROR;
+ } break;
+ case TORCH_STATUS_CHANGED: {
+ CHECK_INTERFACE(ICameraServiceListener, data, reply);
+
+ TorchStatus status = static_cast<TorchStatus>(data.readInt32());
+ String16 cameraId = data.readString16();
+
+ onTorchStatusChanged(status, cameraId);
return NO_ERROR;
} break;
diff --git a/camera/IProCameraCallbacks.cpp b/camera/IProCameraCallbacks.cpp
deleted file mode 100644
index bd3d420..0000000
--- a/camera/IProCameraCallbacks.cpp
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
-**
-** Copyright 2013, 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 "IProCameraCallbacks"
-#include <utils/Log.h>
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <binder/Parcel.h>
-#include <gui/IGraphicBufferProducer.h>
-#include <gui/Surface.h>
-#include <utils/Mutex.h>
-
-#include <camera/IProCameraCallbacks.h>
-
-#include "camera/CameraMetadata.h"
-
-namespace android {
-
-enum {
- NOTIFY_CALLBACK = IBinder::FIRST_CALL_TRANSACTION,
- LOCK_STATUS_CHANGED,
- RESULT_RECEIVED,
-};
-
-class BpProCameraCallbacks: public BpInterface<IProCameraCallbacks>
-{
-public:
- BpProCameraCallbacks(const sp<IBinder>& impl)
- : BpInterface<IProCameraCallbacks>(impl)
- {
- }
-
- // generic callback from camera service to app
- void notifyCallback(int32_t msgType, int32_t ext1, int32_t ext2)
- {
- ALOGV("notifyCallback");
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraCallbacks::getInterfaceDescriptor());
- data.writeInt32(msgType);
- data.writeInt32(ext1);
- data.writeInt32(ext2);
- remote()->transact(NOTIFY_CALLBACK, data, &reply, IBinder::FLAG_ONEWAY);
- }
-
- void onLockStatusChanged(LockStatus newLockStatus) {
- ALOGV("onLockStatusChanged");
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraCallbacks::getInterfaceDescriptor());
- data.writeInt32(newLockStatus);
- remote()->transact(LOCK_STATUS_CHANGED, data, &reply,
- IBinder::FLAG_ONEWAY);
- }
-
- void onResultReceived(int32_t requestId, camera_metadata* result) {
- ALOGV("onResultReceived");
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraCallbacks::getInterfaceDescriptor());
- data.writeInt32(requestId);
- CameraMetadata::writeToParcel(data, result);
- remote()->transact(RESULT_RECEIVED, data, &reply, IBinder::FLAG_ONEWAY);
- }
-};
-
-IMPLEMENT_META_INTERFACE(ProCameraCallbacks,
- "android.hardware.IProCameraCallbacks");
-
-// ----------------------------------------------------------------------
-
-status_t BnProCameraCallbacks::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
- ALOGV("onTransact - code = %d", code);
- switch(code) {
- case NOTIFY_CALLBACK: {
- ALOGV("NOTIFY_CALLBACK");
- CHECK_INTERFACE(IProCameraCallbacks, data, reply);
- int32_t msgType = data.readInt32();
- int32_t ext1 = data.readInt32();
- int32_t ext2 = data.readInt32();
- notifyCallback(msgType, ext1, ext2);
- return NO_ERROR;
- } break;
- case LOCK_STATUS_CHANGED: {
- ALOGV("LOCK_STATUS_CHANGED");
- CHECK_INTERFACE(IProCameraCallbacks, data, reply);
- LockStatus newLockStatus
- = static_cast<LockStatus>(data.readInt32());
- onLockStatusChanged(newLockStatus);
- return NO_ERROR;
- } break;
- case RESULT_RECEIVED: {
- ALOGV("RESULT_RECEIVED");
- CHECK_INTERFACE(IProCameraCallbacks, data, reply);
- int32_t requestId = data.readInt32();
- camera_metadata_t *result = NULL;
- CameraMetadata::readFromParcel(data, &result);
- onResultReceived(requestId, result);
- return NO_ERROR;
- break;
- }
- default:
- return BBinder::onTransact(code, data, reply, flags);
- }
-}
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
-
diff --git a/camera/IProCameraUser.cpp b/camera/IProCameraUser.cpp
deleted file mode 100644
index 9bd7597..0000000
--- a/camera/IProCameraUser.cpp
+++ /dev/null
@@ -1,324 +0,0 @@
-/*
-**
-** Copyright 2013, 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 "IProCameraUser"
-#include <utils/Log.h>
-#include <stdint.h>
-#include <sys/types.h>
-#include <binder/Parcel.h>
-#include <camera/IProCameraUser.h>
-#include <gui/IGraphicBufferProducer.h>
-#include <gui/Surface.h>
-#include "camera/CameraMetadata.h"
-
-namespace android {
-
-enum {
- DISCONNECT = IBinder::FIRST_CALL_TRANSACTION,
- CONNECT,
- EXCLUSIVE_TRY_LOCK,
- EXCLUSIVE_LOCK,
- EXCLUSIVE_UNLOCK,
- HAS_EXCLUSIVE_LOCK,
- SUBMIT_REQUEST,
- CANCEL_REQUEST,
- DELETE_STREAM,
- CREATE_STREAM,
- CREATE_DEFAULT_REQUEST,
- GET_CAMERA_INFO,
-};
-
-class BpProCameraUser: public BpInterface<IProCameraUser>
-{
-public:
- BpProCameraUser(const sp<IBinder>& impl)
- : BpInterface<IProCameraUser>(impl)
- {
- }
-
- // disconnect from camera service
- void disconnect()
- {
- ALOGV("disconnect");
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
- remote()->transact(DISCONNECT, data, &reply);
- reply.readExceptionCode();
- }
-
- virtual status_t connect(const sp<IProCameraCallbacks>& cameraClient)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
- data.writeStrongBinder(IInterface::asBinder(cameraClient));
- remote()->transact(CONNECT, data, &reply);
- return reply.readInt32();
- }
-
- /* Shared ProCameraUser */
-
- virtual status_t exclusiveTryLock()
- {
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
- remote()->transact(EXCLUSIVE_TRY_LOCK, data, &reply);
- return reply.readInt32();
- }
- virtual status_t exclusiveLock()
- {
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
- remote()->transact(EXCLUSIVE_LOCK, data, &reply);
- return reply.readInt32();
- }
-
- virtual status_t exclusiveUnlock()
- {
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
- remote()->transact(EXCLUSIVE_UNLOCK, data, &reply);
- return reply.readInt32();
- }
-
- virtual bool hasExclusiveLock()
- {
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
- remote()->transact(HAS_EXCLUSIVE_LOCK, data, &reply);
- return !!reply.readInt32();
- }
-
- virtual int submitRequest(camera_metadata_t* metadata, bool streaming)
- {
-
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
-
- // arg0+arg1
- CameraMetadata::writeToParcel(data, metadata);
-
- // arg2 = streaming (bool)
- data.writeInt32(streaming);
-
- remote()->transact(SUBMIT_REQUEST, data, &reply);
- return reply.readInt32();
- }
-
- virtual status_t cancelRequest(int requestId)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
- data.writeInt32(requestId);
-
- remote()->transact(CANCEL_REQUEST, data, &reply);
- return reply.readInt32();
- }
-
- virtual status_t deleteStream(int streamId)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
- data.writeInt32(streamId);
-
- remote()->transact(DELETE_STREAM, data, &reply);
- return reply.readInt32();
- }
-
- virtual status_t createStream(int width, int height, int format,
- const sp<IGraphicBufferProducer>& bufferProducer,
- /*out*/
- int* streamId)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
- data.writeInt32(width);
- data.writeInt32(height);
- data.writeInt32(format);
-
- sp<IBinder> b(IInterface::asBinder(bufferProducer));
- data.writeStrongBinder(b);
-
- remote()->transact(CREATE_STREAM, data, &reply);
-
- int sId = reply.readInt32();
- if (streamId) {
- *streamId = sId;
- }
- return reply.readInt32();
- }
-
- // Create a request object from a template.
- virtual status_t createDefaultRequest(int templateId,
- /*out*/
- camera_metadata** request)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
- data.writeInt32(templateId);
- remote()->transact(CREATE_DEFAULT_REQUEST, data, &reply);
- CameraMetadata::readFromParcel(reply, /*out*/request);
- return reply.readInt32();
- }
-
-
- virtual status_t getCameraInfo(int cameraId, camera_metadata** info)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IProCameraUser::getInterfaceDescriptor());
- data.writeInt32(cameraId);
- remote()->transact(GET_CAMERA_INFO, data, &reply);
- CameraMetadata::readFromParcel(reply, /*out*/info);
- return reply.readInt32();
- }
-
-
-private:
-
-
-};
-
-IMPLEMENT_META_INTERFACE(ProCameraUser, "android.hardware.IProCameraUser");
-
-// ----------------------------------------------------------------------
-
-status_t BnProCameraUser::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
- switch(code) {
- case DISCONNECT: {
- ALOGV("DISCONNECT");
- CHECK_INTERFACE(IProCameraUser, data, reply);
- disconnect();
- reply->writeNoException();
- return NO_ERROR;
- } break;
- case CONNECT: {
- CHECK_INTERFACE(IProCameraUser, data, reply);
- sp<IProCameraCallbacks> cameraClient =
- interface_cast<IProCameraCallbacks>(data.readStrongBinder());
- reply->writeInt32(connect(cameraClient));
- return NO_ERROR;
- } break;
-
- /* Shared ProCameraUser */
- case EXCLUSIVE_TRY_LOCK: {
- CHECK_INTERFACE(IProCameraUser, data, reply);
- reply->writeInt32(exclusiveTryLock());
- return NO_ERROR;
- } break;
- case EXCLUSIVE_LOCK: {
- CHECK_INTERFACE(IProCameraUser, data, reply);
- reply->writeInt32(exclusiveLock());
- return NO_ERROR;
- } break;
- case EXCLUSIVE_UNLOCK: {
- CHECK_INTERFACE(IProCameraUser, data, reply);
- reply->writeInt32(exclusiveUnlock());
- return NO_ERROR;
- } break;
- case HAS_EXCLUSIVE_LOCK: {
- CHECK_INTERFACE(IProCameraUser, data, reply);
- reply->writeInt32(hasExclusiveLock());
- return NO_ERROR;
- } break;
- case SUBMIT_REQUEST: {
- CHECK_INTERFACE(IProCameraUser, data, reply);
- camera_metadata_t* metadata;
- CameraMetadata::readFromParcel(data, /*out*/&metadata);
-
- // arg2 = streaming (bool)
- bool streaming = data.readInt32();
-
- // return code: requestId (int32)
- reply->writeInt32(submitRequest(metadata, streaming));
-
- return NO_ERROR;
- } break;
- case CANCEL_REQUEST: {
- CHECK_INTERFACE(IProCameraUser, data, reply);
- int requestId = data.readInt32();
- reply->writeInt32(cancelRequest(requestId));
- return NO_ERROR;
- } break;
- case DELETE_STREAM: {
- CHECK_INTERFACE(IProCameraUser, data, reply);
- int streamId = data.readInt32();
- reply->writeInt32(deleteStream(streamId));
- return NO_ERROR;
- } break;
- case CREATE_STREAM: {
- CHECK_INTERFACE(IProCameraUser, data, reply);
- int width, height, format;
-
- width = data.readInt32();
- height = data.readInt32();
- format = data.readInt32();
-
- sp<IGraphicBufferProducer> bp =
- interface_cast<IGraphicBufferProducer>(data.readStrongBinder());
-
- int streamId = -1;
- status_t ret;
- ret = createStream(width, height, format, bp, &streamId);
-
- reply->writeInt32(streamId);
- reply->writeInt32(ret);
-
- return NO_ERROR;
- } break;
-
- case CREATE_DEFAULT_REQUEST: {
- CHECK_INTERFACE(IProCameraUser, data, reply);
-
- int templateId = data.readInt32();
-
- camera_metadata_t* request = NULL;
- status_t ret;
- ret = createDefaultRequest(templateId, &request);
-
- CameraMetadata::writeToParcel(*reply, request);
- reply->writeInt32(ret);
-
- free_camera_metadata(request);
-
- return NO_ERROR;
- } break;
- case GET_CAMERA_INFO: {
- CHECK_INTERFACE(IProCameraUser, data, reply);
-
- int cameraId = data.readInt32();
-
- camera_metadata_t* info = NULL;
- status_t ret;
- ret = getCameraInfo(cameraId, &info);
-
- CameraMetadata::writeToParcel(*reply, info);
- reply->writeInt32(ret);
-
- free_camera_metadata(info);
-
- return NO_ERROR;
- } break;
- default:
- return BBinder::onTransact(code, data, reply, flags);
- }
-}
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
diff --git a/camera/ProCamera.cpp b/camera/ProCamera.cpp
deleted file mode 100644
index 48f8e8e..0000000
--- a/camera/ProCamera.cpp
+++ /dev/null
@@ -1,436 +0,0 @@
-/*
-**
-** Copyright (C) 2013, 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 "ProCamera"
-#include <utils/Log.h>
-#include <utils/threads.h>
-#include <utils/Mutex.h>
-
-#include <binder/IPCThreadState.h>
-#include <binder/IServiceManager.h>
-#include <binder/IMemory.h>
-
-#include <camera/ProCamera.h>
-#include <camera/IProCameraUser.h>
-#include <camera/IProCameraCallbacks.h>
-
-#include <gui/IGraphicBufferProducer.h>
-
-#include <system/camera_metadata.h>
-
-namespace android {
-
-sp<ProCamera> ProCamera::connect(int cameraId)
-{
- return CameraBaseT::connect(cameraId, String16(),
- ICameraService::USE_CALLING_UID);
-}
-
-ProCamera::ProCamera(int cameraId)
- : CameraBase(cameraId)
-{
-}
-
-CameraTraits<ProCamera>::TCamConnectService CameraTraits<ProCamera>::fnConnectService =
- &ICameraService::connectPro;
-
-ProCamera::~ProCamera()
-{
-
-}
-
-/* IProCameraUser's implementation */
-
-// callback from camera service
-void ProCamera::notifyCallback(int32_t msgType, int32_t ext1, int32_t ext2)
-{
- return CameraBaseT::notifyCallback(msgType, ext1, ext2);
-}
-
-void ProCamera::onLockStatusChanged(
- IProCameraCallbacks::LockStatus newLockStatus)
-{
- ALOGV("%s: newLockStatus = %d", __FUNCTION__, newLockStatus);
-
- sp<ProCameraListener> listener;
- {
- Mutex::Autolock _l(mLock);
- listener = mListener;
- }
- if (listener != NULL) {
- switch (newLockStatus) {
- case IProCameraCallbacks::LOCK_ACQUIRED:
- listener->onLockAcquired();
- break;
- case IProCameraCallbacks::LOCK_RELEASED:
- listener->onLockReleased();
- break;
- case IProCameraCallbacks::LOCK_STOLEN:
- listener->onLockStolen();
- break;
- default:
- ALOGE("%s: Unknown lock status: %d",
- __FUNCTION__, newLockStatus);
- }
- }
-}
-
-void ProCamera::onResultReceived(int32_t requestId, camera_metadata* result) {
- ALOGV("%s: requestId = %d, result = %p", __FUNCTION__, requestId, result);
-
- sp<ProCameraListener> listener;
- {
- Mutex::Autolock _l(mLock);
- listener = mListener;
- }
-
- CameraMetadata tmp(result);
-
- // Unblock waitForFrame(id) callers
- {
- Mutex::Autolock al(mWaitMutex);
- mMetadataReady = true;
- mLatestMetadata = tmp; // make copy
- mWaitCondition.broadcast();
- }
-
- result = tmp.release();
-
- if (listener != NULL) {
- listener->onResultReceived(requestId, result);
- } else {
- free_camera_metadata(result);
- }
-
-}
-
-status_t ProCamera::exclusiveTryLock()
-{
- sp <IProCameraUser> c = mCamera;
- if (c == 0) return NO_INIT;
-
- return c->exclusiveTryLock();
-}
-status_t ProCamera::exclusiveLock()
-{
- sp <IProCameraUser> c = mCamera;
- if (c == 0) return NO_INIT;
-
- return c->exclusiveLock();
-}
-status_t ProCamera::exclusiveUnlock()
-{
- sp <IProCameraUser> c = mCamera;
- if (c == 0) return NO_INIT;
-
- return c->exclusiveUnlock();
-}
-bool ProCamera::hasExclusiveLock()
-{
- sp <IProCameraUser> c = mCamera;
- if (c == 0) return NO_INIT;
-
- return c->hasExclusiveLock();
-}
-
-// Note that the callee gets a copy of the metadata.
-int ProCamera::submitRequest(const struct camera_metadata* metadata,
- bool streaming)
-{
- sp <IProCameraUser> c = mCamera;
- if (c == 0) return NO_INIT;
-
- return c->submitRequest(const_cast<struct camera_metadata*>(metadata),
- streaming);
-}
-
-status_t ProCamera::cancelRequest(int requestId)
-{
- sp <IProCameraUser> c = mCamera;
- if (c == 0) return NO_INIT;
-
- return c->cancelRequest(requestId);
-}
-
-status_t ProCamera::deleteStream(int streamId)
-{
- sp <IProCameraUser> c = mCamera;
- if (c == 0) return NO_INIT;
-
- status_t s = c->deleteStream(streamId);
-
- mStreams.removeItem(streamId);
-
- return s;
-}
-
-status_t ProCamera::createStream(int width, int height, int format,
- const sp<Surface>& surface,
- /*out*/
- int* streamId)
-{
- *streamId = -1;
-
- ALOGV("%s: createStreamW %dx%d (fmt=0x%x)", __FUNCTION__, width, height,
- format);
-
- if (surface == 0) {
- return BAD_VALUE;
- }
-
- return createStream(width, height, format,
- surface->getIGraphicBufferProducer(),
- streamId);
-}
-
-status_t ProCamera::createStream(int width, int height, int format,
- const sp<IGraphicBufferProducer>& bufferProducer,
- /*out*/
- int* streamId) {
- *streamId = -1;
-
- ALOGV("%s: createStreamT %dx%d (fmt=0x%x)", __FUNCTION__, width, height,
- format);
-
- if (bufferProducer == 0) {
- return BAD_VALUE;
- }
-
- sp <IProCameraUser> c = mCamera;
- status_t stat = c->createStream(width, height, format, bufferProducer,
- streamId);
-
- if (stat == OK) {
- StreamInfo s(*streamId);
-
- mStreams.add(*streamId, s);
- }
-
- return stat;
-}
-
-status_t ProCamera::createStreamCpu(int width, int height, int format,
- int heapCount,
- /*out*/
- sp<CpuConsumer>* cpuConsumer,
- int* streamId) {
- return createStreamCpu(width, height, format, heapCount,
- /*synchronousMode*/true,
- cpuConsumer, streamId);
-}
-
-status_t ProCamera::createStreamCpu(int width, int height, int format,
- int heapCount,
- bool synchronousMode,
- /*out*/
- sp<CpuConsumer>* cpuConsumer,
- int* streamId)
-{
- ALOGV("%s: createStreamW %dx%d (fmt=0x%x)", __FUNCTION__, width, height,
- format);
-
- *cpuConsumer = NULL;
-
- sp <IProCameraUser> c = mCamera;
- if (c == 0) return NO_INIT;
-
- sp<IGraphicBufferProducer> producer;
- sp<IGraphicBufferConsumer> consumer;
- BufferQueue::createBufferQueue(&producer, &consumer);
- sp<CpuConsumer> cc = new CpuConsumer(consumer, heapCount
- /*, synchronousMode*/);
- cc->setName(String8("ProCamera::mCpuConsumer"));
-
- sp<Surface> stc = new Surface(producer);
-
- status_t s = createStream(width, height, format,
- stc->getIGraphicBufferProducer(),
- streamId);
-
- if (s != OK) {
- ALOGE("%s: Failure to create stream %dx%d (fmt=0x%x)", __FUNCTION__,
- width, height, format);
- return s;
- }
-
- sp<ProFrameListener> frameAvailableListener =
- new ProFrameListener(this, *streamId);
-
- getStreamInfo(*streamId).cpuStream = true;
- getStreamInfo(*streamId).cpuConsumer = cc;
- getStreamInfo(*streamId).synchronousMode = synchronousMode;
- getStreamInfo(*streamId).stc = stc;
- // for lifetime management
- getStreamInfo(*streamId).frameAvailableListener = frameAvailableListener;
-
- cc->setFrameAvailableListener(frameAvailableListener);
-
- *cpuConsumer = cc;
-
- return s;
-}
-
-camera_metadata* ProCamera::getCameraInfo(int cameraId) {
- ALOGV("%s: cameraId = %d", __FUNCTION__, cameraId);
-
- sp <IProCameraUser> c = mCamera;
- if (c == 0) return NULL;
-
- camera_metadata* ptr = NULL;
- status_t status = c->getCameraInfo(cameraId, &ptr);
-
- if (status != OK) {
- ALOGE("%s: Failed to get camera info, error = %d", __FUNCTION__, status);
- }
-
- return ptr;
-}
-
-status_t ProCamera::createDefaultRequest(int templateId,
- camera_metadata** request) const {
- ALOGV("%s: templateId = %d", __FUNCTION__, templateId);
-
- sp <IProCameraUser> c = mCamera;
- if (c == 0) return NO_INIT;
-
- return c->createDefaultRequest(templateId, request);
-}
-
-void ProCamera::onFrameAvailable(int streamId) {
- ALOGV("%s: streamId = %d", __FUNCTION__, streamId);
-
- sp<ProCameraListener> listener = mListener;
- StreamInfo& stream = getStreamInfo(streamId);
-
- if (listener.get() != NULL) {
- listener->onFrameAvailable(streamId, stream.cpuConsumer);
- }
-
- // Unblock waitForFrame(id) callers
- {
- Mutex::Autolock al(mWaitMutex);
- getStreamInfo(streamId).frameReady++;
- mWaitCondition.broadcast();
- }
-}
-
-int ProCamera::waitForFrameBuffer(int streamId) {
- status_t stat = BAD_VALUE;
- Mutex::Autolock al(mWaitMutex);
-
- StreamInfo& si = getStreamInfo(streamId);
-
- if (si.frameReady > 0) {
- int numFrames = si.frameReady;
- si.frameReady = 0;
- return numFrames;
- } else {
- while (true) {
- stat = mWaitCondition.waitRelative(mWaitMutex,
- mWaitTimeout);
- if (stat != OK) {
- ALOGE("%s: Error while waiting for frame buffer: %d",
- __FUNCTION__, stat);
- return stat;
- }
-
- if (si.frameReady > 0) {
- int numFrames = si.frameReady;
- si.frameReady = 0;
- return numFrames;
- }
- // else it was some other stream that got unblocked
- }
- }
-
- return stat;
-}
-
-int ProCamera::dropFrameBuffer(int streamId, int count) {
- StreamInfo& si = getStreamInfo(streamId);
-
- if (!si.cpuStream) {
- return BAD_VALUE;
- } else if (count < 0) {
- return BAD_VALUE;
- }
-
- if (!si.synchronousMode) {
- ALOGW("%s: No need to drop frames on asynchronous streams,"
- " as asynchronous mode only keeps 1 latest frame around.",
- __FUNCTION__);
- return BAD_VALUE;
- }
-
- int numDropped = 0;
- for (int i = 0; i < count; ++i) {
- CpuConsumer::LockedBuffer buffer;
- if (si.cpuConsumer->lockNextBuffer(&buffer) != OK) {
- break;
- }
-
- si.cpuConsumer->unlockBuffer(buffer);
- numDropped++;
- }
-
- return numDropped;
-}
-
-status_t ProCamera::waitForFrameMetadata() {
- status_t stat = BAD_VALUE;
- Mutex::Autolock al(mWaitMutex);
-
- if (mMetadataReady) {
- return OK;
- } else {
- while (true) {
- stat = mWaitCondition.waitRelative(mWaitMutex,
- mWaitTimeout);
-
- if (stat != OK) {
- ALOGE("%s: Error while waiting for metadata: %d",
- __FUNCTION__, stat);
- return stat;
- }
-
- if (mMetadataReady) {
- mMetadataReady = false;
- return OK;
- }
- // else it was some other stream or metadata
- }
- }
-
- return stat;
-}
-
-CameraMetadata ProCamera::consumeFrameMetadata() {
- Mutex::Autolock al(mWaitMutex);
-
- // Destructive: Subsequent calls return empty metadatas
- CameraMetadata tmp = mLatestMetadata;
- mLatestMetadata.clear();
-
- return tmp;
-}
-
-ProCamera::StreamInfo& ProCamera::getStreamInfo(int streamId) {
- return mStreams.editValueFor(streamId);
-}
-
-}; // namespace android
diff --git a/camera/camera2/ICameraDeviceUser.cpp b/camera/camera2/ICameraDeviceUser.cpp
index 277b5db..89c6fb7 100644
--- a/camera/camera2/ICameraDeviceUser.cpp
+++ b/camera/camera2/ICameraDeviceUser.cpp
@@ -26,6 +26,7 @@
#include <gui/Surface.h>
#include <camera/CameraMetadata.h>
#include <camera/camera2/CaptureRequest.h>
+#include <camera/camera2/OutputConfiguration.h>
namespace android {
@@ -208,20 +209,16 @@
return reply.readInt32();
}
- virtual status_t createStream(int width, int height, int format,
- const sp<IGraphicBufferProducer>& bufferProducer)
+ virtual status_t createStream(const OutputConfiguration& outputConfiguration)
{
Parcel data, reply;
data.writeInterfaceToken(ICameraDeviceUser::getInterfaceDescriptor());
- data.writeInt32(width);
- data.writeInt32(height);
- data.writeInt32(format);
-
- data.writeInt32(1); // marker that bufferProducer is not null
- data.writeString16(String16("unknown_name")); // name of surface
- sp<IBinder> b(IInterface::asBinder(bufferProducer));
- data.writeStrongBinder(b);
-
+ if (outputConfiguration.getGraphicBufferProducer() != NULL) {
+ data.writeInt32(1); // marker that OutputConfiguration is not null. Mimic aidl behavior
+ outputConfiguration.writeToParcel(data);
+ } else {
+ data.writeInt32(0);
+ }
remote()->transact(CREATE_STREAM, data, &reply);
reply.readExceptionCode();
@@ -396,31 +393,15 @@
} break;
case CREATE_STREAM: {
CHECK_INTERFACE(ICameraDeviceUser, data, reply);
- int width, height, format;
- width = data.readInt32();
- ALOGV("%s: CREATE_STREAM: width = %d", __FUNCTION__, width);
- height = data.readInt32();
- ALOGV("%s: CREATE_STREAM: height = %d", __FUNCTION__, height);
- format = data.readInt32();
- ALOGV("%s: CREATE_STREAM: format = %d", __FUNCTION__, format);
-
- sp<IGraphicBufferProducer> bp;
+ status_t ret = BAD_VALUE;
if (data.readInt32() != 0) {
- String16 name = readMaybeEmptyString16(data);
- bp = interface_cast<IGraphicBufferProducer>(
- data.readStrongBinder());
-
- ALOGV("%s: CREATE_STREAM: bp = %p, name = %s", __FUNCTION__,
- bp.get(), String8(name).string());
+ OutputConfiguration outputConfiguration(data);
+ ret = createStream(outputConfiguration);
} else {
- ALOGV("%s: CREATE_STREAM: bp = unset, name = unset",
- __FUNCTION__);
+ ALOGE("%s: cannot take an empty OutputConfiguration", __FUNCTION__);
}
- status_t ret;
- ret = createStream(width, height, format, bp);
-
reply->writeNoException();
ALOGV("%s: CREATE_STREAM: write noException", __FUNCTION__);
reply->writeInt32(ret);
diff --git a/camera/camera2/OutputConfiguration.cpp b/camera/camera2/OutputConfiguration.cpp
new file mode 100644
index 0000000..24acaa0
--- /dev/null
+++ b/camera/camera2/OutputConfiguration.cpp
@@ -0,0 +1,79 @@
+/*
+**
+** Copyright 2015, 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 "OutputConfiguration"
+#include <utils/Log.h>
+
+#include <camera/camera2/OutputConfiguration.h>
+#include <binder/Parcel.h>
+
+namespace android {
+
+
+const int OutputConfiguration::INVALID_ROTATION = -1;
+
+// Read empty strings without printing a false error message.
+String16 OutputConfiguration::readMaybeEmptyString16(const Parcel& parcel) {
+ size_t len;
+ const char16_t* str = parcel.readString16Inplace(&len);
+ if (str != NULL) {
+ return String16(str, len);
+ } else {
+ return String16();
+ }
+}
+
+sp<IGraphicBufferProducer> OutputConfiguration::getGraphicBufferProducer() const {
+ return mGbp;
+}
+
+int OutputConfiguration::getRotation() const {
+ return mRotation;
+}
+
+OutputConfiguration::OutputConfiguration(const Parcel& parcel) {
+ status_t err;
+ int rotation = 0;
+ if ((err = parcel.readInt32(&rotation)) != OK) {
+ ALOGE("%s: Failed to read rotation from parcel", __FUNCTION__);
+ mGbp = NULL;
+ mRotation = INVALID_ROTATION;
+ return;
+ }
+
+ String16 name = readMaybeEmptyString16(parcel);
+ const sp<IGraphicBufferProducer>& gbp =
+ interface_cast<IGraphicBufferProducer>(parcel.readStrongBinder());
+ mGbp = gbp;
+ mRotation = rotation;
+
+ ALOGV("%s: OutputConfiguration: bp = %p, name = %s", __FUNCTION__,
+ gbp.get(), String8(name).string());
+}
+
+status_t OutputConfiguration::writeToParcel(Parcel& parcel) const {
+
+ parcel.writeInt32(mRotation);
+ parcel.writeString16(String16("unknown_name")); // name of surface
+ sp<IBinder> b(IInterface::asBinder(mGbp));
+ parcel.writeStrongBinder(b);
+
+ return OK;
+}
+
+}; // namespace android
+
diff --git a/camera/tests/Android.mk b/camera/tests/Android.mk
index 2db4c14..5d37f9e 100644
--- a/camera/tests/Android.mk
+++ b/camera/tests/Android.mk
@@ -17,7 +17,6 @@
LOCAL_ADDITIONAL_DEPENDENCIES := $(LOCAL_PATH)/Android.mk
LOCAL_SRC_FILES:= \
- ProCameraTests.cpp \
VendorTagDescriptorTests.cpp
LOCAL_SHARED_LIBRARIES := \
diff --git a/camera/tests/ProCameraTests.cpp b/camera/tests/ProCameraTests.cpp
deleted file mode 100644
index 1f5867a..0000000
--- a/camera/tests/ProCameraTests.cpp
+++ /dev/null
@@ -1,1278 +0,0 @@
-/*
- * Copyright (C) 2013 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 <iostream>
-
-#include <binder/IPCThreadState.h>
-#include <utils/Thread.h>
-
-#include "Camera.h"
-#include "ProCamera.h"
-#include <utils/Vector.h>
-#include <utils/Mutex.h>
-#include <utils/Condition.h>
-
-#include <gui/SurfaceComposerClient.h>
-#include <gui/Surface.h>
-
-#include <system/camera_metadata.h>
-#include <hardware/camera2.h> // for CAMERA2_TEMPLATE_PREVIEW only
-#include <camera/CameraMetadata.h>
-
-#include <camera/ICameraServiceListener.h>
-
-namespace android {
-namespace camera2 {
-namespace tests {
-namespace client {
-
-#define CAMERA_ID 0
-#define TEST_DEBUGGING 0
-
-#define TEST_LISTENER_TIMEOUT 1000000000 // 1 second listener timeout
-#define TEST_FORMAT HAL_PIXEL_FORMAT_Y16 //TODO: YUY2 instead
-
-#define TEST_FORMAT_MAIN HAL_PIXEL_FORMAT_Y8
-#define TEST_FORMAT_DEPTH HAL_PIXEL_FORMAT_Y16
-
-// defaults for display "test"
-#define TEST_DISPLAY_FORMAT HAL_PIXEL_FORMAT_Y8
-#define TEST_DISPLAY_WIDTH 320
-#define TEST_DISPLAY_HEIGHT 240
-
-#define TEST_CPU_FRAME_COUNT 2
-#define TEST_CPU_HEAP_COUNT 5
-
-#define TEST_FRAME_PROCESSING_DELAY_US 200000 // 200 ms
-
-#if TEST_DEBUGGING
-#define dout std::cerr
-#else
-#define dout if (0) std::cerr
-#endif
-
-#define EXPECT_OK(x) EXPECT_EQ(OK, (x))
-#define ASSERT_OK(x) ASSERT_EQ(OK, (x))
-
-class ProCameraTest;
-
-struct ServiceListener : public BnCameraServiceListener {
-
- ServiceListener() :
- mLatestStatus(STATUS_UNKNOWN),
- mPrevStatus(STATUS_UNKNOWN)
- {
- }
-
- void onStatusChanged(Status status, int32_t cameraId) {
- dout << "On status changed: 0x" << std::hex
- << (unsigned int) status << " cameraId " << cameraId
- << std::endl;
-
- Mutex::Autolock al(mMutex);
-
- mLatestStatus = status;
- mCondition.broadcast();
- }
-
- status_t waitForStatusChange(Status& newStatus) {
- Mutex::Autolock al(mMutex);
-
- if (mLatestStatus != mPrevStatus) {
- newStatus = mLatestStatus;
- mPrevStatus = mLatestStatus;
- return OK;
- }
-
- status_t stat = mCondition.waitRelative(mMutex,
- TEST_LISTENER_TIMEOUT);
-
- if (stat == OK) {
- newStatus = mLatestStatus;
- mPrevStatus = mLatestStatus;
- }
-
- return stat;
- }
-
- Condition mCondition;
- Mutex mMutex;
-
- Status mLatestStatus;
- Status mPrevStatus;
-};
-
-enum ProEvent {
- UNKNOWN,
- ACQUIRED,
- RELEASED,
- STOLEN,
- FRAME_RECEIVED,
- RESULT_RECEIVED,
-};
-
-inline int ProEvent_Mask(ProEvent e) {
- return (1 << static_cast<int>(e));
-}
-
-typedef Vector<ProEvent> EventList;
-
-class ProCameraTestThread : public Thread
-{
-public:
- ProCameraTestThread() {
- }
-
- virtual bool threadLoop() {
- mProc = ProcessState::self();
- mProc->startThreadPool();
-
- IPCThreadState *ptr = IPCThreadState::self();
-
- ptr->joinThreadPool();
-
- return false;
- }
-
- sp<ProcessState> mProc;
-};
-
-class ProCameraTestListener : public ProCameraListener {
-
-public:
- static const int EVENT_MASK_ALL = 0xFFFFFFFF;
-
- ProCameraTestListener() {
- mEventMask = EVENT_MASK_ALL;
- mDropFrames = false;
- }
-
- status_t WaitForEvent() {
- Mutex::Autolock cal(mConditionMutex);
-
- {
- Mutex::Autolock al(mListenerMutex);
-
- if (mProEventList.size() > 0) {
- return OK;
- }
- }
-
- return mListenerCondition.waitRelative(mConditionMutex,
- TEST_LISTENER_TIMEOUT);
- }
-
- /* Read events into out. Existing queue is flushed */
- void ReadEvents(EventList& out) {
- Mutex::Autolock al(mListenerMutex);
-
- for (size_t i = 0; i < mProEventList.size(); ++i) {
- out.push(mProEventList[i]);
- }
-
- mProEventList.clear();
- }
-
- /**
- * Dequeue 1 event from the event queue.
- * Returns UNKNOWN if queue is empty
- */
- ProEvent ReadEvent() {
- Mutex::Autolock al(mListenerMutex);
-
- if (mProEventList.size() == 0) {
- return UNKNOWN;
- }
-
- ProEvent ev = mProEventList[0];
- mProEventList.removeAt(0);
-
- return ev;
- }
-
- void SetEventMask(int eventMask) {
- Mutex::Autolock al(mListenerMutex);
- mEventMask = eventMask;
- }
-
- // Automatically acquire/release frames as they are available
- void SetDropFrames(bool dropFrames) {
- Mutex::Autolock al(mListenerMutex);
- mDropFrames = dropFrames;
- }
-
-private:
- void QueueEvent(ProEvent ev) {
- bool eventAdded = false;
- {
- Mutex::Autolock al(mListenerMutex);
-
- // Drop events not part of mask
- if (ProEvent_Mask(ev) & mEventMask) {
- mProEventList.push(ev);
- eventAdded = true;
- }
- }
-
- if (eventAdded) {
- mListenerCondition.broadcast();
- }
- }
-
-protected:
-
- //////////////////////////////////////////////////
- ///////// ProCameraListener //////////////////////
- //////////////////////////////////////////////////
-
-
- // Lock has been acquired. Write operations now available.
- virtual void onLockAcquired() {
- QueueEvent(ACQUIRED);
- }
- // Lock has been released with exclusiveUnlock
- virtual void onLockReleased() {
- QueueEvent(RELEASED);
- }
-
- // Lock has been stolen by another client.
- virtual void onLockStolen() {
- QueueEvent(STOLEN);
- }
-
- // Lock free.
- virtual void onTriggerNotify(int32_t ext1, int32_t ext2, int32_t ext3) {
-
- dout << "Trigger notify: " << ext1 << " " << ext2
- << " " << ext3 << std::endl;
- }
-
- virtual void onFrameAvailable(int streamId,
- const sp<CpuConsumer>& consumer) {
-
- QueueEvent(FRAME_RECEIVED);
-
- Mutex::Autolock al(mListenerMutex);
- if (mDropFrames) {
- CpuConsumer::LockedBuffer buf;
- status_t ret;
-
- if (OK == (ret = consumer->lockNextBuffer(&buf))) {
-
- dout << "Frame received on streamId = " << streamId <<
- ", dataPtr = " << (void*)buf.data <<
- ", timestamp = " << buf.timestamp << std::endl;
-
- EXPECT_OK(consumer->unlockBuffer(buf));
- }
- } else {
- dout << "Frame received on streamId = " << streamId << std::endl;
- }
- }
-
- virtual void onResultReceived(int32_t requestId,
- camera_metadata* request) {
- dout << "Result received requestId = " << requestId
- << ", requestPtr = " << (void*)request << std::endl;
- QueueEvent(RESULT_RECEIVED);
- free_camera_metadata(request);
- }
-
- virtual void notify(int32_t msg, int32_t ext1, int32_t ext2) {
- dout << "Notify received: msg " << std::hex << msg
- << ", ext1: " << std::hex << ext1 << ", ext2: " << std::hex << ext2
- << std::endl;
- }
-
- Vector<ProEvent> mProEventList;
- Mutex mListenerMutex;
- Mutex mConditionMutex;
- Condition mListenerCondition;
- int mEventMask;
- bool mDropFrames;
-};
-
-class ProCameraTest : public ::testing::Test {
-
-public:
- ProCameraTest() {
- char* displaySecsEnv = getenv("TEST_DISPLAY_SECS");
- if (displaySecsEnv != NULL) {
- mDisplaySecs = atoi(displaySecsEnv);
- if (mDisplaySecs < 0) {
- mDisplaySecs = 0;
- }
- } else {
- mDisplaySecs = 0;
- }
-
- char* displayFmtEnv = getenv("TEST_DISPLAY_FORMAT");
- if (displayFmtEnv != NULL) {
- mDisplayFmt = FormatFromString(displayFmtEnv);
- } else {
- mDisplayFmt = TEST_DISPLAY_FORMAT;
- }
-
- char* displayWidthEnv = getenv("TEST_DISPLAY_WIDTH");
- if (displayWidthEnv != NULL) {
- mDisplayW = atoi(displayWidthEnv);
- if (mDisplayW < 0) {
- mDisplayW = 0;
- }
- } else {
- mDisplayW = TEST_DISPLAY_WIDTH;
- }
-
- char* displayHeightEnv = getenv("TEST_DISPLAY_HEIGHT");
- if (displayHeightEnv != NULL) {
- mDisplayH = atoi(displayHeightEnv);
- if (mDisplayH < 0) {
- mDisplayH = 0;
- }
- } else {
- mDisplayH = TEST_DISPLAY_HEIGHT;
- }
- }
-
- static void SetUpTestCase() {
- // Binder Thread Pool Initialization
- mTestThread = new ProCameraTestThread();
- mTestThread->run("ProCameraTestThread");
- }
-
- virtual void SetUp() {
- mCamera = ProCamera::connect(CAMERA_ID);
- ASSERT_NE((void*)NULL, mCamera.get());
-
- mListener = new ProCameraTestListener();
- mCamera->setListener(mListener);
- }
-
- virtual void TearDown() {
- ASSERT_NE((void*)NULL, mCamera.get());
- mCamera->disconnect();
- }
-
-protected:
- sp<ProCamera> mCamera;
- sp<ProCameraTestListener> mListener;
-
- static sp<Thread> mTestThread;
-
- int mDisplaySecs;
- int mDisplayFmt;
- int mDisplayW;
- int mDisplayH;
-
- sp<SurfaceComposerClient> mComposerClient;
- sp<SurfaceControl> mSurfaceControl;
-
- sp<SurfaceComposerClient> mDepthComposerClient;
- sp<SurfaceControl> mDepthSurfaceControl;
-
- int getSurfaceWidth() {
- return 512;
- }
- int getSurfaceHeight() {
- return 512;
- }
-
- void createOnScreenSurface(sp<Surface>& surface) {
- mComposerClient = new SurfaceComposerClient;
- ASSERT_EQ(NO_ERROR, mComposerClient->initCheck());
-
- mSurfaceControl = mComposerClient->createSurface(
- String8("ProCameraTest StreamingImage Surface"),
- getSurfaceWidth(), getSurfaceHeight(),
- PIXEL_FORMAT_RGB_888, 0);
-
- mSurfaceControl->setPosition(0, 0);
-
- ASSERT_TRUE(mSurfaceControl != NULL);
- ASSERT_TRUE(mSurfaceControl->isValid());
-
- SurfaceComposerClient::openGlobalTransaction();
- ASSERT_EQ(NO_ERROR, mSurfaceControl->setLayer(0x7FFFFFFF));
- ASSERT_EQ(NO_ERROR, mSurfaceControl->show());
- SurfaceComposerClient::closeGlobalTransaction();
-
- sp<ANativeWindow> window = mSurfaceControl->getSurface();
- surface = mSurfaceControl->getSurface();
-
- ASSERT_NE((void*)NULL, surface.get());
- }
-
- void createDepthOnScreenSurface(sp<Surface>& surface) {
- mDepthComposerClient = new SurfaceComposerClient;
- ASSERT_EQ(NO_ERROR, mDepthComposerClient->initCheck());
-
- mDepthSurfaceControl = mDepthComposerClient->createSurface(
- String8("ProCameraTest StreamingImage Surface"),
- getSurfaceWidth(), getSurfaceHeight(),
- PIXEL_FORMAT_RGB_888, 0);
-
- mDepthSurfaceControl->setPosition(640, 0);
-
- ASSERT_TRUE(mDepthSurfaceControl != NULL);
- ASSERT_TRUE(mDepthSurfaceControl->isValid());
-
- SurfaceComposerClient::openGlobalTransaction();
- ASSERT_EQ(NO_ERROR, mDepthSurfaceControl->setLayer(0x7FFFFFFF));
- ASSERT_EQ(NO_ERROR, mDepthSurfaceControl->show());
- SurfaceComposerClient::closeGlobalTransaction();
-
- sp<ANativeWindow> window = mDepthSurfaceControl->getSurface();
- surface = mDepthSurfaceControl->getSurface();
-
- ASSERT_NE((void*)NULL, surface.get());
- }
-
- template <typename T>
- static bool ExistsItem(T needle, T* array, size_t count) {
- if (!array) {
- return false;
- }
-
- for (size_t i = 0; i < count; ++i) {
- if (array[i] == needle) {
- return true;
- }
- }
- return false;
- }
-
-
- static int FormatFromString(const char* str) {
- std::string s(str);
-
-#define CMP_STR(x, y) \
- if (s == #x) return HAL_PIXEL_FORMAT_ ## y;
-#define CMP_STR_SAME(x) CMP_STR(x, x)
-
- CMP_STR_SAME( Y16);
- CMP_STR_SAME( Y8);
- CMP_STR_SAME( YV12);
- CMP_STR(NV16, YCbCr_422_SP);
- CMP_STR(NV21, YCrCb_420_SP);
- CMP_STR(YUY2, YCbCr_422_I);
- CMP_STR(RAW, RAW_SENSOR);
- CMP_STR(RGBA, RGBA_8888);
-
- std::cerr << "Unknown format string " << str << std::endl;
- return -1;
-
- }
-
- /**
- * Creating a streaming request for these output streams from a template,
- * and submit it
- */
- void createSubmitRequestForStreams(int32_t* streamIds, size_t count, int requestCount=-1) {
-
- ASSERT_NE((void*)NULL, streamIds);
- ASSERT_LT(0u, count);
-
- camera_metadata_t *requestTmp = NULL;
- EXPECT_OK(mCamera->createDefaultRequest(CAMERA2_TEMPLATE_PREVIEW,
- /*out*/&requestTmp));
- ASSERT_NE((void*)NULL, requestTmp);
- CameraMetadata request(requestTmp);
-
- // set the output streams. default is empty
-
- uint32_t tag = static_cast<uint32_t>(ANDROID_REQUEST_OUTPUT_STREAMS);
- request.update(tag, streamIds, count);
-
- requestTmp = request.release();
-
- if (requestCount < 0) {
- EXPECT_OK(mCamera->submitRequest(requestTmp, /*streaming*/true));
- } else {
- for (int i = 0; i < requestCount; ++i) {
- EXPECT_OK(mCamera->submitRequest(requestTmp,
- /*streaming*/false));
- }
- }
- request.acquire(requestTmp);
- }
-};
-
-sp<Thread> ProCameraTest::mTestThread;
-
-TEST_F(ProCameraTest, AvailableFormats) {
- if (HasFatalFailure()) {
- return;
- }
-
- CameraMetadata staticInfo = mCamera->getCameraInfo(CAMERA_ID);
- ASSERT_FALSE(staticInfo.isEmpty());
-
- uint32_t tag = static_cast<uint32_t>(ANDROID_SCALER_AVAILABLE_FORMATS);
- EXPECT_TRUE(staticInfo.exists(tag));
- camera_metadata_entry_t entry = staticInfo.find(tag);
-
- EXPECT_TRUE(ExistsItem<int32_t>(HAL_PIXEL_FORMAT_YV12,
- entry.data.i32, entry.count));
- EXPECT_TRUE(ExistsItem<int32_t>(HAL_PIXEL_FORMAT_YCrCb_420_SP,
- entry.data.i32, entry.count));
-}
-
-// test around exclusiveTryLock (immediate locking)
-TEST_F(ProCameraTest, LockingImmediate) {
-
- if (HasFatalFailure()) {
- return;
- }
-
- mListener->SetEventMask(ProEvent_Mask(ACQUIRED) |
- ProEvent_Mask(STOLEN) |
- ProEvent_Mask(RELEASED));
-
- EXPECT_FALSE(mCamera->hasExclusiveLock());
- EXPECT_EQ(OK, mCamera->exclusiveTryLock());
- // at this point we definitely have the lock
-
- EXPECT_EQ(OK, mListener->WaitForEvent());
- EXPECT_EQ(ACQUIRED, mListener->ReadEvent());
-
- EXPECT_TRUE(mCamera->hasExclusiveLock());
- EXPECT_EQ(OK, mCamera->exclusiveUnlock());
-
- EXPECT_EQ(OK, mListener->WaitForEvent());
- EXPECT_EQ(RELEASED, mListener->ReadEvent());
-
- EXPECT_FALSE(mCamera->hasExclusiveLock());
-}
-
-// test around exclusiveLock (locking at some future point in time)
-TEST_F(ProCameraTest, LockingAsynchronous) {
-
- if (HasFatalFailure()) {
- return;
- }
-
-
- mListener->SetEventMask(ProEvent_Mask(ACQUIRED) |
- ProEvent_Mask(STOLEN) |
- ProEvent_Mask(RELEASED));
-
- // TODO: Add another procamera that has a lock here.
- // then we can be test that the lock wont immediately be acquired
-
- EXPECT_FALSE(mCamera->hasExclusiveLock());
- EXPECT_EQ(OK, mCamera->exclusiveTryLock());
- // at this point we definitely have the lock
-
- EXPECT_EQ(OK, mListener->WaitForEvent());
- EXPECT_EQ(ACQUIRED, mListener->ReadEvent());
-
- EXPECT_TRUE(mCamera->hasExclusiveLock());
- EXPECT_EQ(OK, mCamera->exclusiveUnlock());
-
- EXPECT_EQ(OK, mListener->WaitForEvent());
- EXPECT_EQ(RELEASED, mListener->ReadEvent());
-
- EXPECT_FALSE(mCamera->hasExclusiveLock());
-}
-
-// Stream directly to the screen.
-TEST_F(ProCameraTest, DISABLED_StreamingImageSingle) {
- if (HasFatalFailure()) {
- return;
- }
-
- sp<Surface> surface;
- if (mDisplaySecs > 0) {
- createOnScreenSurface(/*out*/surface);
- }
- else {
- dout << "Skipping, will not render to screen" << std::endl;
- return;
- }
-
- int depthStreamId = -1;
-
- sp<ServiceListener> listener = new ServiceListener();
- EXPECT_OK(ProCamera::addServiceListener(listener));
-
- ServiceListener::Status currentStatus;
-
- // when subscribing a new listener,
- // we immediately get a callback to the current status
- while (listener->waitForStatusChange(/*out*/currentStatus) != OK);
- EXPECT_EQ(ServiceListener::STATUS_PRESENT, currentStatus);
-
- dout << "Will now stream and resume infinitely..." << std::endl;
- while (true) {
-
- if (currentStatus == ServiceListener::STATUS_PRESENT) {
-
- ASSERT_OK(mCamera->createStream(mDisplayW, mDisplayH, mDisplayFmt,
- surface,
- &depthStreamId));
- EXPECT_NE(-1, depthStreamId);
-
- EXPECT_OK(mCamera->exclusiveTryLock());
-
- int32_t streams[] = { depthStreamId };
- ASSERT_NO_FATAL_FAILURE(createSubmitRequestForStreams(
- streams,
- /*count*/1));
- }
-
- ServiceListener::Status stat = ServiceListener::STATUS_UNKNOWN;
-
- // TODO: maybe check for getch every once in a while?
- while (listener->waitForStatusChange(/*out*/stat) != OK);
-
- if (currentStatus != stat) {
- if (stat == ServiceListener::STATUS_PRESENT) {
- dout << "Reconnecting to camera" << std::endl;
- mCamera = ProCamera::connect(CAMERA_ID);
- } else if (stat == ServiceListener::STATUS_NOT_AVAILABLE) {
- dout << "Disconnecting from camera" << std::endl;
- mCamera->disconnect();
- } else if (stat == ServiceListener::STATUS_NOT_PRESENT) {
- dout << "Camera unplugged" << std::endl;
- mCamera = NULL;
- } else {
- dout << "Unknown status change "
- << std::hex << stat << std::endl;
- }
-
- currentStatus = stat;
- }
- }
-
- EXPECT_OK(ProCamera::removeServiceListener(listener));
- EXPECT_OK(mCamera->deleteStream(depthStreamId));
- EXPECT_OK(mCamera->exclusiveUnlock());
-}
-
-// Stream directly to the screen.
-TEST_F(ProCameraTest, DISABLED_StreamingImageDual) {
- if (HasFatalFailure()) {
- return;
- }
- sp<Surface> surface;
- sp<Surface> depthSurface;
- if (mDisplaySecs > 0) {
- createOnScreenSurface(/*out*/surface);
- createDepthOnScreenSurface(/*out*/depthSurface);
- }
-
- int streamId = -1;
- EXPECT_OK(mCamera->createStream(/*width*/1280, /*height*/960,
- TEST_FORMAT_MAIN, surface, &streamId));
- EXPECT_NE(-1, streamId);
-
- int depthStreamId = -1;
- EXPECT_OK(mCamera->createStream(/*width*/320, /*height*/240,
- TEST_FORMAT_DEPTH, depthSurface, &depthStreamId));
- EXPECT_NE(-1, depthStreamId);
-
- EXPECT_OK(mCamera->exclusiveTryLock());
- /*
- */
- /* iterate in a loop submitting requests every frame.
- * what kind of requests doesnt really matter, just whatever.
- */
-
- // it would probably be better to use CameraMetadata from camera service.
- camera_metadata_t *request = NULL;
- EXPECT_OK(mCamera->createDefaultRequest(CAMERA2_TEMPLATE_PREVIEW,
- /*out*/&request));
- EXPECT_NE((void*)NULL, request);
-
- /*FIXME: dont need this later, at which point the above should become an
- ASSERT_NE*/
- if(request == NULL) request = allocate_camera_metadata(10, 100);
-
- // set the output streams to just this stream ID
-
- // wow what a verbose API.
- int32_t allStreams[] = { streamId, depthStreamId };
- // IMPORTANT. bad things will happen if its not a uint8.
- size_t streamCount = sizeof(allStreams) / sizeof(allStreams[0]);
- camera_metadata_entry_t entry;
- uint32_t tag = static_cast<uint32_t>(ANDROID_REQUEST_OUTPUT_STREAMS);
- int find = find_camera_metadata_entry(request, tag, &entry);
- if (find == -ENOENT) {
- if (add_camera_metadata_entry(request, tag, &allStreams,
- /*data_count*/streamCount) != OK) {
- camera_metadata_t *tmp = allocate_camera_metadata(1000, 10000);
- ASSERT_OK(append_camera_metadata(tmp, request));
- free_camera_metadata(request);
- request = tmp;
-
- ASSERT_OK(add_camera_metadata_entry(request, tag, &allStreams,
- /*data_count*/streamCount));
- }
- } else {
- ASSERT_OK(update_camera_metadata_entry(request, entry.index,
- &allStreams, /*data_count*/streamCount, &entry));
- }
-
- EXPECT_OK(mCamera->submitRequest(request, /*streaming*/true));
-
- dout << "will sleep now for " << mDisplaySecs << std::endl;
- sleep(mDisplaySecs);
-
- free_camera_metadata(request);
-
- for (size_t i = 0; i < streamCount; ++i) {
- EXPECT_OK(mCamera->deleteStream(allStreams[i]));
- }
- EXPECT_OK(mCamera->exclusiveUnlock());
-}
-
-TEST_F(ProCameraTest, CpuConsumerSingle) {
- if (HasFatalFailure()) {
- return;
- }
-
- mListener->SetEventMask(ProEvent_Mask(ACQUIRED) |
- ProEvent_Mask(STOLEN) |
- ProEvent_Mask(RELEASED) |
- ProEvent_Mask(FRAME_RECEIVED));
- mListener->SetDropFrames(true);
-
- int streamId = -1;
- sp<CpuConsumer> consumer;
- EXPECT_OK(mCamera->createStreamCpu(/*width*/320, /*height*/240,
- TEST_FORMAT_DEPTH, TEST_CPU_HEAP_COUNT, &consumer, &streamId));
- EXPECT_NE(-1, streamId);
-
- EXPECT_OK(mCamera->exclusiveTryLock());
- EXPECT_EQ(OK, mListener->WaitForEvent());
- EXPECT_EQ(ACQUIRED, mListener->ReadEvent());
- /* iterate in a loop submitting requests every frame.
- * what kind of requests doesnt really matter, just whatever.
- */
-
- // it would probably be better to use CameraMetadata from camera service.
- camera_metadata_t *request = NULL;
- EXPECT_OK(mCamera->createDefaultRequest(CAMERA2_TEMPLATE_PREVIEW,
- /*out*/&request));
- EXPECT_NE((void*)NULL, request);
-
- /*FIXME: dont need this later, at which point the above should become an
- ASSERT_NE*/
- if(request == NULL) request = allocate_camera_metadata(10, 100);
-
- // set the output streams to just this stream ID
-
- int32_t allStreams[] = { streamId };
- camera_metadata_entry_t entry;
- uint32_t tag = static_cast<uint32_t>(ANDROID_REQUEST_OUTPUT_STREAMS);
- int find = find_camera_metadata_entry(request, tag, &entry);
- if (find == -ENOENT) {
- if (add_camera_metadata_entry(request, tag, &allStreams,
- /*data_count*/1) != OK) {
- camera_metadata_t *tmp = allocate_camera_metadata(1000, 10000);
- ASSERT_OK(append_camera_metadata(tmp, request));
- free_camera_metadata(request);
- request = tmp;
-
- ASSERT_OK(add_camera_metadata_entry(request, tag, &allStreams,
- /*data_count*/1));
- }
- } else {
- ASSERT_OK(update_camera_metadata_entry(request, entry.index,
- &allStreams, /*data_count*/1, &entry));
- }
-
- EXPECT_OK(mCamera->submitRequest(request, /*streaming*/true));
-
- // Consume a couple of frames
- for (int i = 0; i < TEST_CPU_FRAME_COUNT; ++i) {
- EXPECT_EQ(OK, mListener->WaitForEvent());
- EXPECT_EQ(FRAME_RECEIVED, mListener->ReadEvent());
- }
-
- // Done: clean up
- free_camera_metadata(request);
- EXPECT_OK(mCamera->deleteStream(streamId));
- EXPECT_OK(mCamera->exclusiveUnlock());
-}
-
-TEST_F(ProCameraTest, CpuConsumerDual) {
- if (HasFatalFailure()) {
- return;
- }
-
- mListener->SetEventMask(ProEvent_Mask(FRAME_RECEIVED));
- mListener->SetDropFrames(true);
-
- int streamId = -1;
- sp<CpuConsumer> consumer;
- EXPECT_OK(mCamera->createStreamCpu(/*width*/1280, /*height*/960,
- TEST_FORMAT_MAIN, TEST_CPU_HEAP_COUNT, &consumer, &streamId));
- EXPECT_NE(-1, streamId);
-
- int depthStreamId = -1;
- EXPECT_OK(mCamera->createStreamCpu(/*width*/320, /*height*/240,
- TEST_FORMAT_DEPTH, TEST_CPU_HEAP_COUNT, &consumer, &depthStreamId));
- EXPECT_NE(-1, depthStreamId);
-
- EXPECT_OK(mCamera->exclusiveTryLock());
- /*
- */
- /* iterate in a loop submitting requests every frame.
- * what kind of requests doesnt really matter, just whatever.
- */
-
- // it would probably be better to use CameraMetadata from camera service.
- camera_metadata_t *request = NULL;
- EXPECT_OK(mCamera->createDefaultRequest(CAMERA2_TEMPLATE_PREVIEW,
- /*out*/&request));
- EXPECT_NE((void*)NULL, request);
-
- if(request == NULL) request = allocate_camera_metadata(10, 100);
-
- // set the output streams to just this stream ID
-
- // wow what a verbose API.
- int32_t allStreams[] = { streamId, depthStreamId };
- size_t streamCount = 2;
- camera_metadata_entry_t entry;
- uint32_t tag = static_cast<uint32_t>(ANDROID_REQUEST_OUTPUT_STREAMS);
- int find = find_camera_metadata_entry(request, tag, &entry);
- if (find == -ENOENT) {
- if (add_camera_metadata_entry(request, tag, &allStreams,
- /*data_count*/streamCount) != OK) {
- camera_metadata_t *tmp = allocate_camera_metadata(1000, 10000);
- ASSERT_OK(append_camera_metadata(tmp, request));
- free_camera_metadata(request);
- request = tmp;
-
- ASSERT_OK(add_camera_metadata_entry(request, tag, &allStreams,
- /*data_count*/streamCount));
- }
- } else {
- ASSERT_OK(update_camera_metadata_entry(request, entry.index,
- &allStreams, /*data_count*/streamCount, &entry));
- }
-
- EXPECT_OK(mCamera->submitRequest(request, /*streaming*/true));
-
- // Consume a couple of frames
- for (int i = 0; i < TEST_CPU_FRAME_COUNT; ++i) {
- // stream id 1
- EXPECT_EQ(OK, mListener->WaitForEvent());
- EXPECT_EQ(FRAME_RECEIVED, mListener->ReadEvent());
-
- // stream id 2
- EXPECT_EQ(OK, mListener->WaitForEvent());
- EXPECT_EQ(FRAME_RECEIVED, mListener->ReadEvent());
-
- //TODO: events should be a struct with some data like the stream id
- }
-
- // Done: clean up
- free_camera_metadata(request);
- EXPECT_OK(mCamera->deleteStream(streamId));
- EXPECT_OK(mCamera->exclusiveUnlock());
-}
-
-TEST_F(ProCameraTest, ResultReceiver) {
- if (HasFatalFailure()) {
- return;
- }
-
- mListener->SetEventMask(ProEvent_Mask(RESULT_RECEIVED));
- mListener->SetDropFrames(true);
- //FIXME: if this is run right after the previous test we get FRAME_RECEIVED
- // need to filter out events at read time
-
- int streamId = -1;
- sp<CpuConsumer> consumer;
- EXPECT_OK(mCamera->createStreamCpu(/*width*/1280, /*height*/960,
- TEST_FORMAT_MAIN, TEST_CPU_HEAP_COUNT, &consumer, &streamId));
- EXPECT_NE(-1, streamId);
-
- EXPECT_OK(mCamera->exclusiveTryLock());
- /*
- */
- /* iterate in a loop submitting requests every frame.
- * what kind of requests doesnt really matter, just whatever.
- */
-
- camera_metadata_t *request = NULL;
- EXPECT_OK(mCamera->createDefaultRequest(CAMERA2_TEMPLATE_PREVIEW,
- /*out*/&request));
- EXPECT_NE((void*)NULL, request);
-
- /*FIXME*/
- if(request == NULL) request = allocate_camera_metadata(10, 100);
-
- // set the output streams to just this stream ID
-
- int32_t allStreams[] = { streamId };
- size_t streamCount = 1;
- camera_metadata_entry_t entry;
- uint32_t tag = static_cast<uint32_t>(ANDROID_REQUEST_OUTPUT_STREAMS);
- int find = find_camera_metadata_entry(request, tag, &entry);
- if (find == -ENOENT) {
- if (add_camera_metadata_entry(request, tag, &allStreams,
- /*data_count*/streamCount) != OK) {
- camera_metadata_t *tmp = allocate_camera_metadata(1000, 10000);
- ASSERT_OK(append_camera_metadata(tmp, request));
- free_camera_metadata(request);
- request = tmp;
-
- ASSERT_OK(add_camera_metadata_entry(request, tag, &allStreams,
- /*data_count*/streamCount));
- }
- } else {
- ASSERT_OK(update_camera_metadata_entry(request, entry.index,
- &allStreams, /*data_count*/streamCount, &entry));
- }
-
- EXPECT_OK(mCamera->submitRequest(request, /*streaming*/true));
-
- // Consume a couple of results
- for (int i = 0; i < TEST_CPU_FRAME_COUNT; ++i) {
- EXPECT_EQ(OK, mListener->WaitForEvent());
- EXPECT_EQ(RESULT_RECEIVED, mListener->ReadEvent());
- }
-
- // Done: clean up
- free_camera_metadata(request);
- EXPECT_OK(mCamera->deleteStream(streamId));
- EXPECT_OK(mCamera->exclusiveUnlock());
-}
-
-// FIXME: This is racy and sometimes fails on waitForFrameMetadata
-TEST_F(ProCameraTest, DISABLED_WaitForResult) {
- if (HasFatalFailure()) {
- return;
- }
-
- mListener->SetDropFrames(true);
-
- int streamId = -1;
- sp<CpuConsumer> consumer;
- EXPECT_OK(mCamera->createStreamCpu(/*width*/1280, /*height*/960,
- TEST_FORMAT_MAIN, TEST_CPU_HEAP_COUNT, &consumer, &streamId));
- EXPECT_NE(-1, streamId);
-
- EXPECT_OK(mCamera->exclusiveTryLock());
-
- int32_t streams[] = { streamId };
- ASSERT_NO_FATAL_FAILURE(createSubmitRequestForStreams(streams, /*count*/1));
-
- // Consume a couple of results
- for (int i = 0; i < TEST_CPU_FRAME_COUNT; ++i) {
- EXPECT_OK(mCamera->waitForFrameMetadata());
- CameraMetadata meta = mCamera->consumeFrameMetadata();
- EXPECT_FALSE(meta.isEmpty());
- }
-
- // Done: clean up
- EXPECT_OK(mCamera->deleteStream(streamId));
- EXPECT_OK(mCamera->exclusiveUnlock());
-}
-
-TEST_F(ProCameraTest, WaitForSingleStreamBuffer) {
- if (HasFatalFailure()) {
- return;
- }
-
- int streamId = -1;
- sp<CpuConsumer> consumer;
- EXPECT_OK(mCamera->createStreamCpu(/*width*/1280, /*height*/960,
- TEST_FORMAT_MAIN, TEST_CPU_HEAP_COUNT, &consumer, &streamId));
- EXPECT_NE(-1, streamId);
-
- EXPECT_OK(mCamera->exclusiveTryLock());
-
- int32_t streams[] = { streamId };
- ASSERT_NO_FATAL_FAILURE(createSubmitRequestForStreams(streams, /*count*/1,
- /*requests*/TEST_CPU_FRAME_COUNT));
-
- // Consume a couple of results
- for (int i = 0; i < TEST_CPU_FRAME_COUNT; ++i) {
- EXPECT_EQ(1, mCamera->waitForFrameBuffer(streamId));
-
- CpuConsumer::LockedBuffer buf;
- EXPECT_OK(consumer->lockNextBuffer(&buf));
-
- dout << "Buffer synchronously received on streamId = " << streamId <<
- ", dataPtr = " << (void*)buf.data <<
- ", timestamp = " << buf.timestamp << std::endl;
-
- EXPECT_OK(consumer->unlockBuffer(buf));
- }
-
- // Done: clean up
- EXPECT_OK(mCamera->deleteStream(streamId));
- EXPECT_OK(mCamera->exclusiveUnlock());
-}
-
-// FIXME: This is racy and sometimes fails on waitForFrameMetadata
-TEST_F(ProCameraTest, DISABLED_WaitForDualStreamBuffer) {
- if (HasFatalFailure()) {
- return;
- }
-
- const int REQUEST_COUNT = TEST_CPU_FRAME_COUNT * 10;
-
- // 15 fps
- int streamId = -1;
- sp<CpuConsumer> consumer;
- EXPECT_OK(mCamera->createStreamCpu(/*width*/1280, /*height*/960,
- TEST_FORMAT_MAIN, TEST_CPU_HEAP_COUNT, &consumer, &streamId));
- EXPECT_NE(-1, streamId);
-
- // 30 fps
- int depthStreamId = -1;
- sp<CpuConsumer> depthConsumer;
- EXPECT_OK(mCamera->createStreamCpu(/*width*/320, /*height*/240,
- TEST_FORMAT_DEPTH, TEST_CPU_HEAP_COUNT, &depthConsumer, &depthStreamId));
- EXPECT_NE(-1, depthStreamId);
-
- EXPECT_OK(mCamera->exclusiveTryLock());
-
- int32_t streams[] = { streamId, depthStreamId };
- ASSERT_NO_FATAL_FAILURE(createSubmitRequestForStreams(streams, /*count*/2,
- /*requests*/REQUEST_COUNT));
-
- int depthFrames = 0;
- int greyFrames = 0;
-
- // Consume two frames simultaneously. Unsynchronized by timestamps.
- for (int i = 0; i < REQUEST_COUNT; ++i) {
-
- // Exhaust event queue so it doesn't keep growing
- while (mListener->ReadEvent() != UNKNOWN);
-
- // Get the metadata
- EXPECT_OK(mCamera->waitForFrameMetadata());
- CameraMetadata meta = mCamera->consumeFrameMetadata();
- EXPECT_FALSE(meta.isEmpty());
-
- // Get the buffers
-
- EXPECT_EQ(1, mCamera->waitForFrameBuffer(depthStreamId));
-
- /**
- * Guaranteed to be able to consume the depth frame,
- * since we waited on it.
- */
- CpuConsumer::LockedBuffer depthBuffer;
- EXPECT_OK(depthConsumer->lockNextBuffer(&depthBuffer));
-
- dout << "Depth Buffer synchronously received on streamId = " <<
- streamId <<
- ", dataPtr = " << (void*)depthBuffer.data <<
- ", timestamp = " << depthBuffer.timestamp << std::endl;
-
- EXPECT_OK(depthConsumer->unlockBuffer(depthBuffer));
-
- depthFrames++;
-
-
- /** Consume Greyscale frames if there are any.
- * There may not be since it runs at half FPS */
- CpuConsumer::LockedBuffer greyBuffer;
- while (consumer->lockNextBuffer(&greyBuffer) == OK) {
-
- dout << "GRAY Buffer synchronously received on streamId = " <<
- streamId <<
- ", dataPtr = " << (void*)greyBuffer.data <<
- ", timestamp = " << greyBuffer.timestamp << std::endl;
-
- EXPECT_OK(consumer->unlockBuffer(greyBuffer));
-
- greyFrames++;
- }
- }
-
- dout << "Done, summary: depth frames " << std::dec << depthFrames
- << ", grey frames " << std::dec << greyFrames << std::endl;
-
- // Done: clean up
- EXPECT_OK(mCamera->deleteStream(streamId));
- EXPECT_OK(mCamera->exclusiveUnlock());
-}
-
-TEST_F(ProCameraTest, WaitForSingleStreamBufferAndDropFramesSync) {
- if (HasFatalFailure()) {
- return;
- }
-
- const int NUM_REQUESTS = 20 * TEST_CPU_FRAME_COUNT;
-
- int streamId = -1;
- sp<CpuConsumer> consumer;
- EXPECT_OK(mCamera->createStreamCpu(/*width*/1280, /*height*/960,
- TEST_FORMAT_MAIN, TEST_CPU_HEAP_COUNT,
- /*synchronousMode*/true, &consumer, &streamId));
- EXPECT_NE(-1, streamId);
-
- EXPECT_OK(mCamera->exclusiveTryLock());
-
- int32_t streams[] = { streamId };
- ASSERT_NO_FATAL_FAILURE(createSubmitRequestForStreams(streams, /*count*/1,
- /*requests*/NUM_REQUESTS));
-
- // Consume a couple of results
- for (int i = 0; i < NUM_REQUESTS; ++i) {
- int numFrames;
- EXPECT_TRUE((numFrames = mCamera->waitForFrameBuffer(streamId)) > 0);
-
- // Drop all but the newest framebuffer
- EXPECT_EQ(numFrames-1, mCamera->dropFrameBuffer(streamId, numFrames-1));
-
- dout << "Dropped " << (numFrames - 1) << " frames" << std::endl;
-
- // Skip the counter ahead, don't try to consume these frames again
- i += numFrames-1;
-
- // "Consume" the buffer
- CpuConsumer::LockedBuffer buf;
- EXPECT_OK(consumer->lockNextBuffer(&buf));
-
- dout << "Buffer synchronously received on streamId = " << streamId <<
- ", dataPtr = " << (void*)buf.data <<
- ", timestamp = " << buf.timestamp << std::endl;
-
- // Process at 10fps, stream is at 15fps.
- // This means we will definitely fill up the buffer queue with
- // extra buffers and need to drop them.
- usleep(TEST_FRAME_PROCESSING_DELAY_US);
-
- EXPECT_OK(consumer->unlockBuffer(buf));
- }
-
- // Done: clean up
- EXPECT_OK(mCamera->deleteStream(streamId));
- EXPECT_OK(mCamera->exclusiveUnlock());
-}
-
-TEST_F(ProCameraTest, WaitForSingleStreamBufferAndDropFramesAsync) {
- if (HasFatalFailure()) {
- return;
- }
-
- const int NUM_REQUESTS = 20 * TEST_CPU_FRAME_COUNT;
-
- int streamId = -1;
- sp<CpuConsumer> consumer;
- EXPECT_OK(mCamera->createStreamCpu(/*width*/1280, /*height*/960,
- TEST_FORMAT_MAIN, TEST_CPU_HEAP_COUNT,
- /*synchronousMode*/false, &consumer, &streamId));
- EXPECT_NE(-1, streamId);
-
- EXPECT_OK(mCamera->exclusiveTryLock());
-
- int32_t streams[] = { streamId };
- ASSERT_NO_FATAL_FAILURE(createSubmitRequestForStreams(streams, /*count*/1,
- /*requests*/NUM_REQUESTS));
-
- uint64_t lastFrameNumber = 0;
- int numFrames;
-
- // Consume a couple of results
- int i;
- for (i = 0; i < NUM_REQUESTS && lastFrameNumber < NUM_REQUESTS; ++i) {
- EXPECT_LT(0, (numFrames = mCamera->waitForFrameBuffer(streamId)));
-
- dout << "Dropped " << (numFrames - 1) << " frames" << std::endl;
-
- // Skip the counter ahead, don't try to consume these frames again
- i += numFrames-1;
-
- // "Consume" the buffer
- CpuConsumer::LockedBuffer buf;
-
- EXPECT_EQ(OK, consumer->lockNextBuffer(&buf));
-
- lastFrameNumber = buf.frameNumber;
-
- dout << "Buffer asynchronously received on streamId = " << streamId <<
- ", dataPtr = " << (void*)buf.data <<
- ", timestamp = " << buf.timestamp <<
- ", framenumber = " << buf.frameNumber << std::endl;
-
- // Process at 10fps, stream is at 15fps.
- // This means we will definitely fill up the buffer queue with
- // extra buffers and need to drop them.
- usleep(TEST_FRAME_PROCESSING_DELAY_US);
-
- EXPECT_OK(consumer->unlockBuffer(buf));
- }
-
- dout << "Done after " << i << " iterations " << std::endl;
-
- // Done: clean up
- EXPECT_OK(mCamera->deleteStream(streamId));
- EXPECT_OK(mCamera->exclusiveUnlock());
-}
-
-
-
-//TODO: refactor into separate file
-TEST_F(ProCameraTest, ServiceListenersSubscribe) {
-
- ASSERT_EQ(4u, sizeof(ServiceListener::Status));
-
- sp<ServiceListener> listener = new ServiceListener();
-
- EXPECT_EQ(BAD_VALUE, ProCamera::removeServiceListener(listener));
- EXPECT_OK(ProCamera::addServiceListener(listener));
-
- EXPECT_EQ(ALREADY_EXISTS, ProCamera::addServiceListener(listener));
- EXPECT_OK(ProCamera::removeServiceListener(listener));
-
- EXPECT_EQ(BAD_VALUE, ProCamera::removeServiceListener(listener));
-}
-
-//TODO: refactor into separate file
-TEST_F(ProCameraTest, ServiceListenersFunctional) {
-
- sp<ServiceListener> listener = new ServiceListener();
-
- EXPECT_OK(ProCamera::addServiceListener(listener));
-
- sp<Camera> cam = Camera::connect(CAMERA_ID,
- /*clientPackageName*/String16(),
- -1);
- EXPECT_NE((void*)NULL, cam.get());
-
- ServiceListener::Status stat = ServiceListener::STATUS_UNKNOWN;
- EXPECT_OK(listener->waitForStatusChange(/*out*/stat));
-
- EXPECT_EQ(ServiceListener::STATUS_NOT_AVAILABLE, stat);
-
- if (cam.get()) {
- cam->disconnect();
- }
-
- EXPECT_OK(listener->waitForStatusChange(/*out*/stat));
- EXPECT_EQ(ServiceListener::STATUS_PRESENT, stat);
-
- EXPECT_OK(ProCamera::removeServiceListener(listener));
-}
-
-
-
-}
-}
-}
-}
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 02df1d2..36a7e73 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -23,7 +23,10 @@
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
+#include <sys/stat.h>
+#include <sys/types.h>
#include <sys/wait.h>
+
#include <termios.h>
#include <unistd.h>
@@ -637,7 +640,13 @@
case FORMAT_MP4: {
// Configure muxer. We have to wait for the CSD blob from the encoder
// before we can start it.
- muxer = new MediaMuxer(fileName, MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+ int fd = open(fileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ if (fd < 0) {
+ fprintf(stderr, "ERROR: couldn't open file\n");
+ abort();
+ }
+ muxer = new MediaMuxer(fd, MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+ close(fd);
if (gRotate) {
muxer->setOrientationHint(90); // TODO: does this do anything?
}
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index 561ce02..0e3bc68 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -169,6 +169,48 @@
include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= \
+ filters/argbtorgba.rs \
+ filters/nightvision.rs \
+ filters/saturation.rs \
+ mediafilter.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright \
+ liblog \
+ libutils \
+ libbinder \
+ libstagefright_foundation \
+ libmedia \
+ libgui \
+ libcutils \
+ libui \
+ libRScpp \
+
+LOCAL_C_INCLUDES:= \
+ $(TOP)/frameworks/av/media/libstagefright \
+ $(TOP)/frameworks/native/include/media/openmax \
+ $(TOP)/frameworks/rs/cpp \
+ $(TOP)/frameworks/rs \
+
+intermediates := $(call intermediates-dir-for,STATIC_LIBRARIES,libRS,TARGET,)
+LOCAL_C_INCLUDES += $(intermediates)
+
+LOCAL_STATIC_LIBRARIES:= \
+ libstagefright_mediafilter
+
+LOCAL_CFLAGS += -Wno-multichar
+
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_MODULE:= mediafilter
+
+include $(BUILD_EXECUTABLE)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
LOCAL_SRC_FILES:= \
muxer.cpp \
diff --git a/cmds/stagefright/SimplePlayer.cpp b/cmds/stagefright/SimplePlayer.cpp
index 4b2f980..ac1a547 100644
--- a/cmds/stagefright/SimplePlayer.cpp
+++ b/cmds/stagefright/SimplePlayer.cpp
@@ -59,14 +59,14 @@
return err;
}
status_t SimplePlayer::setDataSource(const char *path) {
- sp<AMessage> msg = new AMessage(kWhatSetDataSource, id());
+ sp<AMessage> msg = new AMessage(kWhatSetDataSource, this);
msg->setString("path", path);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
}
status_t SimplePlayer::setSurface(const sp<IGraphicBufferProducer> &bufferProducer) {
- sp<AMessage> msg = new AMessage(kWhatSetSurface, id());
+ sp<AMessage> msg = new AMessage(kWhatSetSurface, this);
sp<Surface> surface;
if (bufferProducer != NULL) {
@@ -81,25 +81,25 @@
}
status_t SimplePlayer::prepare() {
- sp<AMessage> msg = new AMessage(kWhatPrepare, id());
+ sp<AMessage> msg = new AMessage(kWhatPrepare, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
}
status_t SimplePlayer::start() {
- sp<AMessage> msg = new AMessage(kWhatStart, id());
+ sp<AMessage> msg = new AMessage(kWhatStart, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
}
status_t SimplePlayer::stop() {
- sp<AMessage> msg = new AMessage(kWhatStop, id());
+ sp<AMessage> msg = new AMessage(kWhatStop, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
}
status_t SimplePlayer::reset() {
- sp<AMessage> msg = new AMessage(kWhatReset, id());
+ sp<AMessage> msg = new AMessage(kWhatReset, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
}
@@ -116,7 +116,7 @@
mState = UNPREPARED;
}
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
sp<AMessage> response = new AMessage;
@@ -139,7 +139,7 @@
err = OK;
}
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
sp<AMessage> response = new AMessage;
@@ -161,7 +161,7 @@
}
}
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
sp<AMessage> response = new AMessage;
@@ -194,7 +194,7 @@
}
}
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
sp<AMessage> response = new AMessage;
@@ -217,7 +217,7 @@
}
}
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
sp<AMessage> response = new AMessage;
@@ -240,7 +240,7 @@
mState = UNINITIALIZED;
}
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
sp<AMessage> response = new AMessage;
@@ -382,7 +382,7 @@
mStartTimeRealUs = -1ll;
- sp<AMessage> msg = new AMessage(kWhatDoMoreStuff, id());
+ sp<AMessage> msg = new AMessage(kWhatDoMoreStuff, this);
msg->setInt32("generation", ++mDoMoreStuffGeneration);
msg->post();
diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index 96073f1..7b0de24 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp
@@ -14,6 +14,10 @@
* limitations under the License.
*/
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+
#include <binder/ProcessState.h>
#include <media/mediarecorder.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -109,7 +113,12 @@
if (fileOut != NULL) {
// target file specified, write encoded AMR output
- sp<AMRWriter> writer = new AMRWriter(fileOut);
+ int fd = open(fileOut, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ if (fd < 0) {
+ return 1;
+ }
+ sp<AMRWriter> writer = new AMRWriter(fd);
+ close(fd);
writer->addSource(encoder);
writer->start();
sleep(duration);
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index fd02bcc..d987250 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp
@@ -45,9 +45,10 @@
fprintf(stderr, "usage: %s [-a] use audio\n"
"\t\t[-v] use video\n"
"\t\t[-p] playback\n"
- "\t\t[-S] allocate buffers from a surface\n",
+ "\t\t[-S] allocate buffers from a surface\n"
+ "\t\t[-R] render output to surface (enables -S)\n"
+ "\t\t[-T] use render timestamps (enables -R)\n",
me);
-
exit(1);
}
@@ -71,7 +72,9 @@
const char *path,
bool useAudio,
bool useVideo,
- const android::sp<android::Surface> &surface) {
+ const android::sp<android::Surface> &surface,
+ bool renderSurface,
+ bool useTimestamp) {
using namespace android;
static int64_t kTimeout = 500ll;
@@ -136,6 +139,7 @@
CHECK(!stateByTrack.isEmpty());
int64_t startTimeUs = ALooper::GetNowUs();
+ int64_t startTimeRender = -1;
for (size_t i = 0; i < stateByTrack.size(); ++i) {
CodecState *state = &stateByTrack.editValueAt(i);
@@ -260,7 +264,23 @@
++state->mNumBuffersDecoded;
state->mNumBytesDecoded += size;
- err = state->mCodec->releaseOutputBuffer(index);
+ if (surface == NULL || !renderSurface) {
+ err = state->mCodec->releaseOutputBuffer(index);
+ } else if (useTimestamp) {
+ if (startTimeRender == -1) {
+ // begin rendering 2 vsyncs (~33ms) after first decode
+ startTimeRender =
+ systemTime(SYSTEM_TIME_MONOTONIC) + 33000000
+ - (presentationTimeUs * 1000);
+ }
+ presentationTimeUs =
+ (presentationTimeUs * 1000) + startTimeRender;
+ err = state->mCodec->renderOutputBufferAndRelease(
+ index, presentationTimeUs);
+ } else {
+ err = state->mCodec->renderOutputBufferAndRelease(index);
+ }
+
CHECK_EQ(err, (status_t)OK);
if (flags & MediaCodec::BUFFER_FLAG_EOS) {
@@ -320,34 +340,42 @@
bool useVideo = false;
bool playback = false;
bool useSurface = false;
+ bool renderSurface = false;
+ bool useTimestamp = false;
int res;
- while ((res = getopt(argc, argv, "havpSD")) >= 0) {
+ while ((res = getopt(argc, argv, "havpSDRT")) >= 0) {
switch (res) {
case 'a':
{
useAudio = true;
break;
}
-
case 'v':
{
useVideo = true;
break;
}
-
case 'p':
{
playback = true;
break;
}
-
+ case 'T':
+ {
+ useTimestamp = true;
+ }
+ // fall through
+ case 'R':
+ {
+ renderSurface = true;
+ }
+ // fall through
case 'S':
{
useSurface = true;
break;
}
-
case '?':
case 'h':
default:
@@ -422,7 +450,8 @@
player->stop();
player->reset();
} else {
- decode(looper, argv[0], useAudio, useVideo, surface);
+ decode(looper, argv[0], useAudio, useVideo, surface, renderSurface,
+ useTimestamp);
}
if (playback || (useSurface && useVideo)) {
diff --git a/include/media/nbaio/roundup.h b/cmds/stagefright/filters/argbtorgba.rs
similarity index 65%
rename from include/media/nbaio/roundup.h
rename to cmds/stagefright/filters/argbtorgba.rs
index 4c3cc25..229ff8c 100644
--- a/include/media/nbaio/roundup.h
+++ b/cmds/stagefright/filters/argbtorgba.rs
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2012 The Android Open Source Project
+ * 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.
@@ -14,18 +14,13 @@
* limitations under the License.
*/
-#ifndef ROUNDUP_H
-#define ROUNDUP_H
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-// Round up to the next highest power of 2
-unsigned roundup(unsigned v);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // ROUNDUP_H
+void root(const uchar4 *v_in, uchar4 *v_out) {
+ v_out->x = v_in->y;
+ v_out->y = v_in->z;
+ v_out->z = v_in->w;
+ v_out->w = v_in->x;
+}
\ No newline at end of file
diff --git a/cmds/stagefright/filters/nightvision.rs b/cmds/stagefright/filters/nightvision.rs
new file mode 100644
index 0000000..f61413c
--- /dev/null
+++ b/cmds/stagefright/filters/nightvision.rs
@@ -0,0 +1,38 @@
+/*
+ * 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.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+const static float3 gNightVisionMult = {0.5f, 1.f, 0.5f};
+
+// calculates luminance of pixel, then biases color balance toward green
+void root(const uchar4 *v_in, uchar4 *v_out) {
+ v_out->x = v_in->x; // don't modify A
+
+ // get RGB, scale 0-255 uchar to 0-1.0 float
+ float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f,
+ v_in->w * 0.003921569f};
+
+ // apply filter
+ float3 result = dot(rgb, gMonoMult) * gNightVisionMult;
+
+ v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+ v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+ v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/cmds/stagefright/filters/saturation.rs b/cmds/stagefright/filters/saturation.rs
new file mode 100644
index 0000000..1de9dd8
--- /dev/null
+++ b/cmds/stagefright/filters/saturation.rs
@@ -0,0 +1,40 @@
+/*
+ * 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.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+
+// global variables (parameters accessible to application code)
+float gSaturation = 1.0f;
+
+void root(const uchar4 *v_in, uchar4 *v_out) {
+ v_out->x = v_in->x; // don't modify A
+
+ // get RGB, scale 0-255 uchar to 0-1.0 float
+ float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f,
+ v_in->w * 0.003921569f};
+
+ // apply saturation filter
+ float3 result = dot(rgb, gMonoMult);
+ result = mix(result, rgb, gSaturation);
+
+ v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+ v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+ v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/cmds/stagefright/mediafilter.cpp b/cmds/stagefright/mediafilter.cpp
new file mode 100644
index 0000000..f77b38b
--- /dev/null
+++ b/cmds/stagefright/mediafilter.cpp
@@ -0,0 +1,785 @@
+/*
+ * 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 "mediafilterTest"
+
+#include <inttypes.h>
+
+#include <binder/ProcessState.h>
+#include <filters/ColorConvert.h>
+#include <gui/ISurfaceComposer.h>
+#include <gui/SurfaceComposerClient.h>
+#include <gui/Surface.h>
+#include <media/ICrypto.h>
+#include <media/IMediaHTTPService.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/NuMediaExtractor.h>
+#include <media/stagefright/RenderScriptWrapper.h>
+#include <OMX_IVCommon.h>
+#include <ui/DisplayInfo.h>
+
+#include "RenderScript.h"
+#include "ScriptC_argbtorgba.h"
+#include "ScriptC_nightvision.h"
+#include "ScriptC_saturation.h"
+
+// test parameters
+static const bool kTestFlush = true; // Note: true will drop 1 out of
+static const int kFlushAfterFrames = 25; // kFlushAfterFrames output frames
+static const int64_t kTimeout = 500ll;
+
+// built-in filter parameters
+static const int32_t kInvert = false; // ZeroFilter param
+static const float kBlurRadius = 15.0f; // IntrinsicBlurFilter param
+static const float kSaturation = 0.0f; // SaturationFilter param
+
+static void usage(const char *me) {
+ fprintf(stderr, "usage: [flags] %s\n"
+ "\t[-b] use IntrinsicBlurFilter\n"
+ "\t[-c] use argb to rgba conversion RSFilter\n"
+ "\t[-n] use night vision RSFilter\n"
+ "\t[-r] use saturation RSFilter\n"
+ "\t[-s] use SaturationFilter\n"
+ "\t[-z] use ZeroFilter (copy filter)\n"
+ "\t[-R] render output to surface (enables -S)\n"
+ "\t[-S] allocate buffers from a surface\n"
+ "\t[-T] use render timestamps (enables -R)\n",
+ me);
+ exit(1);
+}
+
+namespace android {
+
+struct SaturationRSFilter : RenderScriptWrapper::RSFilterCallback {
+ void init(RSC::sp<RSC::RS> context) {
+ mScript = new ScriptC_saturation(context);
+ mScript->set_gSaturation(3.f);
+ }
+
+ virtual status_t processBuffers(
+ RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) {
+ mScript->forEach_root(inBuffer, outBuffer);
+
+ return OK;
+ }
+
+ status_t handleSetParameters(const sp<AMessage> &msg) {
+ return OK;
+ }
+
+private:
+ RSC::sp<ScriptC_saturation> mScript;
+};
+
+struct NightVisionRSFilter : RenderScriptWrapper::RSFilterCallback {
+ void init(RSC::sp<RSC::RS> context) {
+ mScript = new ScriptC_nightvision(context);
+ }
+
+ virtual status_t processBuffers(
+ RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) {
+ mScript->forEach_root(inBuffer, outBuffer);
+
+ return OK;
+ }
+
+ status_t handleSetParameters(const sp<AMessage> &msg) {
+ return OK;
+ }
+
+private:
+ RSC::sp<ScriptC_nightvision> mScript;
+};
+
+struct ARGBToRGBARSFilter : RenderScriptWrapper::RSFilterCallback {
+ void init(RSC::sp<RSC::RS> context) {
+ mScript = new ScriptC_argbtorgba(context);
+ }
+
+ virtual status_t processBuffers(
+ RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) {
+ mScript->forEach_root(inBuffer, outBuffer);
+
+ return OK;
+ }
+
+ status_t handleSetParameters(const sp<AMessage> &msg) {
+ return OK;
+ }
+
+private:
+ RSC::sp<ScriptC_argbtorgba> mScript;
+};
+
+struct CodecState {
+ sp<MediaCodec> mCodec;
+ Vector<sp<ABuffer> > mInBuffers;
+ Vector<sp<ABuffer> > mOutBuffers;
+ bool mSignalledInputEOS;
+ bool mSawOutputEOS;
+ int64_t mNumBuffersDecoded;
+};
+
+struct DecodedFrame {
+ size_t index;
+ size_t offset;
+ size_t size;
+ int64_t presentationTimeUs;
+ uint32_t flags;
+};
+
+enum FilterType {
+ FILTERTYPE_ZERO,
+ FILTERTYPE_INTRINSIC_BLUR,
+ FILTERTYPE_SATURATION,
+ FILTERTYPE_RS_SATURATION,
+ FILTERTYPE_RS_NIGHT_VISION,
+ FILTERTYPE_RS_ARGB_TO_RGBA,
+};
+
+size_t inputFramesSinceFlush = 0;
+void tryCopyDecodedBuffer(
+ List<DecodedFrame> *decodedFrameIndices,
+ CodecState *filterState,
+ CodecState *vidState) {
+ if (decodedFrameIndices->empty()) {
+ return;
+ }
+
+ size_t filterIndex;
+ status_t err = filterState->mCodec->dequeueInputBuffer(
+ &filterIndex, kTimeout);
+ if (err != OK) {
+ return;
+ }
+
+ ++inputFramesSinceFlush;
+
+ DecodedFrame frame = *decodedFrameIndices->begin();
+
+ // only consume a buffer if we are not going to flush, since we expect
+ // the dequeue -> flush -> queue operation to cause an error and
+ // not produce an output frame
+ if (!kTestFlush || inputFramesSinceFlush < kFlushAfterFrames) {
+ decodedFrameIndices->erase(decodedFrameIndices->begin());
+ }
+ size_t outIndex = frame.index;
+
+ const sp<ABuffer> &srcBuffer =
+ vidState->mOutBuffers.itemAt(outIndex);
+ const sp<ABuffer> &destBuffer =
+ filterState->mInBuffers.itemAt(filterIndex);
+
+ sp<AMessage> srcFormat, destFormat;
+ vidState->mCodec->getOutputFormat(&srcFormat);
+ filterState->mCodec->getInputFormat(&destFormat);
+
+ int32_t srcWidth, srcHeight, srcStride, srcSliceHeight;
+ int32_t srcColorFormat, destColorFormat;
+ int32_t destWidth, destHeight, destStride, destSliceHeight;
+ CHECK(srcFormat->findInt32("stride", &srcStride)
+ && srcFormat->findInt32("slice-height", &srcSliceHeight)
+ && srcFormat->findInt32("width", &srcWidth)
+ && srcFormat->findInt32("height", & srcHeight)
+ && srcFormat->findInt32("color-format", &srcColorFormat));
+ CHECK(destFormat->findInt32("stride", &destStride)
+ && destFormat->findInt32("slice-height", &destSliceHeight)
+ && destFormat->findInt32("width", &destWidth)
+ && destFormat->findInt32("height", & destHeight)
+ && destFormat->findInt32("color-format", &destColorFormat));
+
+ CHECK(srcWidth <= destStride && srcHeight <= destSliceHeight);
+
+ convertYUV420spToARGB(
+ srcBuffer->data(),
+ srcBuffer->data() + srcStride * srcSliceHeight,
+ srcWidth,
+ srcHeight,
+ destBuffer->data());
+
+ // copy timestamp
+ int64_t timeUs;
+ CHECK(srcBuffer->meta()->findInt64("timeUs", &timeUs));
+ destBuffer->meta()->setInt64("timeUs", timeUs);
+
+ if (kTestFlush && inputFramesSinceFlush >= kFlushAfterFrames) {
+ inputFramesSinceFlush = 0;
+
+ // check that queueing a buffer that was dequeued before flush
+ // fails with expected error EACCES
+ filterState->mCodec->flush();
+
+ err = filterState->mCodec->queueInputBuffer(
+ filterIndex, 0 /* offset */, destBuffer->size(),
+ timeUs, frame.flags);
+
+ if (err == OK) {
+ ALOGE("FAIL: queue after flush returned OK");
+ } else if (err != -EACCES) {
+ ALOGE("queueInputBuffer after flush returned %d, "
+ "expected -EACCES (-13)", err);
+ }
+ } else {
+ err = filterState->mCodec->queueInputBuffer(
+ filterIndex, 0 /* offset */, destBuffer->size(),
+ timeUs, frame.flags);
+ CHECK(err == OK);
+
+ err = vidState->mCodec->releaseOutputBuffer(outIndex);
+ CHECK(err == OK);
+ }
+}
+
+size_t outputFramesSinceFlush = 0;
+void tryDrainOutputBuffer(
+ CodecState *filterState,
+ const sp<Surface> &surface, bool renderSurface,
+ bool useTimestamp, int64_t *startTimeRender) {
+ size_t index;
+ size_t offset;
+ size_t size;
+ int64_t presentationTimeUs;
+ uint32_t flags;
+ status_t err = filterState->mCodec->dequeueOutputBuffer(
+ &index, &offset, &size, &presentationTimeUs, &flags,
+ kTimeout);
+
+ if (err != OK) {
+ return;
+ }
+
+ ++outputFramesSinceFlush;
+
+ if (kTestFlush && outputFramesSinceFlush >= kFlushAfterFrames) {
+ filterState->mCodec->flush();
+ }
+
+ if (surface == NULL || !renderSurface) {
+ err = filterState->mCodec->releaseOutputBuffer(index);
+ } else if (useTimestamp) {
+ if (*startTimeRender == -1) {
+ // begin rendering 2 vsyncs after first decode
+ *startTimeRender = systemTime(SYSTEM_TIME_MONOTONIC)
+ + 33000000 - (presentationTimeUs * 1000);
+ }
+ presentationTimeUs =
+ (presentationTimeUs * 1000) + *startTimeRender;
+ err = filterState->mCodec->renderOutputBufferAndRelease(
+ index, presentationTimeUs);
+ } else {
+ err = filterState->mCodec->renderOutputBufferAndRelease(index);
+ }
+
+ if (kTestFlush && outputFramesSinceFlush >= kFlushAfterFrames) {
+ outputFramesSinceFlush = 0;
+
+ // releasing the buffer dequeued before flush should cause an error
+ // if so, the frame will also be skipped in output stream
+ if (err == OK) {
+ ALOGE("FAIL: release after flush returned OK");
+ } else if (err != -EACCES) {
+ ALOGE("releaseOutputBuffer after flush returned %d, "
+ "expected -EACCES (-13)", err);
+ }
+ } else {
+ CHECK(err == OK);
+ }
+
+ if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+ ALOGV("reached EOS on output.");
+ filterState->mSawOutputEOS = true;
+ }
+}
+
+static int decode(
+ const sp<ALooper> &looper,
+ const char *path,
+ const sp<Surface> &surface,
+ bool renderSurface,
+ bool useTimestamp,
+ FilterType filterType) {
+
+ static int64_t kTimeout = 500ll;
+
+ sp<NuMediaExtractor> extractor = new NuMediaExtractor;
+ if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
+ fprintf(stderr, "unable to instantiate extractor.\n");
+ return 1;
+ }
+
+ KeyedVector<size_t, CodecState> stateByTrack;
+
+ CodecState *vidState = NULL;
+ for (size_t i = 0; i < extractor->countTracks(); ++i) {
+ sp<AMessage> format;
+ status_t err = extractor->getTrackFormat(i, &format);
+ CHECK(err == OK);
+
+ AString mime;
+ CHECK(format->findString("mime", &mime));
+ bool isVideo = !strncasecmp(mime.c_str(), "video/", 6);
+ if (!isVideo) {
+ continue;
+ }
+
+ ALOGV("selecting track %zu", i);
+
+ err = extractor->selectTrack(i);
+ CHECK(err == OK);
+
+ CodecState *state =
+ &stateByTrack.editValueAt(stateByTrack.add(i, CodecState()));
+
+ vidState = state;
+
+ state->mNumBuffersDecoded = 0;
+
+ state->mCodec = MediaCodec::CreateByType(
+ looper, mime.c_str(), false /* encoder */);
+
+ CHECK(state->mCodec != NULL);
+
+ err = state->mCodec->configure(
+ format, NULL /* surface */, NULL /* crypto */, 0 /* flags */);
+
+ CHECK(err == OK);
+
+ state->mSignalledInputEOS = false;
+ state->mSawOutputEOS = false;
+
+ break;
+ }
+ CHECK(!stateByTrack.isEmpty());
+ CHECK(vidState != NULL);
+ sp<AMessage> vidFormat;
+ vidState->mCodec->getOutputFormat(&vidFormat);
+
+ // set filter to use ARGB8888
+ vidFormat->setInt32("color-format", OMX_COLOR_Format32bitARGB8888);
+ // set app cache directory path
+ vidFormat->setString("cacheDir", "/system/bin");
+
+ // create RenderScript context for RSFilters
+ RSC::sp<RSC::RS> context = new RSC::RS();
+ context->init("/system/bin");
+
+ sp<RenderScriptWrapper::RSFilterCallback> rsFilter;
+
+ // create renderscript wrapper for RSFilters
+ sp<RenderScriptWrapper> rsWrapper = new RenderScriptWrapper;
+ rsWrapper->mContext = context.get();
+
+ CodecState *filterState = new CodecState();
+ filterState->mNumBuffersDecoded = 0;
+
+ sp<AMessage> params = new AMessage();
+
+ switch (filterType) {
+ case FILTERTYPE_ZERO:
+ {
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.zerofilter");
+ params->setInt32("invert", kInvert);
+ break;
+ }
+ case FILTERTYPE_INTRINSIC_BLUR:
+ {
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.intrinsicblur");
+ params->setFloat("blur-radius", kBlurRadius);
+ break;
+ }
+ case FILTERTYPE_SATURATION:
+ {
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.saturation");
+ params->setFloat("saturation", kSaturation);
+ break;
+ }
+ case FILTERTYPE_RS_SATURATION:
+ {
+ SaturationRSFilter *satFilter = new SaturationRSFilter;
+ satFilter->init(context);
+ rsFilter = satFilter;
+ rsWrapper->mCallback = rsFilter;
+ vidFormat->setObject("rs-wrapper", rsWrapper);
+
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.RenderScript");
+ break;
+ }
+ case FILTERTYPE_RS_NIGHT_VISION:
+ {
+ NightVisionRSFilter *nightVisionFilter = new NightVisionRSFilter;
+ nightVisionFilter->init(context);
+ rsFilter = nightVisionFilter;
+ rsWrapper->mCallback = rsFilter;
+ vidFormat->setObject("rs-wrapper", rsWrapper);
+
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.RenderScript");
+ break;
+ }
+ case FILTERTYPE_RS_ARGB_TO_RGBA:
+ {
+ ARGBToRGBARSFilter *argbToRgbaFilter = new ARGBToRGBARSFilter;
+ argbToRgbaFilter->init(context);
+ rsFilter = argbToRgbaFilter;
+ rsWrapper->mCallback = rsFilter;
+ vidFormat->setObject("rs-wrapper", rsWrapper);
+
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.RenderScript");
+ break;
+ }
+ default:
+ {
+ LOG_ALWAYS_FATAL("mediacodec.cpp error: unrecognized FilterType");
+ break;
+ }
+ }
+ CHECK(filterState->mCodec != NULL);
+
+ status_t err = filterState->mCodec->configure(
+ vidFormat /* format */, surface, NULL /* crypto */, 0 /* flags */);
+ CHECK(err == OK);
+
+ filterState->mSignalledInputEOS = false;
+ filterState->mSawOutputEOS = false;
+
+ int64_t startTimeUs = ALooper::GetNowUs();
+ int64_t startTimeRender = -1;
+
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ sp<MediaCodec> codec = state->mCodec;
+
+ CHECK_EQ((status_t)OK, codec->start());
+
+ CHECK_EQ((status_t)OK, codec->getInputBuffers(&state->mInBuffers));
+ CHECK_EQ((status_t)OK, codec->getOutputBuffers(&state->mOutBuffers));
+
+ ALOGV("got %zu input and %zu output buffers",
+ state->mInBuffers.size(), state->mOutBuffers.size());
+ }
+
+ CHECK_EQ((status_t)OK, filterState->mCodec->setParameters(params));
+
+ if (kTestFlush) {
+ status_t flushErr = filterState->mCodec->flush();
+ if (flushErr == OK) {
+ ALOGE("FAIL: Flush before start returned OK");
+ } else {
+ ALOGV("Flush before start returned status %d, usually ENOSYS (-38)",
+ flushErr);
+ }
+ }
+
+ CHECK_EQ((status_t)OK, filterState->mCodec->start());
+ CHECK_EQ((status_t)OK, filterState->mCodec->getInputBuffers(
+ &filterState->mInBuffers));
+ CHECK_EQ((status_t)OK, filterState->mCodec->getOutputBuffers(
+ &filterState->mOutBuffers));
+
+ if (kTestFlush) {
+ status_t flushErr = filterState->mCodec->flush();
+ if (flushErr != OK) {
+ ALOGE("FAIL: Flush after start returned %d, expect OK (0)",
+ flushErr);
+ } else {
+ ALOGV("Flush immediately after start OK");
+ }
+ }
+
+ List<DecodedFrame> decodedFrameIndices;
+
+ // loop until decoder reaches EOS
+ bool sawInputEOS = false;
+ bool sawOutputEOSOnAllTracks = false;
+ while (!sawOutputEOSOnAllTracks) {
+ if (!sawInputEOS) {
+ size_t trackIndex;
+ status_t err = extractor->getSampleTrackIndex(&trackIndex);
+
+ if (err != OK) {
+ ALOGV("saw input eos");
+ sawInputEOS = true;
+ } else {
+ CodecState *state = &stateByTrack.editValueFor(trackIndex);
+
+ size_t index;
+ err = state->mCodec->dequeueInputBuffer(&index, kTimeout);
+
+ if (err == OK) {
+ ALOGV("filling input buffer %zu", index);
+
+ const sp<ABuffer> &buffer = state->mInBuffers.itemAt(index);
+
+ err = extractor->readSampleData(buffer);
+ CHECK(err == OK);
+
+ int64_t timeUs;
+ err = extractor->getSampleTime(&timeUs);
+ CHECK(err == OK);
+
+ uint32_t bufferFlags = 0;
+
+ err = state->mCodec->queueInputBuffer(
+ index, 0 /* offset */, buffer->size(),
+ timeUs, bufferFlags);
+
+ CHECK(err == OK);
+
+ extractor->advance();
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+ }
+ } else {
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ if (!state->mSignalledInputEOS) {
+ size_t index;
+ status_t err =
+ state->mCodec->dequeueInputBuffer(&index, kTimeout);
+
+ if (err == OK) {
+ ALOGV("signalling input EOS on track %zu", i);
+
+ err = state->mCodec->queueInputBuffer(
+ index, 0 /* offset */, 0 /* size */,
+ 0ll /* timeUs */, MediaCodec::BUFFER_FLAG_EOS);
+
+ CHECK(err == OK);
+
+ state->mSignalledInputEOS = true;
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+ }
+ }
+ }
+
+ sawOutputEOSOnAllTracks = true;
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ if (state->mSawOutputEOS) {
+ continue;
+ } else {
+ sawOutputEOSOnAllTracks = false;
+ }
+
+ DecodedFrame frame;
+ status_t err = state->mCodec->dequeueOutputBuffer(
+ &frame.index, &frame.offset, &frame.size,
+ &frame.presentationTimeUs, &frame.flags, kTimeout);
+
+ if (err == OK) {
+ ALOGV("draining decoded buffer %zu, time = %lld us",
+ frame.index, frame.presentationTimeUs);
+
+ ++(state->mNumBuffersDecoded);
+
+ decodedFrameIndices.push_back(frame);
+
+ if (frame.flags & MediaCodec::BUFFER_FLAG_EOS) {
+ ALOGV("reached EOS on decoder output.");
+ state->mSawOutputEOS = true;
+ }
+
+ } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
+ ALOGV("INFO_OUTPUT_BUFFERS_CHANGED");
+ CHECK_EQ((status_t)OK, state->mCodec->getOutputBuffers(
+ &state->mOutBuffers));
+
+ ALOGV("got %zu output buffers", state->mOutBuffers.size());
+ } else if (err == INFO_FORMAT_CHANGED) {
+ sp<AMessage> format;
+ CHECK_EQ((status_t)OK, state->mCodec->getOutputFormat(&format));
+
+ ALOGV("INFO_FORMAT_CHANGED: %s",
+ format->debugString().c_str());
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+
+ tryCopyDecodedBuffer(&decodedFrameIndices, filterState, vidState);
+
+ tryDrainOutputBuffer(
+ filterState, surface, renderSurface,
+ useTimestamp, &startTimeRender);
+ }
+ }
+
+ // after EOS on decoder, let filter reach EOS
+ while (!filterState->mSawOutputEOS) {
+ tryCopyDecodedBuffer(&decodedFrameIndices, filterState, vidState);
+
+ tryDrainOutputBuffer(
+ filterState, surface, renderSurface,
+ useTimestamp, &startTimeRender);
+ }
+
+ int64_t elapsedTimeUs = ALooper::GetNowUs() - startTimeUs;
+
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ CHECK_EQ((status_t)OK, state->mCodec->release());
+
+ printf("track %zu: %" PRId64 " frames decoded and filtered, "
+ "%.2f fps.\n", i, state->mNumBuffersDecoded,
+ state->mNumBuffersDecoded * 1E6 / elapsedTimeUs);
+ }
+
+ return 0;
+}
+
+} // namespace android
+
+int main(int argc, char **argv) {
+ using namespace android;
+
+ const char *me = argv[0];
+
+ bool useSurface = false;
+ bool renderSurface = false;
+ bool useTimestamp = false;
+ FilterType filterType = FILTERTYPE_ZERO;
+
+ int res;
+ while ((res = getopt(argc, argv, "bcnrszTRSh")) >= 0) {
+ switch (res) {
+ case 'b':
+ {
+ filterType = FILTERTYPE_INTRINSIC_BLUR;
+ break;
+ }
+ case 'c':
+ {
+ filterType = FILTERTYPE_RS_ARGB_TO_RGBA;
+ break;
+ }
+ case 'n':
+ {
+ filterType = FILTERTYPE_RS_NIGHT_VISION;
+ break;
+ }
+ case 'r':
+ {
+ filterType = FILTERTYPE_RS_SATURATION;
+ break;
+ }
+ case 's':
+ {
+ filterType = FILTERTYPE_SATURATION;
+ break;
+ }
+ case 'z':
+ {
+ filterType = FILTERTYPE_ZERO;
+ break;
+ }
+ case 'T':
+ {
+ useTimestamp = true;
+ }
+ // fall through
+ case 'R':
+ {
+ renderSurface = true;
+ }
+ // fall through
+ case 'S':
+ {
+ useSurface = true;
+ break;
+ }
+ case '?':
+ case 'h':
+ default:
+ {
+ usage(me);
+ break;
+ }
+ }
+ }
+
+ argc -= optind;
+ argv += optind;
+
+ if (argc != 1) {
+ usage(me);
+ }
+
+ ProcessState::self()->startThreadPool();
+
+ DataSource::RegisterDefaultSniffers();
+
+ android::sp<ALooper> looper = new ALooper;
+ looper->start();
+
+ android::sp<SurfaceComposerClient> composerClient;
+ android::sp<SurfaceControl> control;
+ android::sp<Surface> surface;
+
+ if (useSurface) {
+ composerClient = new SurfaceComposerClient;
+ CHECK_EQ((status_t)OK, composerClient->initCheck());
+
+ android::sp<IBinder> display(SurfaceComposerClient::getBuiltInDisplay(
+ ISurfaceComposer::eDisplayIdMain));
+ DisplayInfo info;
+ SurfaceComposerClient::getDisplayInfo(display, &info);
+ ssize_t displayWidth = info.w;
+ ssize_t displayHeight = info.h;
+
+ ALOGV("display is %zd x %zd", displayWidth, displayHeight);
+
+ control = composerClient->createSurface(
+ String8("A Surface"), displayWidth, displayHeight,
+ PIXEL_FORMAT_RGBA_8888, 0);
+
+ 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();
+
+ surface = control->getSurface();
+ CHECK(surface != NULL);
+ }
+
+ decode(looper, argv[0], surface, renderSurface, useTimestamp, filterType);
+
+ if (useSurface) {
+ composerClient->dispose();
+ }
+
+ looper->stop();
+
+ return 0;
+}
diff --git a/cmds/stagefright/muxer.cpp b/cmds/stagefright/muxer.cpp
index f4a33e8..461b56c 100644
--- a/cmds/stagefright/muxer.cpp
+++ b/cmds/stagefright/muxer.cpp
@@ -17,6 +17,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "muxer"
#include <inttypes.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
#include <utils/Log.h>
#include <binder/ProcessState.h>
@@ -72,8 +75,15 @@
ALOGV("input file %s, output file %s", path, outputFileName);
ALOGV("useAudio %d, useVideo %d", useAudio, useVideo);
- sp<MediaMuxer> muxer = new MediaMuxer(outputFileName,
+ int fd = open(outputFileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+
+ if (fd < 0) {
+ ALOGE("couldn't open file");
+ return fd;
+ }
+ sp<MediaMuxer> muxer = new MediaMuxer(fd,
MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+ close(fd);
size_t trackCount = extractor->countTracks();
// Map the extractor's track index to the muxer's track index.
diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index 9f547c7..2ad40bd 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp
@@ -17,6 +17,10 @@
#include "SineSource.h"
#include <inttypes.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+
#include <binder/ProcessState.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioPlayer.h>
@@ -300,7 +304,13 @@
client.interface(), enc_meta, true /* createEncoder */, source,
0, preferSoftwareCodec ? OMXCodec::kPreferSoftwareCodecs : 0);
- sp<MPEG4Writer> writer = new MPEG4Writer(fileName);
+ int fd = open(fileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ if (fd < 0) {
+ fprintf(stderr, "couldn't open file");
+ return 1;
+ }
+ sp<MPEG4Writer> writer = new MPEG4Writer(fd);
+ close(fd);
writer->addSource(encoder);
int64_t start = systemTime();
CHECK_EQ((status_t)OK, writer->start());
diff --git a/cmds/stagefright/sf2.cpp b/cmds/stagefright/sf2.cpp
index 0f729a3..172dc36 100644
--- a/cmds/stagefright/sf2.cpp
+++ b/cmds/stagefright/sf2.cpp
@@ -72,7 +72,7 @@
}
void startAsync() {
- (new AMessage(kWhatStart, id()))->post();
+ (new AMessage(kWhatStart, this))->post();
}
protected:
@@ -100,7 +100,7 @@
if (ctrlc) {
printf("\n");
printStatistics();
- (new AMessage(kWhatStop, id()))->post();
+ (new AMessage(kWhatStop, this))->post();
ctrlc = false;
}
switch (msg->what()) {
@@ -149,7 +149,7 @@
mDecodeLooper->registerHandler(mCodec);
mCodec->setNotificationMessage(
- new AMessage(kWhatCodecNotify, id()));
+ new AMessage(kWhatCodecNotify, this));
sp<AMessage> format = makeFormat(mSource->getFormat());
@@ -168,7 +168,7 @@
mFinalResult = OK;
mSeekState = SEEK_NONE;
- // (new AMessage(kWhatSeek, id()))->post(5000000ll);
+ // (new AMessage(kWhatSeek, this))->post(5000000ll);
break;
}
@@ -225,12 +225,12 @@
printf((what == CodecBase::kWhatEOS) ? "$\n" : "E\n");
printStatistics();
- (new AMessage(kWhatStop, id()))->post();
+ (new AMessage(kWhatStop, this))->post();
} else if (what == CodecBase::kWhatFlushCompleted) {
mSeekState = SEEK_FLUSH_COMPLETED;
mCodec->signalResume();
- (new AMessage(kWhatSeek, id()))->post(5000000ll);
+ (new AMessage(kWhatSeek, this))->post(5000000ll);
} else if (what == CodecBase::kWhatOutputFormatChanged) {
} else if (what == CodecBase::kWhatShutdownCompleted) {
mDecodeLooper->unregisterHandler(mCodec->id());
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 81edcb4..318b56d 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -19,6 +19,8 @@
#include <stdlib.h>
#include <string.h>
#include <sys/time.h>
+#include <sys/types.h>
+#include <sys/stat.h>
//#define LOG_NDEBUG 0
#define LOG_TAG "stagefright"
@@ -506,8 +508,13 @@
sp<MPEG4Writer> writer =
new MPEG4Writer(gWriteMP4Filename.string());
#else
+ int fd = open(gWriteMP4Filename.string(), O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ if (fd < 0) {
+ fprintf(stderr, "couldn't open file");
+ return;
+ }
sp<MPEG2TSWriter> writer =
- new MPEG2TSWriter(gWriteMP4Filename.string());
+ new MPEG2TSWriter(fd);
#endif
// at most one minute.
diff --git a/drm/mediadrm/plugins/clearkey/DrmPlugin.cpp b/drm/mediadrm/plugins/clearkey/DrmPlugin.cpp
index 96fca94..6b8c772 100644
--- a/drm/mediadrm/plugins/clearkey/DrmPlugin.cpp
+++ b/drm/mediadrm/plugins/clearkey/DrmPlugin.cpp
@@ -48,12 +48,13 @@
KeyType keyType,
const KeyedVector<String8, String8>& optionalParameters,
Vector<uint8_t>& request,
- String8& defaultUrl) {
+ String8& defaultUrl,
+ DrmPlugin::KeyRequestType *keyRequestType) {
UNUSED(optionalParameters);
if (keyType != kKeyType_Streaming) {
return android::ERROR_DRM_CANNOT_HANDLE;
}
-
+ *keyRequestType = DrmPlugin::kKeyRequestType_Initial;
sp<Session> session = mSessionLibrary->findSession(scope);
defaultUrl.clear();
return session->getKeyRequest(initData, initDataType, &request);
diff --git a/drm/mediadrm/plugins/clearkey/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/DrmPlugin.h
index 6139f1f..ba4aefe 100644
--- a/drm/mediadrm/plugins/clearkey/DrmPlugin.h
+++ b/drm/mediadrm/plugins/clearkey/DrmPlugin.h
@@ -54,7 +54,8 @@
KeyType keyType,
const KeyedVector<String8, String8>& optionalParameters,
Vector<uint8_t>& request,
- String8& defaultUrl);
+ String8& defaultUrl,
+ DrmPlugin::KeyRequestType *keyRequestType);
virtual status_t provideKeyResponse(
const Vector<uint8_t>& scope,
diff --git a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
index 7eac0a1..9b786c5 100644
--- a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
+++ b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.cpp
@@ -111,7 +111,8 @@
Vector<uint8_t> const &initData,
String8 const &mimeType, KeyType keyType,
KeyedVector<String8, String8> const &optionalParameters,
- Vector<uint8_t> &request, String8 &defaultUrl)
+ Vector<uint8_t> &request, String8 &defaultUrl,
+ KeyRequestType *keyRequestType)
{
Mutex::Autolock lock(mLock);
ALOGD("MockDrmPlugin::getKeyRequest(sessionId=%s, initData=%s, mimeType=%s"
@@ -149,6 +150,7 @@
// Properties used in mock test, set by cts test app returned from mock plugin
// byte[] mock-request -> request
// string mock-default-url -> defaultUrl
+ // string mock-key-request-type -> keyRequestType
index = mByteArrayProperties.indexOfKey(String8("mock-request"));
if (index < 0) {
@@ -165,6 +167,16 @@
} else {
defaultUrl = mStringProperties.valueAt(index);
}
+
+ index = mStringProperties.indexOfKey(String8("mock-keyRequestType"));
+ if (index < 0) {
+ ALOGD("Missing 'mock-keyRequestType' parameter for mock");
+ return BAD_VALUE;
+ } else {
+ *keyRequestType = static_cast<KeyRequestType>(
+ atoi(mStringProperties.valueAt(index).string()));
+ }
+
return OK;
}
diff --git a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.h b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.h
index d1d8058..d0f2ddb 100644
--- a/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.h
+++ b/drm/mediadrm/plugins/mock/MockDrmCryptoPlugin.h
@@ -62,7 +62,8 @@
Vector<uint8_t> const &initData,
String8 const &mimeType, KeyType keyType,
KeyedVector<String8, String8> const &optionalParameters,
- Vector<uint8_t> &request, String8 &defaultUrl);
+ Vector<uint8_t> &request, String8 &defaultUrl,
+ KeyRequestType *keyRequestType);
status_t provideKeyResponse(Vector<uint8_t> const &sessionId,
Vector<uint8_t> const &response,
diff --git a/include/camera/CameraMetadata.h b/include/camera/CameraMetadata.h
index 1254d3c..953d711 100644
--- a/include/camera/CameraMetadata.h
+++ b/include/camera/CameraMetadata.h
@@ -56,7 +56,7 @@
* thread-safety, it simply prevents the camera_metadata_t pointer returned
* here from being accidentally invalidated by CameraMetadata operations.
*/
- const camera_metadata_t* getAndLock();
+ const camera_metadata_t* getAndLock() const;
/**
* Unlock the CameraMetadata for use again. After this unlock, the pointer
@@ -208,7 +208,7 @@
private:
camera_metadata_t *mBuffer;
- bool mLocked;
+ mutable bool mLocked;
/**
* Check if tag has a given type
diff --git a/include/camera/CameraParameters.h b/include/camera/CameraParameters.h
index c6074fc..ba33ffe 100644
--- a/include/camera/CameraParameters.h
+++ b/include/camera/CameraParameters.h
@@ -108,6 +108,9 @@
*/
void getSupportedPreviewFormats(Vector<int>& formats) const;
+ // Returns true if no keys are present
+ bool isEmpty() const;
+
// Parameter keys to communicate between camera application and driver.
// The access (read/write, read only, or write only) is viewed from the
// perspective of applications, not driver.
diff --git a/include/camera/ICameraService.h b/include/camera/ICameraService.h
index f7f06bb..cad275e 100644
--- a/include/camera/ICameraService.h
+++ b/include/camera/ICameraService.h
@@ -25,8 +25,6 @@
class ICamera;
class ICameraClient;
-class IProCameraUser;
-class IProCameraCallbacks;
class ICameraServiceListener;
class ICameraDeviceUser;
class ICameraDeviceCallbacks;
@@ -44,7 +42,6 @@
GET_NUMBER_OF_CAMERAS = IBinder::FIRST_CALL_TRANSACTION,
GET_CAMERA_INFO,
CONNECT,
- CONNECT_PRO,
CONNECT_DEVICE,
ADD_LISTENER,
REMOVE_LISTENER,
@@ -53,6 +50,8 @@
GET_LEGACY_PARAMETERS,
SUPPORTS_CAMERA_API,
CONNECT_LEGACY,
+ SET_TORCH_MODE,
+ NOTIFY_SYSTEM_EVENT,
};
enum {
@@ -66,7 +65,18 @@
enum {
CAMERA_HAL_API_VERSION_UNSPECIFIED = -1
- };
+ };
+
+ /**
+ * Keep up-to-date with declarations in
+ * frameworks/base/services/core/java/com/android/server/camera/CameraService.java
+ *
+ * These event codes are intended to be used with the notifySystemEvent call.
+ */
+ enum {
+ NO_EVENT = 0,
+ USER_SWITCHED,
+ };
public:
DECLARE_META_INTERFACE(CameraService);
@@ -104,13 +114,6 @@
/*out*/
sp<ICamera>& device) = 0;
- virtual status_t connectPro(const sp<IProCameraCallbacks>& cameraCb,
- int cameraId,
- const String16& clientPackageName,
- int clientUid,
- /*out*/
- sp<IProCameraUser>& device) = 0;
-
virtual status_t connectDevice(
const sp<ICameraDeviceCallbacks>& cameraCb,
int cameraId,
@@ -142,6 +145,26 @@
int clientUid,
/*out*/
sp<ICamera>& device) = 0;
+
+ /**
+ * Turn on or off a camera's torch mode. Torch mode will be turned off by
+ * camera service if the lastest client binder that turns it on dies.
+ *
+ * return values:
+ * 0: on a successful operation.
+ * -ENOSYS: the camera device doesn't support this operation. It it returned
+ * if and only if android.flash.into.available is false.
+ * -EBUSY: the camera device is opened.
+ * -EINVAL: camera_id is invalid or clientBinder is NULL when enabling a
+ * torch mode.
+ */
+ virtual status_t setTorchMode(const String16& cameraId, bool enabled,
+ const sp<IBinder>& clientBinder) = 0;
+
+ /**
+ * Notify the camera service of a system event. Should only be called from system_server.
+ */
+ virtual void notifySystemEvent(int eventId, int arg0) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/camera/ICameraServiceListener.h b/include/camera/ICameraServiceListener.h
index 0a0e43a..709ff31 100644
--- a/include/camera/ICameraServiceListener.h
+++ b/include/camera/ICameraServiceListener.h
@@ -66,9 +66,35 @@
STATUS_UNKNOWN = 0xFFFFFFFF,
};
+ /**
+ * The torch mode status of a camera.
+ *
+ * Initial status will be transmitted with onTorchStatusChanged immediately
+ * after this listener is added to the service listener list.
+ *
+ * The enums should be set to values matching
+ * include/hardware/camera_common.h
+ */
+ enum TorchStatus {
+ // The camera's torch mode has become not available to use via
+ // setTorchMode().
+ TORCH_STATUS_NOT_AVAILABLE = TORCH_MODE_STATUS_NOT_AVAILABLE,
+ // The camera's torch mode is off and available to be turned on via
+ // setTorchMode().
+ TORCH_STATUS_AVAILABLE_OFF = TORCH_MODE_STATUS_AVAILABLE_OFF,
+ // The camera's torch mode is on and available to be turned off via
+ // setTorchMode().
+ TORCH_STATUS_AVAILABLE_ON = TORCH_MODE_STATUS_AVAILABLE_ON,
+
+ // Use to initialize variables only
+ TORCH_STATUS_UNKNOWN = 0xFFFFFFFF,
+ };
+
DECLARE_META_INTERFACE(CameraServiceListener);
virtual void onStatusChanged(Status status, int32_t cameraId) = 0;
+
+ virtual void onTorchStatusChanged(TorchStatus status, const String16& cameraId) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/camera/IProCameraCallbacks.h b/include/camera/IProCameraCallbacks.h
deleted file mode 100644
index e8abb89..0000000
--- a/include/camera/IProCameraCallbacks.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (C) 2013 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_IPROCAMERA_CALLBACKS_H
-#define ANDROID_HARDWARE_IPROCAMERA_CALLBACKS_H
-
-#include <utils/RefBase.h>
-#include <binder/IInterface.h>
-#include <binder/Parcel.h>
-#include <binder/IMemory.h>
-#include <utils/Timers.h>
-#include <system/camera.h>
-
-struct camera_metadata;
-
-namespace android {
-
-class IProCameraCallbacks : public IInterface
-{
- /**
- * Keep up-to-date with IProCameraCallbacks.aidl in frameworks/base
- */
-public:
- DECLARE_META_INTERFACE(ProCameraCallbacks);
-
- virtual void notifyCallback(int32_t msgType,
- int32_t ext1,
- int32_t ext2) = 0;
-
- enum LockStatus {
- LOCK_ACQUIRED,
- LOCK_RELEASED,
- LOCK_STOLEN,
- };
-
- virtual void onLockStatusChanged(LockStatus newLockStatus) = 0;
-
- /** Missing by design: implementation is client-side in ProCamera.cpp **/
- // virtual void onBufferReceived(int streamId,
- // const CpuConsumer::LockedBufer& buf);
- virtual void onResultReceived(int32_t requestId,
- camera_metadata* result) = 0;
-};
-
-// ----------------------------------------------------------------------------
-
-class BnProCameraCallbacks : public BnInterface<IProCameraCallbacks>
-{
-public:
- virtual status_t onTransact( uint32_t code,
- const Parcel& data,
- Parcel* reply,
- uint32_t flags = 0);
-};
-
-}; // namespace android
-
-#endif
diff --git a/include/camera/IProCameraUser.h b/include/camera/IProCameraUser.h
deleted file mode 100644
index 2ccc4d2..0000000
--- a/include/camera/IProCameraUser.h
+++ /dev/null
@@ -1,100 +0,0 @@
-/*
- * Copyright (C) 2013 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_IPROCAMERAUSER_H
-#define ANDROID_HARDWARE_IPROCAMERAUSER_H
-
-#include <utils/RefBase.h>
-#include <binder/IInterface.h>
-#include <binder/Parcel.h>
-#include <binder/IMemory.h>
-#include <utils/String8.h>
-#include <camera/IProCameraCallbacks.h>
-
-struct camera_metadata;
-
-namespace android {
-
-class IProCameraUserClient;
-class IGraphicBufferProducer;
-class Surface;
-
-class IProCameraUser: public IInterface
-{
- /**
- * Keep up-to-date with IProCameraUser.aidl in frameworks/base
- */
-public:
- DECLARE_META_INTERFACE(ProCameraUser);
-
- virtual void disconnect() = 0;
-
- // connect to the service, given a callbacks listener
- virtual status_t connect(const sp<IProCameraCallbacks>& callbacks)
- = 0;
-
- /**
- * Locking
- **/
- virtual status_t exclusiveTryLock() = 0;
- virtual status_t exclusiveLock() = 0;
- virtual status_t exclusiveUnlock() = 0;
-
- virtual bool hasExclusiveLock() = 0;
-
- /**
- * Request Handling
- **/
-
- // Note that the callee gets a copy of the metadata.
- virtual int submitRequest(struct camera_metadata* metadata,
- bool streaming = false) = 0;
- virtual status_t cancelRequest(int requestId) = 0;
-
- virtual status_t deleteStream(int streamId) = 0;
- virtual status_t createStream(
- int width, int height, int format,
- const sp<IGraphicBufferProducer>& bufferProducer,
- /*out*/
- int* streamId) = 0;
-
- // Create a request object from a template.
- virtual status_t createDefaultRequest(int templateId,
- /*out*/
- camera_metadata** request)
- = 0;
-
- // Get static camera metadata
- virtual status_t getCameraInfo(int cameraId,
- /*out*/
- camera_metadata** info) = 0;
-
-};
-
-// ----------------------------------------------------------------------------
-
-class BnProCameraUser: public BnInterface<IProCameraUser>
-{
-public:
- virtual status_t onTransact( uint32_t code,
- const Parcel& data,
- Parcel* reply,
- uint32_t flags = 0);
-};
-
-}; // namespace android
-
-#endif
diff --git a/include/camera/ProCamera.h b/include/camera/ProCamera.h
deleted file mode 100644
index e9b687a..0000000
--- a/include/camera/ProCamera.h
+++ /dev/null
@@ -1,319 +0,0 @@
-/*
- * Copyright (C) 2013 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_PRO_CAMERA_H
-#define ANDROID_HARDWARE_PRO_CAMERA_H
-
-#include <utils/Timers.h>
-#include <utils/KeyedVector.h>
-#include <gui/IGraphicBufferProducer.h>
-#include <system/camera.h>
-#include <camera/IProCameraCallbacks.h>
-#include <camera/IProCameraUser.h>
-#include <camera/Camera.h>
-#include <camera/CameraMetadata.h>
-#include <camera/ICameraService.h>
-#include <gui/CpuConsumer.h>
-
-#include <gui/Surface.h>
-
-#include <utils/Condition.h>
-#include <utils/Mutex.h>
-
-#include <camera/CameraBase.h>
-
-struct camera_metadata;
-
-namespace android {
-
-// All callbacks on this class are concurrent
-// (they come from separate threads)
-class ProCameraListener : virtual public RefBase
-{
-public:
- virtual void notify(int32_t msgType, int32_t ext1, int32_t ext2) = 0;
-
- // Lock has been acquired. Write operations now available.
- virtual void onLockAcquired() = 0;
- // Lock has been released with exclusiveUnlock.
- virtual void onLockReleased() = 0;
- // Lock has been stolen by another client.
- virtual void onLockStolen() = 0;
-
- // Lock free.
- virtual void onTriggerNotify(int32_t msgType, int32_t ext1, int32_t ext2)
- = 0;
- // onFrameAvailable and OnResultReceived can come in with any order,
- // use android.sensor.timestamp and LockedBuffer.timestamp to correlate them
-
- /**
- * A new metadata buffer has been received.
- * -- Ownership of request passes on to the callee, free with
- * free_camera_metadata.
- */
- virtual void onResultReceived(int32_t frameId, camera_metadata* result) = 0;
-
- // TODO: make onFrameAvailable pure virtual
-
- // A new frame buffer has been received for this stream.
- // -- This callback only fires for createStreamCpu streams
- // -- A buffer may be obtained by calling cpuConsumer->lockNextBuffer
- // -- Use buf.timestamp to correlate with result's android.sensor.timestamp
- // -- The buffer should be accessed with CpuConsumer::lockNextBuffer
- // and CpuConsumer::unlockBuffer
- virtual void onFrameAvailable(int /*streamId*/,
- const sp<CpuConsumer>& /*cpuConsumer*/) {
- }
-
-};
-
-class ProCamera;
-
-template <>
-struct CameraTraits<ProCamera>
-{
- typedef ProCameraListener TCamListener;
- typedef IProCameraUser TCamUser;
- typedef IProCameraCallbacks TCamCallbacks;
- typedef status_t (ICameraService::*TCamConnectService)(const sp<IProCameraCallbacks>&,
- int, const String16&, int,
- /*out*/
- sp<IProCameraUser>&);
- static TCamConnectService fnConnectService;
-};
-
-
-class ProCamera :
- public CameraBase<ProCamera>,
- public BnProCameraCallbacks
-{
-public:
- /**
- * Connect a shared camera. By default access is restricted to read only
- * (Lock free) operations. To be able to submit custom requests a lock needs
- * to be acquired with exclusive[Try]Lock.
- */
- static sp<ProCamera> connect(int cameraId);
- virtual ~ProCamera();
-
- /**
- * Exclusive Locks:
- * - We may request exclusive access to a camera if no other
- * clients are using the camera. This works as a traditional
- * client, writing/reading any camera state.
- * - An application opening the camera (a regular 'Camera') will
- * always steal away the exclusive lock from a ProCamera,
- * this will call onLockReleased.
- * - onLockAcquired will be called again once it is possible
- * to again exclusively lock the camera.
- *
- */
-
- /**
- * All exclusiveLock/unlock functions are asynchronous. The remote endpoint
- * shall not block while waiting to acquire the lock. Instead the lock
- * notifications will come in asynchronously on the listener.
- */
-
- /**
- * Attempt to acquire the lock instantly (non-blocking)
- * - If this succeeds, you do not need to wait for onLockAcquired
- * but the event will still be fired
- *
- * Returns -EBUSY if already locked. 0 on success.
- */
- status_t exclusiveTryLock();
- // always returns 0. wait for onLockAcquired before lock is acquired.
- status_t exclusiveLock();
- // release a lock if we have one, or cancel the lock request.
- status_t exclusiveUnlock();
-
- // exclusive lock = do whatever we want. no lock = read only.
- bool hasExclusiveLock();
-
- /**
- * < 0 error, >= 0 the request ID. streaming to have the request repeat
- * until cancelled.
- * The request queue is flushed when a lock is released or stolen
- * if not locked will return PERMISSION_DENIED
- */
- int submitRequest(const struct camera_metadata* metadata,
- bool streaming = false);
- // if not locked will return PERMISSION_DENIED, BAD_VALUE if requestId bad
- status_t cancelRequest(int requestId);
-
- /**
- * Ask for a stream to be enabled.
- * Lock free. Service maintains counter of streams.
- */
- status_t requestStream(int streamId);
-// TODO: remove requestStream, its useless.
-
- /**
- * Delete a stream.
- * Lock free.
- *
- * NOTE: As a side effect this cancels ALL streaming requests.
- *
- * Errors: BAD_VALUE if unknown stream ID.
- * PERMISSION_DENIED if the stream wasn't yours
- */
- status_t deleteStream(int streamId);
-
- /**
- * Create a new HW stream, whose sink will be the window.
- * Lock free. Service maintains counter of streams.
- * Errors: -EBUSY if too many streams created
- */
- status_t createStream(int width, int height, int format,
- const sp<Surface>& surface,
- /*out*/
- int* streamId);
-
- /**
- * Create a new HW stream, whose sink will be the SurfaceTexture.
- * Lock free. Service maintains counter of streams.
- * Errors: -EBUSY if too many streams created
- */
- status_t createStream(int width, int height, int format,
- const sp<IGraphicBufferProducer>& bufferProducer,
- /*out*/
- int* streamId);
- status_t createStreamCpu(int width, int height, int format,
- int heapCount,
- /*out*/
- sp<CpuConsumer>* cpuConsumer,
- int* streamId);
- status_t createStreamCpu(int width, int height, int format,
- int heapCount,
- bool synchronousMode,
- /*out*/
- sp<CpuConsumer>* cpuConsumer,
- int* streamId);
-
- // Create a request object from a template.
- status_t createDefaultRequest(int templateId,
- /*out*/
- camera_metadata** request) const;
-
- // Get static camera metadata
- camera_metadata* getCameraInfo(int cameraId);
-
- // Blocks until a frame is available (CPU streams only)
- // - Obtain the frame data by calling CpuConsumer::lockNextBuffer
- // - Release the frame data after use with CpuConsumer::unlockBuffer
- // Return value:
- // - >0 - number of frames available to be locked
- // - <0 - error (refer to error codes)
- // Error codes:
- // -ETIMEDOUT if it took too long to get a frame
- int waitForFrameBuffer(int streamId);
-
- // Blocks until a metadata result is available
- // - Obtain the metadata by calling consumeFrameMetadata()
- // Error codes:
- // -ETIMEDOUT if it took too long to get a frame
- status_t waitForFrameMetadata();
-
- // Get the latest metadata. This is destructive.
- // - Calling this repeatedly will produce empty metadata objects.
- // - Use waitForFrameMetadata to sync until new data is available.
- CameraMetadata consumeFrameMetadata();
-
- // Convenience method to drop frame buffers (CPU streams only)
- // Return values:
- // >=0 - number of frames dropped (up to count)
- // <0 - error code
- // Error codes:
- // BAD_VALUE - invalid streamId or count passed
- int dropFrameBuffer(int streamId, int count);
-
-protected:
- ////////////////////////////////////////////////////////
- // IProCameraCallbacks implementation
- ////////////////////////////////////////////////////////
- virtual void notifyCallback(int32_t msgType,
- int32_t ext,
- int32_t ext2);
-
- virtual void onLockStatusChanged(
- IProCameraCallbacks::LockStatus newLockStatus);
-
- virtual void onResultReceived(int32_t requestId,
- camera_metadata* result);
-private:
- ProCamera(int cameraId);
-
- class ProFrameListener : public CpuConsumer::FrameAvailableListener {
- public:
- ProFrameListener(wp<ProCamera> camera, int streamID) {
- mCamera = camera;
- mStreamId = streamID;
- }
-
- protected:
- virtual void onFrameAvailable(const BufferItem& /* item */) {
- sp<ProCamera> c = mCamera.promote();
- if (c.get() != NULL) {
- c->onFrameAvailable(mStreamId);
- }
- }
-
- private:
- wp<ProCamera> mCamera;
- int mStreamId;
- };
- friend class ProFrameListener;
-
- struct StreamInfo
- {
- StreamInfo(int streamId) {
- this->streamID = streamId;
- cpuStream = false;
- frameReady = 0;
- }
-
- StreamInfo() {
- streamID = -1;
- cpuStream = false;
- }
-
- int streamID;
- bool cpuStream;
- sp<CpuConsumer> cpuConsumer;
- bool synchronousMode;
- sp<ProFrameListener> frameAvailableListener;
- sp<Surface> stc;
- int frameReady;
- };
-
- Condition mWaitCondition;
- Mutex mWaitMutex;
- static const nsecs_t mWaitTimeout = 1000000000; // 1sec
- KeyedVector<int, StreamInfo> mStreams;
- bool mMetadataReady;
- CameraMetadata mLatestMetadata;
-
- void onFrameAvailable(int streamId);
-
- StreamInfo& getStreamInfo(int streamId);
-
- friend class CameraBase;
-};
-
-}; // namespace android
-
-#endif
diff --git a/include/camera/camera2/ICameraDeviceUser.h b/include/camera/camera2/ICameraDeviceUser.h
index 35488bb..e9f1f5a 100644
--- a/include/camera/camera2/ICameraDeviceUser.h
+++ b/include/camera/camera2/ICameraDeviceUser.h
@@ -27,9 +27,9 @@
class ICameraDeviceUserClient;
class IGraphicBufferProducer;
-class Surface;
class CaptureRequest;
class CameraMetadata;
+class OutputConfiguration;
enum {
NO_IN_FLIGHT_REPEATING_FRAMES = -1,
@@ -100,9 +100,8 @@
virtual status_t endConfigure() = 0;
virtual status_t deleteStream(int streamId) = 0;
- virtual status_t createStream(
- int width, int height, int format,
- const sp<IGraphicBufferProducer>& bufferProducer) = 0;
+
+ virtual status_t createStream(const OutputConfiguration& outputConfiguration) = 0;
// Create a request object from a template.
virtual status_t createDefaultRequest(int templateId,
diff --git a/include/camera/camera2/OutputConfiguration.h b/include/camera/camera2/OutputConfiguration.h
new file mode 100644
index 0000000..e6b679f
--- /dev/null
+++ b/include/camera/camera2/OutputConfiguration.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2015 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_CAMERA2_OUTPUTCONFIGURATION_H
+#define ANDROID_HARDWARE_CAMERA2_OUTPUTCONFIGURATION_H
+
+#include <utils/RefBase.h>
+#include <gui/IGraphicBufferProducer.h>
+
+namespace android {
+
+class Surface;
+
+class OutputConfiguration : public virtual RefBase {
+public:
+
+ static const int INVALID_ROTATION;
+ sp<IGraphicBufferProducer> getGraphicBufferProducer() const;
+ int getRotation() const;
+
+ /**
+ * Keep impl up-to-date with OutputConfiguration.java in frameworks/base
+ */
+ status_t writeToParcel(Parcel& parcel) const;
+ // getGraphicBufferProducer will be NULL if error occurred
+ // getRotation will be INVALID_ROTATION if error occurred
+ OutputConfiguration(const Parcel& parcel);
+
+private:
+ sp<IGraphicBufferProducer> mGbp;
+ int mRotation;
+
+ // helper function
+ static String16 readMaybeEmptyString16(const Parcel& parcel);
+};
+}; // namespace android
+
+#endif
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index f70d981..7be2c3e 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -42,8 +42,7 @@
EVENT_MORE_DATA = 0, // Request to read available data from buffer.
// If this event is delivered but the callback handler
// does not want to read the available data, the handler must
- // explicitly
- // ignore the event by setting frameCount to zero.
+ // explicitly ignore the event by setting frameCount to zero.
EVENT_OVERRUN = 1, // Buffer overrun occurred.
EVENT_MARKER = 2, // Record head is at the specified marker position
// (See setMarkerPosition()).
@@ -53,7 +52,7 @@
// voluntary invalidation by mediaserver, or mediaserver crash.
};
- /* Client should declare Buffer on the stack and pass address to obtainBuffer()
+ /* Client should declare a Buffer and pass address to obtainBuffer()
* and releaseBuffer(). See also callback_t for EVENT_MORE_DATA.
*/
@@ -62,20 +61,25 @@
public:
// FIXME use m prefix
size_t frameCount; // number of sample frames corresponding to size;
- // on input it is the number of frames available,
- // on output is the number of frames actually drained
- // (currently ignored but will make the primary field in future)
+ // on input to obtainBuffer() it is the number of frames desired
+ // on output from obtainBuffer() it is the number of available
+ // frames to be read
+ // on input to releaseBuffer() it is currently ignored
size_t size; // input/output in bytes == frameCount * frameSize
- // on output is the number of bytes actually drained
- // FIXME this is redundant with respect to frameCount,
- // and TRANSFER_OBTAIN mode is broken for 8-bit data
- // since we don't define the frame format
+ // on input to obtainBuffer() it is ignored
+ // on output from obtainBuffer() it is the number of available
+ // bytes to be read, which is frameCount * frameSize
+ // on input to releaseBuffer() it is the number of bytes to
+ // release
+ // FIXME This is redundant with respect to frameCount. Consider
+ // removing size and making frameCount the primary field.
union {
void* raw;
short* i16; // signed 16-bit
int8_t* i8; // unsigned 8-bit, offset by 0x80
+ // input to obtainBuffer(): unused, output: pointer to buffer
};
};
@@ -88,8 +92,8 @@
* user: Pointer to context for use by the callback receiver.
* info: Pointer to optional parameter according to event type:
* - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read
- * more bytes than indicated by 'size' field and update 'size' if fewer bytes are
- * consumed.
+ * more bytes than indicated by 'size' field and update 'size' if
+ * fewer bytes are consumed.
* - EVENT_OVERRUN: unused.
* - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames.
* - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames.
@@ -106,6 +110,7 @@
* - BAD_VALUE: unsupported configuration
* frameCount is guaranteed to be non-zero if status is NO_ERROR,
* and is undefined otherwise.
+ * FIXME This API assumes a route, and so should be deprecated.
*/
static status_t getMinFrameCount(size_t* frameCount,
@@ -118,7 +123,7 @@
enum transfer_type {
TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters
TRANSFER_CALLBACK, // callback EVENT_MORE_DATA
- TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer()
+ TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer()
TRANSFER_SYNC, // synchronous read()
};
@@ -144,15 +149,16 @@
* be larger if the requested size is not compatible with current audio HAL
* latency. Zero means to use a default value.
* cbf: Callback function. If not null, this function is called periodically
- * to consume new data and inform of marker, position updates, etc.
+ * to consume new data in TRANSFER_CALLBACK mode
+ * and inform of marker, position updates, etc.
* user: Context for use by the callback receiver.
* notificationFrames: The callback function is called each time notificationFrames PCM
* frames are ready in record track output buffer.
* sessionId: Not yet supported.
* transferType: How data is transferred from AudioRecord.
* flags: See comments on audio_input_flags_t in <system/audio.h>
+ * pAttributes: If not NULL, supersedes inputSource for use case selection.
* threadCanCallJava: Not present in parameter list, and so is fixed at false.
- * pAttributes: if not NULL, supersedes inputSource for use case selection
*/
AudioRecord(audio_source_t inputSource,
@@ -177,6 +183,7 @@
/* Initialize an AudioRecord that was created using the AudioRecord() constructor.
* Don't call set() more than once, or after an AudioRecord() constructor that takes parameters.
+ * set() is not multi-thread safe.
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful intialization
* - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use
@@ -211,7 +218,7 @@
status_t initCheck() const { return mStatus; }
/* Returns this track's estimated latency in milliseconds.
- * This includes the latency due to AudioRecord buffer size,
+ * This includes the latency due to AudioRecord buffer size, resampling if applicable,
* and audio hardware driver.
*/
uint32_t latency() const { return mLatency; }
@@ -243,11 +250,6 @@
*/
uint32_t getSampleRate() const { return mSampleRate; }
- /* Return the notification frame count.
- * This is approximately how often the callback is invoked, for transfer type TRANSFER_CALLBACK.
- */
- size_t notificationFrames() const { return mNotificationFramesAct; }
-
/* Sets marker position. When record reaches the number of frames specified,
* a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
* with marker == 0 cancels marker notification callback.
@@ -309,7 +311,12 @@
* Returned value:
* handle on audio hardware input
*/
- audio_io_handle_t getInput() const;
+// FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp
+ audio_io_handle_t getInput() const __attribute__((__deprecated__))
+ { return getInputPrivate(); }
+private:
+ audio_io_handle_t getInputPrivate() const;
+public:
/* Returns the audio session ID associated with this AudioRecord.
*
@@ -323,7 +330,8 @@
*/
int getSessionId() const { return mSessionId; }
- /* Obtains a buffer of up to "audioBuffer->frameCount" full frames.
+ /* Public API for TRANSFER_OBTAIN mode.
+ * Obtains a buffer of up to "audioBuffer->frameCount" full frames.
* After draining these frames of data, the caller should release them with releaseBuffer().
* If the track buffer is not empty, obtainBuffer() returns as many contiguous
* full frames as are available immediately.
@@ -336,9 +344,6 @@
* or return WOULD_BLOCK depending on the value of the "waitCount"
* parameter.
*
- * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications,
- * which should use read() or callback EVENT_MORE_DATA instead.
- *
* Interpretation of waitCount:
* +n limits wait time to n * WAIT_PERIOD_MS,
* -1 causes an (almost) infinite wait time,
@@ -347,6 +352,8 @@
* Buffer fields
* On entry:
* frameCount number of frames requested
+ * size ignored
+ * raw ignored
* After error return:
* frameCount 0
* size 0
@@ -357,9 +364,7 @@
* raw pointer to the buffer
*/
- /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */
- status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
- __attribute__((__deprecated__));
+ status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
private:
/* If nonContig is non-NULL, it is an output parameter that will be set to the number of
@@ -372,9 +377,15 @@
struct timespec *elapsed = NULL, size_t *nonContig = NULL);
public:
- /* Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. */
- // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
- void releaseBuffer(Buffer* audioBuffer);
+ /* Public API for TRANSFER_OBTAIN mode.
+ * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill.
+ *
+ * Buffer fields:
+ * frameCount currently ignored but recommend to set to actual number of frames consumed
+ * size actual number of bytes consumed, must be multiple of frameSize
+ * raw ignored
+ */
+ void releaseBuffer(const Buffer* audioBuffer);
/* As a convenience we provide a read() interface to the audio buffer.
* Input parameter 'size' is in byte units.
@@ -386,8 +397,11 @@
* WOULD_BLOCK when obtainBuffer() returns same, or
* AudioRecord was stopped during the read
* or any other error code returned by IAudioRecord::start() or restoreRecord_l().
+ * Default behavior is to only return when all data has been transferred. Set 'blocking' to
+ * false for the method to return immediately without waiting to try multiple times to read
+ * the full content of the buffer.
*/
- ssize_t read(void* buffer, size_t size);
+ ssize_t read(void* buffer, size_t size, bool blocking = true);
/* Return the number of input frames lost in the audio driver since the last call of this
* function. Audio driver is expected to reset the value to 0 and restart counting upon
@@ -416,6 +430,7 @@
void pause(); // suspend thread from execution at next loop boundary
void resume(); // allow thread to execute, if not requested to exit
+ void wake(); // wake to handle changed notification conditions.
private:
void pauseInternal(nsecs_t ns = 0LL);
@@ -430,7 +445,9 @@
bool mPaused; // whether thread is requested to pause at next loop entry
bool mPausedInt; // whether thread internally requests pause
nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored
- bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request
+ bool mIgnoreNextPausedInt; // skip any internal pause and go immediately
+ // to processAudioBuffer() as state may have changed
+ // since pause time calculated.
};
// body of AudioRecordThread::threadLoop()
@@ -458,7 +475,7 @@
bool mActive;
// for client callback handler
- callback_t mCbf; // callback handler for events, or NULL
+ callback_t mCbf; // callback handler for events, or NULL
void* mUserData;
// for notification APIs
@@ -475,10 +492,10 @@
bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer()
uint32_t mObservedSequence; // last observed value of mSequence
- uint32_t mMarkerPosition; // in wrapping (overflow) frame units
+ uint32_t mMarkerPosition; // in wrapping (overflow) frame units
bool mMarkerReached;
- uint32_t mNewPosition; // in frames
- uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS
+ uint32_t mNewPosition; // in frames
+ uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS
status_t mStatus;
diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h
index 97847a0..b705efa 100644
--- a/include/media/AudioResamplerPublic.h
+++ b/include/media/AudioResamplerPublic.h
@@ -26,4 +26,17 @@
// TODO: replace with an API
#define AUDIO_RESAMPLER_DOWN_RATIO_MAX 256
+// Returns the source frames needed to resample to destination frames. This is not a precise
+// value and depends on the resampler (and possibly how it handles rounding internally).
+// Nevertheless, this should be an upper bound on the requirements of the resampler.
+// If srcSampleRate and dstSampleRate are equal, then it returns destination frames, which
+// may not be true if the resampler is asynchronous.
+static inline size_t sourceFramesNeeded(
+ uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate) {
+ // +1 for rounding - always do this even if matched ratio (resampler may use phases not ratio)
+ // +1 for additional sample needed for interpolation
+ return srcSampleRate == dstSampleRate ? dstFramesRequired :
+ size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1);
+}
+
#endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 843a354..f5db1bb 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -98,10 +98,13 @@
// Returned samplingRate and frameCount output values are guaranteed
// to be non-zero if status == NO_ERROR
+ // FIXME This API assumes a route, and so should be deprecated.
static status_t getOutputSamplingRate(uint32_t* samplingRate,
audio_stream_type_t stream);
+ // FIXME This API assumes a route, and so should be deprecated.
static status_t getOutputFrameCount(size_t* frameCount,
audio_stream_type_t stream);
+ // FIXME This API assumes a route, and so should be deprecated.
static status_t getOutputLatency(uint32_t* latency,
audio_stream_type_t stream);
static status_t getSamplingRate(audio_io_handle_t output,
@@ -110,19 +113,20 @@
// audio_stream->get_buffer_size()/audio_stream_out_frame_size()
static status_t getFrameCount(audio_io_handle_t output,
size_t* frameCount);
- // returns the audio output stream latency in ms. Corresponds to
+ // returns the audio output latency in ms. Corresponds to
// audio_stream_out->get_latency()
static status_t getLatency(audio_io_handle_t output,
uint32_t* latency);
// return status NO_ERROR implies *buffSize > 0
+ // FIXME This API assumes a route, and so should deprecated.
static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
audio_channel_mask_t channelMask, size_t* buffSize);
static status_t setVoiceVolume(float volume);
// return the number of audio frames written by AudioFlinger to audio HAL and
- // audio dsp to DAC since the specified output I/O handle has exited standby.
+ // audio dsp to DAC since the specified output has exited standby.
// returned status (from utils/Errors.h) can be:
// - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
// - INVALID_OPERATION: Not supported on current hardware platform
@@ -201,7 +205,7 @@
// IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
//
static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state,
- const char *device_address);
+ const char *device_address, const char *device_name);
static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address);
static status_t setPhoneState(audio_mode_t state);
@@ -342,7 +346,8 @@
};
- static void setAudioPortCallback(sp<AudioPortCallback> callBack);
+ static status_t addAudioPortCallback(const sp<AudioPortCallback>& callBack);
+ static status_t removeAudioPortCallback(const sp<AudioPortCallback>& callBack);
private:
@@ -369,12 +374,19 @@
AudioPolicyServiceClient() {
}
+ status_t addAudioPortCallback(const sp<AudioPortCallback>& callBack);
+ status_t removeAudioPortCallback(const sp<AudioPortCallback>& callBack);
+
// DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
// IAudioPolicyServiceClient
virtual void onAudioPortListUpdate();
virtual void onAudioPatchListUpdate();
+
+ private:
+ Mutex mLock;
+ Vector <sp <AudioPortCallback> > mAudioPortCallbacks;
};
static sp<AudioFlingerClient> gAudioFlingerClient;
@@ -386,7 +398,6 @@
static Mutex gLockCache; // protects gOutputs, gPrevInSamplingRate, gPrevInFormat,
// gPrevInChannelMask and gInBuffSize
static Mutex gLockAPS; // protects gAudioPolicyService and gAudioPolicyServiceClient
- static Mutex gLockAPC; // protects gAudioPortCallback
static sp<IAudioFlinger> gAudioFlinger;
static audio_error_callback gAudioErrorCallback;
@@ -401,8 +412,6 @@
// list of output descriptors containing cached parameters
// (sampling rate, framecount, channel count...)
static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;
-
- static sp<AudioPortCallback> gAudioPortCallback;
};
}; // namespace android
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index fd51b8f..d9b7057 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -63,7 +63,7 @@
// See AudioTimestamp for the information included with event.
};
- /* Client should declare Buffer on the stack and pass address to obtainBuffer()
+ /* Client should declare a Buffer and pass the address to obtainBuffer()
* and releaseBuffer(). See also callback_t for EVENT_MORE_DATA.
*/
@@ -72,22 +72,26 @@
public:
// FIXME use m prefix
size_t frameCount; // number of sample frames corresponding to size;
- // on input it is the number of frames desired,
- // on output is the number of frames actually filled
- // (currently ignored, but will make the primary field in future)
+ // on input to obtainBuffer() it is the number of frames desired,
+ // on output from obtainBuffer() it is the number of available
+ // [empty slots for] frames to be filled
+ // on input to releaseBuffer() it is currently ignored
size_t size; // input/output in bytes == frameCount * frameSize
- // on input it is unused
- // on output is the number of bytes actually filled
- // FIXME this is redundant with respect to frameCount,
- // and TRANSFER_OBTAIN mode is broken for 8-bit data
- // since we don't define the frame format
+ // on input to obtainBuffer() it is ignored
+ // on output from obtainBuffer() it is the number of available
+ // [empty slots for] bytes to be filled,
+ // which is frameCount * frameSize
+ // on input to releaseBuffer() it is the number of bytes to
+ // release
+ // FIXME This is redundant with respect to frameCount. Consider
+ // removing size and making frameCount the primary field.
union {
void* raw;
short* i16; // signed 16-bit
int8_t* i8; // unsigned 8-bit, offset by 0x80
- }; // input: unused, output: pointer to buffer
+ }; // input to obtainBuffer(): unused, output: pointer to buffer
};
/* As a convenience, if a callback is supplied, a handler thread
@@ -121,6 +125,7 @@
* - BAD_VALUE: unsupported configuration
* frameCount is guaranteed to be non-zero if status is NO_ERROR,
* and is undefined otherwise.
+ * FIXME This API assumes a route, and so should be deprecated.
*/
static status_t getMinFrameCount(size_t* frameCount,
@@ -132,7 +137,7 @@
enum transfer_type {
TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters
TRANSFER_CALLBACK, // callback EVENT_MORE_DATA
- TRANSFER_OBTAIN, // FIXME deprecated: call obtainBuffer() and releaseBuffer()
+ TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer()
TRANSFER_SYNC, // synchronous write()
TRANSFER_SHARED, // shared memory
};
@@ -145,18 +150,15 @@
/* Creates an AudioTrack object and registers it with AudioFlinger.
* Once created, the track needs to be started before it can be used.
* Unspecified values are set to appropriate default values.
- * With this constructor, the track is configured for streaming mode.
- * Data to be rendered is supplied by write() or by the callback EVENT_MORE_DATA.
- * Intermixing a combination of write() and non-ignored EVENT_MORE_DATA is not allowed.
*
* Parameters:
*
* streamType: Select the type of audio stream this track is attached to
* (e.g. AUDIO_STREAM_MUSIC).
* sampleRate: Data source sampling rate in Hz.
- * format: Audio format. For mixed tracks, any PCM format supported by server is OK
- * or AUDIO_FORMAT_PCM_8_BIT which is handled on client side. For direct
- * and offloaded tracks, the possible format(s) depends on the output sink.
+ * format: Audio format. For mixed tracks, any PCM format supported by server is OK.
+ * For direct and offloaded tracks, the possible format(s) depends on the
+ * output sink.
* channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true.
* frameCount: Minimum size of track PCM buffer in frames. This defines the
* application's contribution to the
@@ -165,20 +167,28 @@
* configuration. Zero means to use a default value.
* flags: See comments on audio_output_flags_t in <system/audio.h>.
* cbf: Callback function. If not null, this function is called periodically
- * to provide new data and inform of marker, position updates, etc.
+ * to provide new data in TRANSFER_CALLBACK mode
+ * and inform of marker, position updates, etc.
* user: Context for use by the callback receiver.
* notificationFrames: The callback function is called each time notificationFrames PCM
* frames have been consumed from track input buffer.
* This is expressed in units of frames at the initial source sample rate.
* sessionId: Specific session ID, or zero to use default.
* transferType: How data is transferred to AudioTrack.
+ * offloadInfo: If not NULL, provides offload parameters for
+ * AudioSystem::getOutputForAttr().
+ * uid: User ID of the app which initially requested this AudioTrack
+ * for power management tracking, or -1 for current user ID.
+ * pid: Process ID of the app which initially requested this AudioTrack
+ * for power management tracking, or -1 for current process ID.
+ * pAttributes: If not NULL, supersedes streamType for use case selection.
* threadCanCallJava: Not present in parameter list, and so is fixed at false.
*/
AudioTrack( audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- audio_channel_mask_t,
+ audio_channel_mask_t channelMask,
size_t frameCount = 0,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
callback_t cbf = NULL,
@@ -193,9 +203,10 @@
/* Creates an audio track and registers it with AudioFlinger.
* With this constructor, the track is configured for static buffer mode.
- * The format must not be 8-bit linear PCM.
* Data to be rendered is passed in a shared memory buffer
- * identified by the argument sharedBuffer, which must be non-0.
+ * identified by the argument sharedBuffer, which should be non-0.
+ * If sharedBuffer is zero, this constructor is equivalent to the previous constructor
+ * but without the ability to specify a non-zero value for the frameCount parameter.
* The memory should be initialized to the desired data before calling start().
* The write() method is not supported in this case.
* It is recommended to pass a callback function to be notified of playback end by an
@@ -227,6 +238,7 @@
/* Initialize an AudioTrack that was created using the AudioTrack() constructor.
* Don't call set() more than once, or after the AudioTrack() constructors that take parameters.
+ * set() is not multi-thread safe.
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful initialization
* - INVALID_OPERATION: AudioTrack is already initialized
@@ -461,7 +473,9 @@
* handle on audio hardware output, or AUDIO_IO_HANDLE_NONE if the
* track needed to be re-created but that failed
*/
+private:
audio_io_handle_t getOutput() const;
+public:
/* Returns the unique session ID associated with this track.
*
@@ -487,10 +501,18 @@
*/
status_t attachAuxEffect(int effectId);
- /* Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames.
+ /* Public API for TRANSFER_OBTAIN mode.
+ * Obtains a buffer of up to "audioBuffer->frameCount" empty slots for frames.
* After filling these slots with data, the caller should release them with releaseBuffer().
* If the track buffer is not full, obtainBuffer() returns as many contiguous
* [empty slots for] frames as are available immediately.
+ *
+ * If nonContig is non-NULL, it is an output parameter that will be set to the number of
+ * additional non-contiguous frames that are predicted to be available immediately,
+ * if the client were to release the first frames and then call obtainBuffer() again.
+ * This value is only a prediction, and needs to be confirmed.
+ * It will be set to zero for an error return.
+ *
* If the track buffer is full and track is stopped, obtainBuffer() returns WOULD_BLOCK
* regardless of the value of waitCount.
* If the track buffer is full and track is not stopped, obtainBuffer() blocks with a
@@ -499,10 +521,6 @@
* is exhausted, at which point obtainBuffer() will either block
* or return WOULD_BLOCK depending on the value of the "waitCount"
* parameter.
- * Each sample is 16-bit signed PCM.
- *
- * obtainBuffer() and releaseBuffer() are deprecated for direct use by applications,
- * which should use write() or callback EVENT_MORE_DATA instead.
*
* Interpretation of waitCount:
* +n limits wait time to n * WAIT_PERIOD_MS,
@@ -511,24 +529,27 @@
*
* Buffer fields
* On entry:
- * frameCount number of frames requested
+ * frameCount number of [empty slots for] frames requested
+ * size ignored
+ * raw ignored
* After error return:
* frameCount 0
* size 0
* raw undefined
* After successful return:
- * frameCount actual number of frames available, <= number requested
+ * frameCount actual number of [empty slots for] frames available, <= number requested
* size actual number of bytes available
* raw pointer to the buffer
*/
-
- /* FIXME Deprecated public API for TRANSFER_OBTAIN mode */
- status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
- __attribute__((__deprecated__));
+ status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount,
+ size_t *nonContig = NULL);
private:
/* If nonContig is non-NULL, it is an output parameter that will be set to the number of
- * additional non-contiguous frames that are available immediately.
+ * additional non-contiguous frames that are predicted to be available immediately,
+ * if the client were to release the first frames and then call obtainBuffer() again.
+ * This value is only a prediction, and needs to be confirmed.
+ * It will be set to zero for an error return.
* FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(),
* in case the requested amount of frames is in two or more non-contiguous regions.
* FIXME requested and elapsed are both relative times. Consider changing to absolute time.
@@ -537,9 +558,15 @@
struct timespec *elapsed = NULL, size_t *nonContig = NULL);
public:
- /* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */
- // FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
- void releaseBuffer(Buffer* audioBuffer);
+ /* Public API for TRANSFER_OBTAIN mode.
+ * Release a filled buffer of frames for AudioFlinger to process.
+ *
+ * Buffer fields:
+ * frameCount currently ignored but recommend to set to actual number of frames filled
+ * size actual number of bytes filled, must be multiple of frameSize
+ * raw ignored
+ */
+ void releaseBuffer(const Buffer* audioBuffer);
/* As a convenience we provide a write() interface to the audio buffer.
* Input parameter 'size' is in byte units.
@@ -551,7 +578,7 @@
* WOULD_BLOCK when obtainBuffer() returns same, or
* AudioTrack was stopped during the write
* or any other error code returned by IAudioTrack::start() or restoreTrack_l().
- * Default behavior is to only return until all data has been transferred. Set 'blocking' to
+ * Default behavior is to only return when all data has been transferred. Set 'blocking' to
* false for the method to return immediately without waiting to try multiple times to write
* the full content of the buffer.
*/
@@ -559,6 +586,7 @@
/*
* Dumps the state of an audio track.
+ * Not a general-purpose API; intended only for use by media player service to dump its tracks.
*/
status_t dump(int fd, const Vector<String16>& args) const;
@@ -600,8 +628,6 @@
AudioTrack(const AudioTrack& other);
AudioTrack& operator = (const AudioTrack& other);
- void setAttributesFromStreamType(audio_stream_type_t streamType);
-
/* a small internal class to handle the callback */
class AudioTrackThread : public Thread
{
@@ -614,6 +640,7 @@
void pause(); // suspend thread from execution at next loop boundary
void resume(); // allow thread to execute, if not requested to exit
+ void wake(); // wake to handle changed notification conditions.
private:
void pauseInternal(nsecs_t ns = 0LL);
@@ -628,7 +655,9 @@
bool mPaused; // whether thread is requested to pause at next loop entry
bool mPausedInt; // whether thread internally requests pause
nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored
- bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request
+ bool mIgnoreNextPausedInt; // skip any internal pause and go immediately
+ // to processAudioBuffer() as state may have changed
+ // since pause time calculated.
};
// body of AudioTrackThread::threadLoop()
@@ -641,10 +670,6 @@
static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
nsecs_t processAudioBuffer();
- bool isOffloaded() const;
- bool isDirect() const;
- bool isOffloadedOrDirect() const;
-
// caller must hold lock on mLock for all _l methods
status_t createTrack_l();
@@ -657,6 +682,10 @@
// FIXME enum is faster than strcmp() for parameter 'from'
status_t restoreTrack_l(const char *from);
+ bool isOffloaded() const;
+ bool isDirect() const;
+ bool isOffloadedOrDirect() const;
+
bool isOffloaded_l() const
{ return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
@@ -680,7 +709,7 @@
float mVolume[2];
float mSendLevel;
- mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it.
+ mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it
size_t mFrameCount; // corresponds to current IAudioTrack, value is
// reported back by AudioFlinger to the client
size_t mReqFrameCount; // frame count to request the first or next time
@@ -698,10 +727,7 @@
const audio_offload_info_t* mOffloadInfo;
audio_attributes_t mAttributes;
- // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's
- // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
- size_t mFrameSize; // app-level frame size
- size_t mFrameSizeAF; // AudioFlinger frame size
+ size_t mFrameSize; // frame size in bytes
status_t mStatus;
@@ -732,17 +758,25 @@
bool mRefreshRemaining; // processAudioBuffer() should refresh
// mRemainingFrames and mRetryOnPartialBuffer
+ // used for static track cbf and restoration
+ int32_t mLoopCount; // last setLoop loopCount; zero means disabled
+ uint32_t mLoopStart; // last setLoop loopStart
+ uint32_t mLoopEnd; // last setLoop loopEnd
+ int32_t mLoopCountNotified; // the last loopCount notified by callback.
+ // mLoopCountNotified counts down, matching
+ // the remaining loop count for static track
+ // playback.
+
// These are private to processAudioBuffer(), and are not protected by a lock
uint32_t mRemainingFrames; // number of frames to request in obtainBuffer()
bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer()
uint32_t mObservedSequence; // last observed value of mSequence
- uint32_t mLoopPeriod; // in frames, zero means looping is disabled
-
uint32_t mMarkerPosition; // in wrapping (overflow) frame units
bool mMarkerReached;
uint32_t mNewPosition; // in frames
uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS
+
uint32_t mServer; // in frames, last known mProxy->getPosition()
// which is count of frames consumed by server,
// reset by new IAudioTrack,
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index b1ed7b0..64a3212 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -171,6 +171,8 @@
////////////////////////////////////////////////////////////////////////////////
int EffectIsNullUuid(const effect_uuid_t *pEffectUuid);
+int EffectDumpEffects(int fd);
+
#if __cplusplus
} // extern "C"
#endif
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 31a14f0..f927a80 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -94,6 +94,8 @@
sp<IMemory>& buffers, // return value 0 means it follows cblk
status_t *status) = 0;
+ // FIXME Surprisingly, sampleRate/format/frameCount/latency don't work for input handles
+
/* query the audio hardware state. This state never changes,
* and therefore can be cached.
*/
@@ -142,6 +144,7 @@
virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0;
// retrieve the audio recording buffer size
+ // FIXME This API assumes a route, and so should be deprecated.
virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
audio_channel_mask_t channelMask) const = 0;
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index c98c475..fecc6f1 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -44,7 +44,8 @@
//
virtual status_t setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address) = 0;
+ const char *device_address,
+ const char *device_name) = 0;
virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address) = 0;
virtual status_t setPhoneState(audio_mode_t state) = 0;
diff --git a/include/media/IDrm.h b/include/media/IDrm.h
index affcbd7..9449beb6 100644
--- a/include/media/IDrm.h
+++ b/include/media/IDrm.h
@@ -47,7 +47,8 @@
Vector<uint8_t> const &initData,
String8 const &mimeType, DrmPlugin::KeyType keyType,
KeyedVector<String8, String8> const &optionalParameters,
- Vector<uint8_t> &request, String8 &defaultUrl) = 0;
+ Vector<uint8_t> &request, String8 &defaultUrl,
+ DrmPlugin::KeyRequestType *keyRequestType) = 0;
virtual status_t provideKeyResponse(Vector<uint8_t> const &sessionId,
Vector<uint8_t> const &response,
diff --git a/include/media/IMediaPlayer.h b/include/media/IMediaPlayer.h
index db62cd5..4153c25 100644
--- a/include/media/IMediaPlayer.h
+++ b/include/media/IMediaPlayer.h
@@ -56,6 +56,7 @@
virtual status_t stop() = 0;
virtual status_t pause() = 0;
virtual status_t isPlaying(bool* state) = 0;
+ virtual status_t setPlaybackRate(float rate) = 0;
virtual status_t seekTo(int msec) = 0;
virtual status_t getCurrentPosition(int* msec) = 0;
virtual status_t getDuration(int* msec) = 0;
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index 67b599a..49a3d61 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -49,7 +49,8 @@
virtual sp<IMediaRecorder> createMediaRecorder() = 0;
virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
- virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
+ virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0)
+ = 0;
virtual sp<IOMX> getOMX() = 0;
virtual sp<ICrypto> makeCrypto() = 0;
diff --git a/include/media/IMediaRecorder.h b/include/media/IMediaRecorder.h
index 3e67550..509c06b 100644
--- a/include/media/IMediaRecorder.h
+++ b/include/media/IMediaRecorder.h
@@ -41,7 +41,6 @@
virtual status_t setOutputFormat(int of) = 0;
virtual status_t setVideoEncoder(int ve) = 0;
virtual status_t setAudioEncoder(int ae) = 0;
- virtual status_t setOutputFile(const char* path) = 0;
virtual status_t setOutputFile(int fd, int64_t offset, int64_t length) = 0;
virtual status_t setVideoSize(int width, int height) = 0;
virtual status_t setVideoFrameRate(int frames_per_second) = 0;
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
index 627f23b..6def65b 100644
--- a/include/media/IOMX.h
+++ b/include/media/IOMX.h
@@ -147,6 +147,7 @@
INTERNAL_OPTION_SUSPEND, // data is a bool
INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY, // data is an int64_t
INTERNAL_OPTION_MAX_TIMESTAMP_GAP, // data is int64_t
+ INTERNAL_OPTION_MAX_FPS, // data is float
INTERNAL_OPTION_START_TIME, // data is an int64_t
INTERNAL_OPTION_TIME_LAPSE, // data is an int64_t[2]
};
diff --git a/include/media/IResourceManagerClient.h b/include/media/IResourceManagerClient.h
new file mode 100644
index 0000000..3587aea
--- /dev/null
+++ b/include/media/IResourceManagerClient.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright 2015 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_IRESOURCEMANAGERCLIENT_H
+#define ANDROID_IRESOURCEMANAGERCLIENT_H
+
+#include <utils/RefBase.h>
+#include <binder/IInterface.h>
+#include <binder/Parcel.h>
+
+namespace android {
+
+class IResourceManagerClient: public IInterface
+{
+public:
+ DECLARE_META_INTERFACE(ResourceManagerClient);
+
+ virtual bool reclaimResource() = 0;
+};
+
+// ----------------------------------------------------------------------------
+
+class BnResourceManagerClient: public BnInterface<IResourceManagerClient>
+{
+public:
+ virtual status_t onTransact(uint32_t code,
+ const Parcel &data,
+ Parcel *reply,
+ uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif // ANDROID_IRESOURCEMANAGERCLIENT_H
diff --git a/include/media/IResourceManagerService.h b/include/media/IResourceManagerService.h
new file mode 100644
index 0000000..067392c
--- /dev/null
+++ b/include/media/IResourceManagerService.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright 2015 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_IRESOURCEMANAGERSERVICE_H
+#define ANDROID_IRESOURCEMANAGERSERVICE_H
+
+#include <utils/Errors.h> // for status_t
+#include <utils/KeyedVector.h>
+#include <utils/RefBase.h>
+#include <utils/String8.h>
+#include <binder/IInterface.h>
+#include <binder/Parcel.h>
+
+#include <media/IResourceManagerClient.h>
+#include <media/MediaResource.h>
+#include <media/MediaResourcePolicy.h>
+
+namespace android {
+
+class IResourceManagerService: public IInterface
+{
+public:
+ DECLARE_META_INTERFACE(ResourceManagerService);
+
+ virtual void config(const Vector<MediaResourcePolicy> &policies) = 0;
+
+ virtual void addResource(
+ int pid,
+ int64_t clientId,
+ const sp<IResourceManagerClient> client,
+ const Vector<MediaResource> &resources) = 0;
+
+ virtual void removeResource(int64_t clientId) = 0;
+
+ virtual bool reclaimResource(
+ int callingPid,
+ const Vector<MediaResource> &resources) = 0;
+};
+
+// ----------------------------------------------------------------------------
+
+class BnResourceManagerService: public BnInterface<IResourceManagerService>
+{
+public:
+ virtual status_t onTransact(uint32_t code,
+ const Parcel &data,
+ Parcel *reply,
+ uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif // ANDROID_IRESOURCEMANAGERSERVICE_H
diff --git a/include/media/IStreamSource.h b/include/media/IStreamSource.h
index 677119b..149bd49 100644
--- a/include/media/IStreamSource.h
+++ b/include/media/IStreamSource.h
@@ -81,6 +81,13 @@
// with the next PTS occuring in the stream. The value is of type int64_t.
static const char *const kKeyMediaTimeUs;
+ // Optionally signalled as part of a discontinuity that includes
+ // DISCONTINUITY_TIME. It indicates the media time (in us) of a recent
+ // sample from the same content, and is used as a hint for the parser to
+ // handle PTS wraparound. This is required when a new parser is created
+ // to continue parsing content from the same timeline.
+ static const char *const kKeyRecentMediaTimeUs;
+
virtual void issueCommand(
Command cmd, bool synchronous, const sp<AMessage> &msg = NULL) = 0;
};
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 4a6bf28..d6fe390 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -113,7 +113,19 @@
const audio_offload_info_t *offloadInfo = NULL) = 0;
virtual status_t start() = 0;
- virtual ssize_t write(const void* buffer, size_t size) = 0;
+
+ /* Input parameter |size| is in byte units stored in |buffer|.
+ * Data is copied over and actual number of bytes written (>= 0)
+ * is returned, or no data is copied and a negative status code
+ * is returned (even when |blocking| is true).
+ * When |blocking| is false, AudioSink will immediately return after
+ * part of or full |buffer| is copied over.
+ * When |blocking| is true, AudioSink will wait to copy the entire
+ * buffer, unless an error occurs or the copy operation is
+ * prematurely stopped.
+ */
+ virtual ssize_t write(const void* buffer, size_t size, bool blocking = true) = 0;
+
virtual void stop() = 0;
virtual void flush() = 0;
virtual void pause() = 0;
@@ -156,6 +168,7 @@
virtual status_t stop() = 0;
virtual status_t pause() = 0;
virtual bool isPlaying() = 0;
+ virtual status_t setPlaybackRate(float rate) { return INVALID_OPERATION; }
virtual status_t seekTo(int msec) = 0;
virtual status_t getCurrentPosition(int *msec) = 0;
virtual status_t getDuration(int *msec) = 0;
diff --git a/include/media/MediaRecorderBase.h b/include/media/MediaRecorderBase.h
index d7ac302..f55063e 100644
--- a/include/media/MediaRecorderBase.h
+++ b/include/media/MediaRecorderBase.h
@@ -43,7 +43,6 @@
virtual status_t setCamera(const sp<ICamera>& camera,
const sp<ICameraRecordingProxy>& proxy) = 0;
virtual status_t setPreviewSurface(const sp<IGraphicBufferProducer>& surface) = 0;
- virtual status_t setOutputFile(const char *path) = 0;
virtual status_t setOutputFile(int fd, int64_t offset, int64_t length) = 0;
virtual status_t setOutputFileAuxiliary(int fd) {return INVALID_OPERATION;}
virtual status_t setParameters(const String8& params) = 0;
diff --git a/include/media/MediaResource.h b/include/media/MediaResource.h
new file mode 100644
index 0000000..0b57c84
--- /dev/null
+++ b/include/media/MediaResource.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright 2015 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_MEDIA_RESOURCE_H
+#define ANDROID_MEDIA_RESOURCE_H
+
+#include <binder/Parcel.h>
+#include <utils/String8.h>
+
+namespace android {
+
+extern const char kResourceSecureCodec[];
+extern const char kResourceNonSecureCodec[];
+extern const char kResourceGraphicMemory[];
+
+class MediaResource {
+public:
+ MediaResource();
+ MediaResource(String8 type, uint64_t value);
+ MediaResource(String8 type, String8 subType, uint64_t value);
+
+ void readFromParcel(const Parcel &parcel);
+ void writeToParcel(Parcel *parcel) const;
+
+ String8 toString() const;
+
+ bool operator==(const MediaResource &other) const;
+ bool operator!=(const MediaResource &other) const;
+
+ String8 mType;
+ String8 mSubType;
+ uint64_t mValue;
+};
+
+}; // namespace android
+
+#endif // ANDROID_MEDIA_RESOURCE_H
diff --git a/include/media/MediaResourcePolicy.h b/include/media/MediaResourcePolicy.h
new file mode 100644
index 0000000..1e1c341
--- /dev/null
+++ b/include/media/MediaResourcePolicy.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright 2015 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_MEDIA_RESOURCE_POLICY_H
+#define ANDROID_MEDIA_RESOURCE_POLICY_H
+
+#include <binder/Parcel.h>
+#include <utils/String8.h>
+
+namespace android {
+
+extern const char kPolicySupportsMultipleSecureCodecs[];
+extern const char kPolicySupportsSecureWithNonSecureCodec[];
+
+class MediaResourcePolicy {
+public:
+ MediaResourcePolicy();
+ MediaResourcePolicy(String8 type, uint64_t value);
+
+ void readFromParcel(const Parcel &parcel);
+ void writeToParcel(Parcel *parcel) const;
+
+ String8 toString() const;
+
+ String8 mType;
+ uint64_t mValue;
+};
+
+}; // namespace android
+
+#endif // ANDROID_MEDIA_RESOURCE_POLICY_H
diff --git a/include/media/SingleStateQueue.h b/include/media/SingleStateQueue.h
index 04c5fd0..d423962 100644
--- a/include/media/SingleStateQueue.h
+++ b/include/media/SingleStateQueue.h
@@ -21,6 +21,7 @@
// Non-blocking single-reader / single-writer multi-word atomic load / store
#include <stdint.h>
+#include <cutils/atomic.h>
namespace android {
@@ -31,6 +32,12 @@
class Mutator;
class Observer;
+ enum SSQ_STATUS {
+ SSQ_PENDING, /* = 0 */
+ SSQ_READ,
+ SSQ_DONE,
+ };
+
struct Shared {
// needs to be part of a union so don't define constructor or destructor
@@ -41,28 +48,56 @@
void init() { mAck = 0; mSequence = 0; }
volatile int32_t mAck;
-#if 0
- int mPad[7];
- // cache line boundary
-#endif
volatile int32_t mSequence;
T mValue;
};
class Mutator {
public:
- Mutator(Shared *shared);
- /*virtual*/ ~Mutator() { }
+ Mutator(Shared *shared)
+ : mSequence(0), mShared(shared)
+ {
+ // exactly one of Mutator and Observer must initialize, currently it is Observer
+ // shared->init();
+ }
// push new value onto state queue, overwriting previous value;
// returns a sequence number which can be used with ack()
- int32_t push(const T& value);
+ int32_t push(const T& value)
+ {
+ Shared *shared = mShared;
+ int32_t sequence = mSequence;
+ sequence++;
+ android_atomic_acquire_store(sequence, &shared->mSequence);
+ shared->mValue = value;
+ sequence++;
+ android_atomic_release_store(sequence, &shared->mSequence);
+ mSequence = sequence;
+ // consider signalling a futex here, if we know that observer is waiting
+ return sequence;
+ }
- // return true if most recent push has been observed
- bool ack();
+ // returns the status of the last state push. This may be a stale value.
+ //
+ // SSQ_PENDING, or 0, means it has not been observed
+ // SSQ_READ means it has been read
+ // SSQ_DONE means it has been acted upon, after Observer::done() is called
+ enum SSQ_STATUS ack() const
+ {
+ // in the case of SSQ_DONE, prevent any subtle data-races of subsequent reads
+ // being performed (out-of-order) before the ack read, should the caller be
+ // depending on sequentiality of reads.
+ const int32_t ack = android_atomic_acquire_load(&mShared->mAck);
+ return ack - mSequence & ~1 ? SSQ_PENDING /* seq differ */ :
+ ack & 1 ? SSQ_DONE : SSQ_READ;
+ }
// return true if a push with specified sequence number or later has been observed
- bool ack(int32_t sequence);
+ bool ack(int32_t sequence) const
+ {
+ // this relies on 2's complement rollover to detect an ancient sequence number
+ return mShared->mAck - sequence >= 0;
+ }
private:
int32_t mSequence;
@@ -71,11 +106,54 @@
class Observer {
public:
- Observer(Shared *shared);
- /*virtual*/ ~Observer() { }
+ Observer(Shared *shared)
+ : mSequence(0), mSeed(1), mShared(shared)
+ {
+ // exactly one of Mutator and Observer must initialize, currently it is Observer
+ shared->init();
+ }
// return true if value has changed
- bool poll(T& value);
+ bool poll(T& value)
+ {
+ Shared *shared = mShared;
+ int32_t before = shared->mSequence;
+ if (before == mSequence) {
+ return false;
+ }
+ for (int tries = 0; ; ) {
+ const int MAX_TRIES = 5;
+ if (before & 1) {
+ if (++tries >= MAX_TRIES) {
+ return false;
+ }
+ before = shared->mSequence;
+ } else {
+ android_memory_barrier();
+ T temp = shared->mValue;
+ int32_t after = android_atomic_release_load(&shared->mSequence);
+ if (after == before) {
+ value = temp;
+ shared->mAck = before;
+ mSequence = before; // mSequence is even after poll success
+ return true;
+ }
+ if (++tries >= MAX_TRIES) {
+ return false;
+ }
+ before = after;
+ }
+ }
+ }
+
+ // (optional) used to indicate to the Mutator that the state that has been polled
+ // has also been acted upon.
+ void done()
+ {
+ const int32_t ack = mShared->mAck + 1;
+ // ensure all previous writes have been performed.
+ android_atomic_release_store(ack, &mShared->mAck); // mSequence is odd after "done"
+ }
private:
int32_t mSequence;
diff --git a/include/media/StringArray.h b/include/media/StringArray.h
index ae47085..48d98bf 100644
--- a/include/media/StringArray.h
+++ b/include/media/StringArray.h
@@ -16,7 +16,7 @@
//
// Sortable array of strings. STL-ish, but STL-free.
-//
+//
#ifndef _LIBS_MEDIA_STRING_ARRAY_H
#define _LIBS_MEDIA_STRING_ARRAY_H
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index 5830933..808e893 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -220,6 +220,7 @@
status_t stop();
status_t pause();
bool isPlaying();
+ status_t setPlaybackRate(float rate);
status_t getVideoWidth(int *w);
status_t getVideoHeight(int *h);
status_t seekTo(int msec);
@@ -274,6 +275,7 @@
int mVideoWidth;
int mVideoHeight;
int mAudioSessionId;
+ float mPlaybackRate;
float mSendLevel;
struct sockaddr_in mRetransmitEndpoint;
bool mRetransmitEndpointValid;
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
index b0a62a7..74a6469 100644
--- a/include/media/mediarecorder.h
+++ b/include/media/mediarecorder.h
@@ -221,7 +221,6 @@
status_t setOutputFormat(int of);
status_t setVideoEncoder(int ve);
status_t setAudioEncoder(int ae);
- status_t setOutputFile(const char* path);
status_t setOutputFile(int fd, int64_t offset, int64_t length);
status_t setVideoSize(int width, int height);
status_t setVideoFrameRate(int frames_per_second);
diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index d422576..d9bbc8d 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -231,7 +231,8 @@
virtual status_t getTimestamp(AudioTimestamp& timestamp) { return INVALID_OPERATION; }
protected:
- NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { }
+ NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0)
+ { }
virtual ~NBAIO_Sink() { }
// Implementations are free to ignore these if they don't need them
@@ -322,7 +323,8 @@
virtual void onTimestamp(const AudioTimestamp& timestamp) { }
protected:
- NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { }
+ NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0)
+ { }
virtual ~NBAIO_Source() { }
// Implementations are free to ignore these if they don't need them
diff --git a/include/media/nbaio/NBLog.h b/include/media/nbaio/NBLog.h
index bcbbc04..1297b51 100644
--- a/include/media/nbaio/NBLog.h
+++ b/include/media/nbaio/NBLog.h
@@ -21,7 +21,7 @@
#include <binder/IMemory.h>
#include <utils/Mutex.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
namespace android {
diff --git a/include/media/stagefright/AACWriter.h b/include/media/stagefright/AACWriter.h
index d22707a..86417a5 100644
--- a/include/media/stagefright/AACWriter.h
+++ b/include/media/stagefright/AACWriter.h
@@ -27,7 +27,6 @@
struct MetaData;
struct AACWriter : public MediaWriter {
- AACWriter(const char *filename);
AACWriter(int fd);
status_t initCheck() const;
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index cd2bd27..c1483f3 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -214,6 +214,7 @@
int64_t mRepeatFrameDelayUs;
int64_t mMaxPtsGapUs;
+ float mMaxFps;
int64_t mTimePerFrameUs;
int64_t mTimePerCaptureUs;
@@ -298,6 +299,8 @@
status_t setupRawAudioFormat(
OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels);
+ status_t setPriority(int32_t priority);
+
status_t setMinBufferSize(OMX_U32 portIndex, size_t size);
status_t setupMPEG4EncoderParameters(const sp<AMessage> &msg);
diff --git a/include/media/stagefright/AMRWriter.h b/include/media/stagefright/AMRWriter.h
index 392f968..bac878b 100644
--- a/include/media/stagefright/AMRWriter.h
+++ b/include/media/stagefright/AMRWriter.h
@@ -29,7 +29,6 @@
struct MetaData;
struct AMRWriter : public MediaWriter {
- AMRWriter(const char *filename);
AMRWriter(int fd);
status_t initCheck() const;
diff --git a/include/media/stagefright/BufferProducerWrapper.h b/include/media/stagefright/BufferProducerWrapper.h
index d8acf30..4caa2c6 100644
--- a/include/media/stagefright/BufferProducerWrapper.h
+++ b/include/media/stagefright/BufferProducerWrapper.h
@@ -19,6 +19,7 @@
#define BUFFER_PRODUCER_WRAPPER_H_
#include <gui/IGraphicBufferProducer.h>
+#include <media/stagefright/foundation/ABase.h>
namespace android {
diff --git a/include/media/stagefright/MPEG2TSWriter.h b/include/media/stagefright/MPEG2TSWriter.h
index 2e2922e..3d7960b 100644
--- a/include/media/stagefright/MPEG2TSWriter.h
+++ b/include/media/stagefright/MPEG2TSWriter.h
@@ -29,7 +29,6 @@
struct MPEG2TSWriter : public MediaWriter {
MPEG2TSWriter(int fd);
- MPEG2TSWriter(const char *filename);
MPEG2TSWriter(
void *cookie,
diff --git a/include/media/stagefright/MPEG4Writer.h b/include/media/stagefright/MPEG4Writer.h
index 26ce5f9..a195fe8 100644
--- a/include/media/stagefright/MPEG4Writer.h
+++ b/include/media/stagefright/MPEG4Writer.h
@@ -26,13 +26,13 @@
namespace android {
+class AMessage;
class MediaBuffer;
class MediaSource;
class MetaData;
class MPEG4Writer : public MediaWriter {
public:
- MPEG4Writer(const char *filename);
MPEG4Writer(int fd);
// Limitations
@@ -49,6 +49,7 @@
virtual status_t dump(int fd, const Vector<String16>& args);
void beginBox(const char *fourcc);
+ void beginBox(uint32_t id);
void writeInt8(int8_t x);
void writeInt16(int16_t x);
void writeInt32(int32_t x);
@@ -63,6 +64,7 @@
int32_t getTimeScale() const { return mTimeScale; }
status_t setGeoData(int latitudex10000, int longitudex10000);
+ status_t setCaptureRate(float captureFps);
virtual void setStartTimeOffsetMs(int ms) { mStartTimeOffsetMs = ms; }
virtual int32_t getStartTimeOffsetMs() const { return mStartTimeOffsetMs; }
@@ -89,6 +91,7 @@
off64_t mFreeBoxOffset;
bool mStreamableFile;
off64_t mEstimatedMoovBoxSize;
+ off64_t mMoovExtraSize;
uint32_t mInterleaveDurationUs;
int32_t mTimeScale;
int64_t mStartTimestampUs;
@@ -103,6 +106,8 @@
List<off64_t> mBoxes;
+ sp<AMessage> mMetaKeys;
+
void setStartTimestampUs(int64_t timeUs);
int64_t getStartTimestampUs(); // Not const
status_t startTracks(MetaData *params);
@@ -196,6 +201,12 @@
void writeGeoDataBox();
void writeLatitude(int degreex10000);
void writeLongitude(int degreex10000);
+
+ void addDeviceMeta();
+ void writeHdlr();
+ void writeKeys();
+ void writeIlst();
+ void writeMetaBox();
void sendSessionSummary();
void release();
status_t reset();
diff --git a/include/media/stagefright/MediaClock.h b/include/media/stagefright/MediaClock.h
new file mode 100644
index 0000000..e9c09a1
--- /dev/null
+++ b/include/media/stagefright/MediaClock.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (C) 2015 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 MEDIA_CLOCK_H_
+
+#define MEDIA_CLOCK_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/Mutex.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+struct AMessage;
+
+struct MediaClock : public RefBase {
+ MediaClock();
+
+ void setStartingTimeMedia(int64_t startingTimeMediaUs);
+
+ void clearAnchor();
+ // It's required to use timestamp of just rendered frame as
+ // anchor time in paused state.
+ void updateAnchor(
+ int64_t anchorTimeMediaUs,
+ int64_t anchorTimeRealUs,
+ int64_t maxTimeMediaUs = INT64_MAX);
+
+ void updateMaxTimeMedia(int64_t maxTimeMediaUs);
+
+ void setPlaybackRate(float rate);
+
+ // query media time corresponding to real time |realUs|, and save the
+ // result in |outMediaUs|.
+ status_t getMediaTime(
+ int64_t realUs,
+ int64_t *outMediaUs,
+ bool allowPastMaxTime = false) const;
+ // query real time corresponding to media time |targetMediaUs|.
+ // The result is saved in |outRealUs|.
+ status_t getRealTimeFor(int64_t targetMediaUs, int64_t *outRealUs) const;
+
+protected:
+ virtual ~MediaClock();
+
+private:
+ status_t getMediaTime_l(
+ int64_t realUs,
+ int64_t *outMediaUs,
+ bool allowPastMaxTime) const;
+
+ mutable Mutex mLock;
+
+ int64_t mAnchorTimeMediaUs;
+ int64_t mAnchorTimeRealUs;
+ int64_t mMaxTimeMediaUs;
+ int64_t mStartingTimeMediaUs;
+
+ float mPlaybackRate;
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaClock);
+};
+
+} // namespace android
+
+#endif // MEDIA_CLOCK_H_
diff --git a/include/media/stagefright/MediaCodec.h b/include/media/stagefright/MediaCodec.h
index d448097..d055341 100644
--- a/include/media/stagefright/MediaCodec.h
+++ b/include/media/stagefright/MediaCodec.h
@@ -27,6 +27,7 @@
struct ABuffer;
struct AMessage;
+struct AReplyToken;
struct AString;
struct CodecBase;
struct ICrypto;
@@ -50,6 +51,7 @@
CB_OUTPUT_AVAILABLE = 2,
CB_ERROR = 3,
CB_OUTPUT_FORMAT_CHANGED = 4,
+ CB_RESOURCE_RECLAIMED = 5,
};
struct BatteryNotifier;
@@ -222,7 +224,7 @@
sp<ALooper> mCodecLooper;
sp<CodecBase> mCodec;
AString mComponentName;
- uint32_t mReplyID;
+ sp<AReplyToken> mReplyID;
uint32_t mFlags;
status_t mStickyError;
sp<Surface> mNativeWindow;
@@ -249,10 +251,10 @@
Vector<BufferInfo> mPortBuffers[2];
int32_t mDequeueInputTimeoutGeneration;
- uint32_t mDequeueInputReplyID;
+ sp<AReplyToken> mDequeueInputReplyID;
int32_t mDequeueOutputTimeoutGeneration;
- uint32_t mDequeueOutputReplyID;
+ sp<AReplyToken> mDequeueOutputReplyID;
sp<ICrypto> mCrypto;
@@ -267,7 +269,7 @@
static status_t PostAndAwaitResponse(
const sp<AMessage> &msg, sp<AMessage> *response);
- static void PostReplyWithError(int32_t replyID, int32_t err);
+ static void PostReplyWithError(const sp<AReplyToken> &replyID, int32_t err);
status_t init(const AString &name, bool nameIsType, bool encoder);
@@ -283,8 +285,8 @@
size_t portIndex, size_t index,
sp<ABuffer> *buffer, sp<AMessage> *format);
- bool handleDequeueInputBuffer(uint32_t replyID, bool newRequest = false);
- bool handleDequeueOutputBuffer(uint32_t replyID, bool newRequest = false);
+ bool handleDequeueInputBuffer(const sp<AReplyToken> &replyID, bool newRequest = false);
+ bool handleDequeueOutputBuffer(const sp<AReplyToken> &replyID, bool newRequest = false);
void cancelPendingDequeueOperations();
void extractCSD(const sp<AMessage> &format);
diff --git a/include/media/stagefright/MediaCodecSource.h b/include/media/stagefright/MediaCodecSource.h
index 0970b2b..7b8f59d 100644
--- a/include/media/stagefright/MediaCodecSource.h
+++ b/include/media/stagefright/MediaCodecSource.h
@@ -25,6 +25,7 @@
class ALooper;
class AMessage;
+struct AReplyToken;
class IGraphicBufferProducer;
class MediaCodec;
class MetaData;
@@ -99,7 +100,7 @@
sp<Puller> mPuller;
sp<MediaCodec> mEncoder;
uint32_t mFlags;
- List<uint32_t> mStopReplyIDQueue;
+ List<sp<AReplyToken>> mStopReplyIDQueue;
bool mIsVideo;
bool mStarted;
bool mStopping;
diff --git a/include/media/stagefright/MediaFilter.h b/include/media/stagefright/MediaFilter.h
new file mode 100644
index 0000000..7b3f700
--- /dev/null
+++ b/include/media/stagefright/MediaFilter.h
@@ -0,0 +1,167 @@
+/*
+ * 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 MEDIA_FILTER_H_
+#define MEDIA_FILTER_H_
+
+#include <media/stagefright/CodecBase.h>
+
+namespace android {
+
+struct ABuffer;
+struct GraphicBufferListener;
+struct MemoryDealer;
+struct SimpleFilter;
+
+struct MediaFilter : public CodecBase {
+ MediaFilter();
+
+ virtual void setNotificationMessage(const sp<AMessage> &msg);
+
+ virtual void initiateAllocateComponent(const sp<AMessage> &msg);
+ virtual void initiateConfigureComponent(const sp<AMessage> &msg);
+ virtual void initiateCreateInputSurface();
+ virtual void initiateStart();
+ virtual void initiateShutdown(bool keepComponentAllocated = false);
+
+ virtual void signalFlush();
+ virtual void signalResume();
+
+ virtual void signalRequestIDRFrame();
+ virtual void signalSetParameters(const sp<AMessage> &msg);
+ virtual void signalEndOfInputStream();
+
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+ struct PortDescription : public CodecBase::PortDescription {
+ virtual size_t countBuffers();
+ virtual IOMX::buffer_id bufferIDAt(size_t index) const;
+ virtual sp<ABuffer> bufferAt(size_t index) const;
+
+ protected:
+ PortDescription();
+
+ private:
+ friend struct MediaFilter;
+
+ Vector<IOMX::buffer_id> mBufferIDs;
+ Vector<sp<ABuffer> > mBuffers;
+
+ void addBuffer(IOMX::buffer_id id, const sp<ABuffer> &buffer);
+
+ DISALLOW_EVIL_CONSTRUCTORS(PortDescription);
+ };
+
+protected:
+ virtual ~MediaFilter();
+
+private:
+ struct BufferInfo {
+ enum Status {
+ OWNED_BY_US,
+ OWNED_BY_UPSTREAM,
+ };
+
+ IOMX::buffer_id mBufferID;
+ int32_t mGeneration;
+ int32_t mOutputFlags;
+ Status mStatus;
+
+ sp<ABuffer> mData;
+ };
+
+ enum State {
+ UNINITIALIZED,
+ INITIALIZED,
+ CONFIGURED,
+ STARTED,
+ };
+
+ enum {
+ kWhatInputBufferFilled = 'inpF',
+ kWhatOutputBufferDrained = 'outD',
+ kWhatShutdown = 'shut',
+ kWhatFlush = 'flus',
+ kWhatResume = 'resm',
+ kWhatAllocateComponent = 'allo',
+ kWhatConfigureComponent = 'conf',
+ kWhatCreateInputSurface = 'cisf',
+ kWhatSignalEndOfInputStream = 'eois',
+ kWhatStart = 'star',
+ kWhatSetParameters = 'setP',
+ kWhatProcessBuffers = 'proc',
+ };
+
+ enum {
+ kPortIndexInput = 0,
+ kPortIndexOutput = 1
+ };
+
+ // member variables
+ AString mComponentName;
+ State mState;
+ status_t mInputEOSResult;
+ int32_t mWidth, mHeight;
+ int32_t mStride, mSliceHeight;
+ int32_t mColorFormatIn, mColorFormatOut;
+ size_t mMaxInputSize, mMaxOutputSize;
+ int32_t mGeneration;
+ sp<AMessage> mNotify;
+ sp<AMessage> mInputFormat;
+ sp<AMessage> mOutputFormat;
+
+ sp<MemoryDealer> mDealer[2];
+ Vector<BufferInfo> mBuffers[2];
+ Vector<BufferInfo*> mAvailableInputBuffers;
+ Vector<BufferInfo*> mAvailableOutputBuffers;
+ bool mPortEOS[2];
+
+ sp<SimpleFilter> mFilter;
+ sp<GraphicBufferListener> mGraphicBufferListener;
+
+ // helper functions
+ void signalProcessBuffers();
+ void signalError(status_t error);
+
+ status_t allocateBuffersOnPort(OMX_U32 portIndex);
+ BufferInfo *findBufferByID(
+ uint32_t portIndex, IOMX::buffer_id bufferID,
+ ssize_t *index = NULL);
+ void postFillThisBuffer(BufferInfo *info);
+ void postDrainThisBuffer(BufferInfo *info);
+ void postEOS();
+ void sendFormatChange();
+ void requestFillEmptyInput();
+ void processBuffers();
+
+ void onAllocateComponent(const sp<AMessage> &msg);
+ void onConfigureComponent(const sp<AMessage> &msg);
+ void onStart();
+ void onInputBufferFilled(const sp<AMessage> &msg);
+ void onOutputBufferDrained(const sp<AMessage> &msg);
+ void onShutdown(const sp<AMessage> &msg);
+ void onFlush();
+ void onSetParameters(const sp<AMessage> &msg);
+ void onCreateInputSurface();
+ void onInputFrameAvailable();
+ void onSignalEndOfInputStream();
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaFilter);
+};
+
+} // namespace android
+
+#endif // MEDIA_FILTER_H_
diff --git a/include/media/stagefright/MediaMuxer.h b/include/media/stagefright/MediaMuxer.h
index 9da98d9..e6538d1 100644
--- a/include/media/stagefright/MediaMuxer.h
+++ b/include/media/stagefright/MediaMuxer.h
@@ -50,9 +50,6 @@
OUTPUT_FORMAT_LIST_END // must be last - used to validate format type
};
- // Construct the muxer with the output file path.
- MediaMuxer(const char *path, OutputFormat format);
-
// Construct the muxer with the file descriptor. Note that the MediaMuxer
// will close this file at stop().
MediaMuxer(int fd, OutputFormat format);
diff --git a/include/media/stagefright/MediaSync.h b/include/media/stagefright/MediaSync.h
new file mode 100644
index 0000000..8bb8c7f
--- /dev/null
+++ b/include/media/stagefright/MediaSync.h
@@ -0,0 +1,239 @@
+/*
+ * Copyright 2015 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 MEDIA_SYNC_H
+#define MEDIA_SYNC_H
+
+#include <gui/IConsumerListener.h>
+#include <gui/IProducerListener.h>
+
+#include <media/stagefright/foundation/AHandler.h>
+
+#include <utils/Condition.h>
+#include <utils/Mutex.h>
+
+namespace android {
+
+class AudioTrack;
+class BufferItem;
+class Fence;
+class GraphicBuffer;
+class IGraphicBufferConsumer;
+class IGraphicBufferProducer;
+struct MediaClock;
+
+// MediaSync manages media playback and its synchronization to a media clock
+// source. It can be also used for video-only playback.
+//
+// For video playback, it requires an output surface and provides an input
+// surface. It then controls the rendering of input buffers (buffer queued to
+// the input surface) on the output surface to happen at the appropriate time.
+//
+// For audio playback, it requires an audio track and takes updates of
+// information of rendered audio data to maintain media clock when audio track
+// serves as media clock source. (TODO: move audio rendering from JAVA to
+// native code).
+//
+// It can use the audio or video track as media clock source, as well as an
+// external clock. (TODO: actually support external clock as media clock
+// sources; use video track as media clock source for audio-and-video stream).
+//
+// In video-only mode, MediaSync will playback every video frame even though
+// a video frame arrives late based on its timestamp and last frame's.
+//
+// The client needs to configure surface (for output video rendering) and audio
+// track (for querying information of audio rendering) for MediaSync.
+//
+// Then the client needs to obtain a surface from MediaSync and render video
+// frames onto that surface. Internally, the MediaSync will receive those video
+// frames and render them onto the output surface at the appropriate time.
+//
+// The client needs to call updateQueuedAudioData() immediately after it writes
+// audio data to the audio track. Such information will be used to update media
+// clock.
+//
+class MediaSync : public AHandler {
+public:
+ // Create an instance of MediaSync.
+ static sp<MediaSync> create();
+
+ // Called when MediaSync is used to render video. It should be called
+ // before createInputSurface().
+ status_t configureSurface(const sp<IGraphicBufferProducer> &output);
+
+ // Called when audio track is used as media clock source. It should be
+ // called before updateQueuedAudioData().
+ // |nativeSampleRateInHz| is the sample rate of audio data fed into audio
+ // track. It's the same number used to create AudioTrack.
+ status_t configureAudioTrack(
+ const sp<AudioTrack> &audioTrack, uint32_t nativeSampleRateInHz);
+
+ // Create a surface for client to render video frames. This is the surface
+ // on which the client should render video frames. Those video frames will
+ // be internally directed to output surface for rendering at appropriate
+ // time.
+ status_t createInputSurface(sp<IGraphicBufferProducer> *outBufferProducer);
+
+ // Update just-rendered audio data size and the presentation timestamp of
+ // the first frame of that audio data. It should be called immediately
+ // after the client write audio data into AudioTrack.
+ // This function assumes continous audio stream.
+ // TODO: support gap or backwards updates.
+ status_t updateQueuedAudioData(
+ size_t sizeInBytes, int64_t presentationTimeUs);
+
+ // Set the consumer name of the input queue.
+ void setName(const AString &name);
+
+ // Set the playback in a desired speed.
+ // This method can be called any time.
+ // |rate| is the ratio between desired speed and the normal one, and should
+ // be non-negative. The meaning of rate values:
+ // 1.0 -- normal playback
+ // 0.0 -- stop or pause
+ // larger than 1.0 -- faster than normal speed
+ // between 0.0 and 1.0 -- slower than normal speed
+ status_t setPlaybackRate(float rate);
+
+ // Get the media clock used by the MediaSync so that the client can obtain
+ // corresponding media time or real time via
+ // MediaClock::getMediaTime() and MediaClock::getRealTimeFor().
+ sp<const MediaClock> getMediaClock();
+
+protected:
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum {
+ kWhatDrainVideo = 'dVid',
+ };
+
+ static const int MAX_OUTSTANDING_BUFFERS = 2;
+
+ // This is a thin wrapper class that lets us listen to
+ // IConsumerListener::onFrameAvailable from mInput.
+ class InputListener : public BnConsumerListener,
+ public IBinder::DeathRecipient {
+ public:
+ InputListener(const sp<MediaSync> &sync);
+ virtual ~InputListener();
+
+ // From IConsumerListener
+ virtual void onFrameAvailable(const BufferItem &item);
+
+ // From IConsumerListener
+ // We don't care about released buffers because we detach each buffer as
+ // soon as we acquire it. See the comment for onBufferReleased below for
+ // some clarifying notes about the name.
+ virtual void onBuffersReleased() {}
+
+ // From IConsumerListener
+ // We don't care about sideband streams, since we won't relay them.
+ virtual void onSidebandStreamChanged();
+
+ // From IBinder::DeathRecipient
+ virtual void binderDied(const wp<IBinder> &who);
+
+ private:
+ sp<MediaSync> mSync;
+ };
+
+ // This is a thin wrapper class that lets us listen to
+ // IProducerListener::onBufferReleased from mOutput.
+ class OutputListener : public BnProducerListener,
+ public IBinder::DeathRecipient {
+ public:
+ OutputListener(const sp<MediaSync> &sync);
+ virtual ~OutputListener();
+
+ // From IProducerListener
+ virtual void onBufferReleased();
+
+ // From IBinder::DeathRecipient
+ virtual void binderDied(const wp<IBinder> &who);
+
+ private:
+ sp<MediaSync> mSync;
+ };
+
+ // mIsAbandoned is set to true when the input or output dies.
+ // Once the MediaSync has been abandoned by one side, it will disconnect
+ // from the other side and not attempt to communicate with it further.
+ bool mIsAbandoned;
+
+ mutable Mutex mMutex;
+ Condition mReleaseCondition;
+ size_t mNumOutstandingBuffers;
+ sp<IGraphicBufferConsumer> mInput;
+ sp<IGraphicBufferProducer> mOutput;
+
+ sp<AudioTrack> mAudioTrack;
+ uint32_t mNativeSampleRateInHz;
+ int64_t mNumFramesWritten;
+ bool mHasAudio;
+
+ int64_t mNextBufferItemMediaUs;
+ List<BufferItem> mBufferItems;
+ sp<ALooper> mLooper;
+ float mPlaybackRate;
+
+ sp<MediaClock> mMediaClock;
+
+ MediaSync();
+
+ // Must be accessed through RefBase
+ virtual ~MediaSync();
+
+ int64_t getRealTime(int64_t mediaTimeUs, int64_t nowUs);
+ int64_t getDurationIfPlayedAtNativeSampleRate_l(int64_t numFrames);
+ int64_t getPlayedOutAudioDurationMedia_l(int64_t nowUs);
+
+ void onDrainVideo_l();
+
+ // This implements the onFrameAvailable callback from IConsumerListener.
+ // It gets called from an InputListener.
+ // During this callback, we detach the buffer from the input, and queue
+ // it for rendering on the output. This call can block if there are too
+ // many outstanding buffers. If it blocks, it will resume when
+ // onBufferReleasedByOutput releases a buffer back to the input.
+ void onFrameAvailableFromInput();
+
+ // Send |bufferItem| to the output for rendering.
+ void renderOneBufferItem_l(const BufferItem &bufferItem);
+
+ // This implements the onBufferReleased callback from IProducerListener.
+ // It gets called from an OutputListener.
+ // During this callback, we detach the buffer from the output, and release
+ // it to the input. A blocked onFrameAvailable call will be allowed to proceed.
+ void onBufferReleasedByOutput();
+
+ // Return |buffer| back to the input.
+ void returnBufferToInput_l(const sp<GraphicBuffer> &buffer, const sp<Fence> &fence);
+
+ // When this is called, the MediaSync disconnects from (i.e., abandons) its
+ // input or output, and signals any waiting onFrameAvailable calls to wake
+ // up. This must be called with mMutex locked.
+ void onAbandoned_l(bool isInput);
+
+ // helper.
+ bool isPlaying() { return mPlaybackRate != 0.0; }
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaSync);
+};
+
+} // namespace android
+
+#endif
diff --git a/include/media/stagefright/ProcessInfo.h b/include/media/stagefright/ProcessInfo.h
new file mode 100644
index 0000000..ec0cdff
--- /dev/null
+++ b/include/media/stagefright/ProcessInfo.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2015 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 PROCESS_INFO_H_
+
+#define PROCESS_INFO_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/ProcessInfoInterface.h>
+
+namespace android {
+
+struct ProcessInfo : public ProcessInfoInterface {
+ ProcessInfo();
+
+ virtual bool getPriority(int pid, int* priority);
+
+protected:
+ virtual ~ProcessInfo();
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(ProcessInfo);
+};
+
+} // namespace android
+
+#endif // PROCESS_INFO_H_
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/include/media/stagefright/ProcessInfoInterface.h
similarity index 60%
copy from services/audiopolicy/AudioPolicyFactory.cpp
copy to include/media/stagefright/ProcessInfoInterface.h
index 2ae7bc1..222f92d 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/include/media/stagefright/ProcessInfoInterface.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2014 The Android Open Source Project
+ * Copyright (C) 2015 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.
@@ -14,19 +14,20 @@
* limitations under the License.
*/
-#include "AudioPolicyManager.h"
+#ifndef PROCESS_INFO_INTERFACE_H_
+#define PROCESS_INFO_INTERFACE_H_
+
+#include <utils/RefBase.h>
namespace android {
-extern "C" AudioPolicyInterface* createAudioPolicyManager(
- AudioPolicyClientInterface *clientInterface)
-{
- return new AudioPolicyManager(clientInterface);
-}
+struct ProcessInfoInterface : public RefBase {
+ virtual bool getPriority(int pid, int* priority) = 0;
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
-{
- delete interface;
-}
+protected:
+ virtual ~ProcessInfoInterface() {}
+};
-}; // namespace android
+} // namespace android
+
+#endif // PROCESS_INFO_INTERFACE_H_
diff --git a/include/media/stagefright/RenderScriptWrapper.h b/include/media/stagefright/RenderScriptWrapper.h
new file mode 100644
index 0000000..b42649e
--- /dev/null
+++ b/include/media/stagefright/RenderScriptWrapper.h
@@ -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.
+ */
+
+#ifndef RENDERSCRIPT_WRAPPER_H_
+#define RENDERSCRIPT_WRAPPER_H_
+
+#include <RenderScript.h>
+
+namespace android {
+
+struct RenderScriptWrapper : public RefBase {
+public:
+ struct RSFilterCallback : public RefBase {
+ public:
+ // called by RSFilter to process each input buffer
+ virtual status_t processBuffers(
+ RSC::Allocation* inBuffer,
+ RSC::Allocation* outBuffer) = 0;
+
+ virtual status_t handleSetParameters(const sp<AMessage> &msg) = 0;
+ };
+
+ sp<RSFilterCallback> mCallback;
+ RSC::sp<RSC::RS> mContext;
+};
+
+} // namespace android
+
+#endif // RENDERSCRIPT_WRAPPER_H_
diff --git a/include/media/stagefright/Utils.h b/include/media/stagefright/Utils.h
index a795c80..ec3a10e 100644
--- a/include/media/stagefright/Utils.h
+++ b/include/media/stagefright/Utils.h
@@ -65,6 +65,17 @@
AString uriDebugString(const AString &uri, bool incognito = false);
+struct HLSTime {
+ int32_t mSeq;
+ int64_t mTimeUs;
+ sp<AMessage> mMeta;
+
+ HLSTime(const sp<AMessage> &meta = NULL);
+ int64_t getSegmentTimeUs(bool midpoint = false) const;
+};
+
+bool operator <(const HLSTime &t0, const HLSTime &t1);
+
} // namespace android
#endif // UTILS_H_
diff --git a/include/media/stagefright/foundation/ABase.h b/include/media/stagefright/foundation/ABase.h
index 72e3d87..ef1e010 100644
--- a/include/media/stagefright/foundation/ABase.h
+++ b/include/media/stagefright/foundation/ABase.h
@@ -18,7 +18,9 @@
#define A_BASE_H_
+#ifndef ARRAY_SIZE
#define ARRAY_SIZE(a) (sizeof(a) / sizeof(*(a)))
+#endif
#define DISALLOW_EVIL_CONSTRUCTORS(name) \
name(const name &); \
diff --git a/include/media/stagefright/foundation/AHandler.h b/include/media/stagefright/foundation/AHandler.h
index 41ade77..fe02a86 100644
--- a/include/media/stagefright/foundation/AHandler.h
+++ b/include/media/stagefright/foundation/AHandler.h
@@ -29,6 +29,7 @@
struct AHandler : public RefBase {
AHandler()
: mID(0),
+ mVerboseStats(false),
mMessageCounter(0) {
}
@@ -36,23 +37,40 @@
return mID;
}
- sp<ALooper> looper();
+ sp<ALooper> looper() const {
+ return mLooper.promote();
+ }
+
+ wp<ALooper> getLooper() const {
+ return mLooper;
+ }
+
+ wp<AHandler> getHandler() const {
+ // allow getting a weak reference to a const handler
+ return const_cast<AHandler *>(this);
+ }
protected:
virtual void onMessageReceived(const sp<AMessage> &msg) = 0;
private:
- friend struct ALooperRoster;
+ friend struct AMessage; // deliverMessage()
+ friend struct ALooperRoster; // setID()
ALooper::handler_id mID;
+ wp<ALooper> mLooper;
- void setID(ALooper::handler_id id) {
+ inline void setID(ALooper::handler_id id, wp<ALooper> looper) {
mID = id;
+ mLooper = looper;
}
+ bool mVerboseStats;
uint32_t mMessageCounter;
KeyedVector<uint32_t, uint32_t> mMessages;
+ void deliverMessage(const sp<AMessage> &msg);
+
DISALLOW_EVIL_CONSTRUCTORS(AHandler);
};
diff --git a/include/media/stagefright/foundation/ALooper.h b/include/media/stagefright/foundation/ALooper.h
index 70e0c5e..09c469b 100644
--- a/include/media/stagefright/foundation/ALooper.h
+++ b/include/media/stagefright/foundation/ALooper.h
@@ -30,6 +30,7 @@
struct AHandler;
struct AMessage;
+struct AReplyToken;
struct ALooper : public RefBase {
typedef int32_t event_id;
@@ -53,11 +54,15 @@
static int64_t GetNowUs();
+ const char *getName() const {
+ return mName.c_str();
+ }
+
protected:
virtual ~ALooper();
private:
- friend struct ALooperRoster;
+ friend struct AMessage; // post()
struct Event {
int64_t mWhenUs;
@@ -75,12 +80,32 @@
sp<LooperThread> mThread;
bool mRunningLocally;
+ // use a separate lock for reply handling, as it is always on another thread
+ // use a central lock, however, to avoid creating a mutex for each reply
+ Mutex mRepliesLock;
+ Condition mRepliesCondition;
+
+ // START --- methods used only by AMessage
+
+ // posts a message on this looper with the given timeout
void post(const sp<AMessage> &msg, int64_t delayUs);
+
+ // creates a reply token to be used with this looper
+ sp<AReplyToken> createReplyToken();
+ // waits for a response for the reply token. If status is OK, the response
+ // is stored into the supplied variable. Otherwise, it is unchanged.
+ status_t awaitResponse(const sp<AReplyToken> &replyToken, sp<AMessage> *response);
+ // posts a reply for a reply token. If the reply could be successfully posted,
+ // it returns OK. Otherwise, it returns an error value.
+ status_t postReply(const sp<AReplyToken> &replyToken, const sp<AMessage> &msg);
+
+ // END --- methods used only by AMessage
+
bool loop();
DISALLOW_EVIL_CONSTRUCTORS(ALooper);
};
-} // namespace android
+} // namespace android
#endif // A_LOOPER_H_
diff --git a/include/media/stagefright/foundation/ALooperRoster.h b/include/media/stagefright/foundation/ALooperRoster.h
index a0be8eb..9912455 100644
--- a/include/media/stagefright/foundation/ALooperRoster.h
+++ b/include/media/stagefright/foundation/ALooperRoster.h
@@ -33,16 +33,6 @@
void unregisterHandler(ALooper::handler_id handlerID);
void unregisterStaleHandlers();
- status_t postMessage(const sp<AMessage> &msg, int64_t delayUs = 0);
- void deliverMessage(const sp<AMessage> &msg);
-
- status_t postAndAwaitResponse(
- const sp<AMessage> &msg, sp<AMessage> *response);
-
- void postReply(uint32_t replyID, const sp<AMessage> &reply);
-
- sp<ALooper> findLooper(ALooper::handler_id handlerID);
-
void dump(int fd, const Vector<String16>& args);
private:
@@ -54,10 +44,6 @@
Mutex mLock;
KeyedVector<ALooper::handler_id, HandlerInfo> mHandlers;
ALooper::handler_id mNextHandlerID;
- uint32_t mNextReplyID;
- Condition mRepliesCondition;
-
- KeyedVector<uint32_t, sp<AMessage> > mReplies;
DISALLOW_EVIL_CONSTRUCTORS(ALooperRoster);
};
diff --git a/include/media/stagefright/foundation/AMessage.h b/include/media/stagefright/foundation/AMessage.h
index a9e235b..4c6bd21 100644
--- a/include/media/stagefright/foundation/AMessage.h
+++ b/include/media/stagefright/foundation/AMessage.h
@@ -26,11 +26,41 @@
namespace android {
struct ABuffer;
+struct AHandler;
struct AString;
struct Parcel;
+struct AReplyToken : public RefBase {
+ AReplyToken(const sp<ALooper> &looper)
+ : mLooper(looper),
+ mReplied(false) {
+ }
+
+private:
+ friend struct AMessage;
+ friend struct ALooper;
+ wp<ALooper> mLooper;
+ sp<AMessage> mReply;
+ bool mReplied;
+
+ sp<ALooper> getLooper() const {
+ return mLooper.promote();
+ }
+ // if reply is not set, returns false; otherwise, it retrieves the reply and returns true
+ bool retrieveReply(sp<AMessage> *reply) {
+ if (mReplied) {
+ *reply = mReply;
+ mReply.clear();
+ }
+ return mReplied;
+ }
+ // sets the reply for this token. returns OK or error
+ status_t setReply(const sp<AMessage> &reply);
+};
+
struct AMessage : public RefBase {
- AMessage(uint32_t what = 0, ALooper::handler_id target = 0);
+ AMessage();
+ AMessage(uint32_t what, const sp<const AHandler> &handler);
static sp<AMessage> FromParcel(const Parcel &parcel);
void writeToParcel(Parcel *parcel) const;
@@ -38,8 +68,7 @@
void setWhat(uint32_t what);
uint32_t what() const;
- void setTarget(ALooper::handler_id target);
- ALooper::handler_id target() const;
+ void setTarget(const sp<const AHandler> &handler);
void clear();
@@ -76,18 +105,22 @@
const char *name,
int32_t *left, int32_t *top, int32_t *right, int32_t *bottom) const;
- void post(int64_t delayUs = 0);
+ status_t post(int64_t delayUs = 0);
// Posts the message to its target and waits for a response (or error)
// before returning.
status_t postAndAwaitResponse(sp<AMessage> *response);
// If this returns true, the sender of this message is synchronously
- // awaiting a response, the "replyID" can be used to send the response
- // via "postReply" below.
- bool senderAwaitsResponse(uint32_t *replyID) const;
+ // awaiting a response and the reply token is consumed from the message
+ // and stored into replyID. The reply token must be used to send the response
+ // using "postReply" below.
+ bool senderAwaitsResponse(sp<AReplyToken> *replyID);
- void postReply(uint32_t replyID);
+ // Posts the message as a response to a reply token. A reply token can
+ // only be used once. Returns OK if the response could be posted; otherwise,
+ // an error.
+ status_t postReply(const sp<AReplyToken> &replyID);
// Performs a deep-copy of "this", contained messages are in turn "dup'ed".
// Warning: RefBase items, i.e. "objects" are _not_ copied but only have
@@ -117,9 +150,16 @@
virtual ~AMessage();
private:
+ friend struct ALooper; // deliver()
+
uint32_t mWhat;
+
+ // used only for debugging
ALooper::handler_id mTarget;
+ wp<AHandler> mHandler;
+ wp<ALooper> mLooper;
+
struct Rect {
int32_t mLeft, mTop, mRight, mBottom;
};
@@ -157,6 +197,8 @@
size_t findItemIndex(const char *name, size_t len) const;
+ void deliver();
+
DISALLOW_EVIL_CONSTRUCTORS(AMessage);
};
diff --git a/include/ndk/NdkMediaCodec.h b/include/ndk/NdkMediaCodec.h
index c07f4c9..4f6a1ef 100644
--- a/include/ndk/NdkMediaCodec.h
+++ b/include/ndk/NdkMediaCodec.h
@@ -142,7 +142,8 @@
/**
* Get the index of the next available buffer of processed data.
*/
-ssize_t AMediaCodec_dequeueOutputBuffer(AMediaCodec*, AMediaCodecBufferInfo *info, int64_t timeoutUs);
+ssize_t AMediaCodec_dequeueOutputBuffer(AMediaCodec*, AMediaCodecBufferInfo *info,
+ int64_t timeoutUs);
AMediaFormat* AMediaCodec_getOutputFormat(AMediaCodec*);
/**
diff --git a/include/ndk/NdkMediaExtractor.h b/include/ndk/NdkMediaExtractor.h
index 7a4e702..7324d31 100644
--- a/include/ndk/NdkMediaExtractor.h
+++ b/include/ndk/NdkMediaExtractor.h
@@ -55,12 +55,14 @@
/**
* Set the file descriptor from which the extractor will read.
*/
-media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor*, int fd, off64_t offset, off64_t length);
+media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor*, int fd, off64_t offset,
+ off64_t length);
/**
* Set the URI from which the extractor will read.
*/
-media_status_t AMediaExtractor_setDataSource(AMediaExtractor*, const char *location); // TODO support headers
+media_status_t AMediaExtractor_setDataSource(AMediaExtractor*, const char *location);
+ // TODO support headers
/**
* Return the number of tracks in the previously specified media file
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 31dff36..5644428 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -24,9 +24,8 @@
#include <utils/threads.h>
#include <utils/Log.h>
#include <utils/RefBase.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
#include <media/SingleStateQueue.h>
-#include <private/media/StaticAudioTrackState.h>
namespace android {
@@ -54,22 +53,64 @@
struct AudioTrackSharedStreaming {
// similar to NBAIO MonoPipe
// in continuously incrementing frame units, take modulo buffer size, which must be a power of 2
- volatile int32_t mFront; // read by server
- volatile int32_t mRear; // write by client
+ volatile int32_t mFront; // read by consumer (output: server, input: client)
+ 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 uint32_t mUnderrunFrames; // server increments for each unavailable but desired frame
};
+// Represents a single state of an AudioTrack that was created in static mode (shared memory buffer
+// supplied by the client). This state needs to be communicated from the client to server. As this
+// state is too large to be updated atomically without a mutex, and mutexes aren't allowed here, the
+// state is wrapped by a SingleStateQueue.
+struct StaticAudioTrackState {
+ // Do not define constructors, destructors, or virtual methods as this is part of a
+ // union in shared memory and they will not get called properly.
+
+ // These fields should both be size_t, but since they are located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+
+ // The state has a sequence counter to indicate whether changes are made to loop or position.
+ // The sequence counter also currently indicates whether loop or position is first depending
+ // on which is greater; it jumps by max(mLoopSequence, mPositionSequence) + 1.
+
+ uint32_t mLoopStart;
+ uint32_t mLoopEnd;
+ int32_t mLoopCount;
+ uint32_t mLoopSequence; // a sequence counter to indicate changes to loop
+ uint32_t mPosition;
+ uint32_t mPositionSequence; // a sequence counter to indicate changes to position
+};
+
typedef SingleStateQueue<StaticAudioTrackState> StaticAudioTrackSingleStateQueue;
+struct StaticAudioTrackPosLoop {
+ // Do not define constructors, destructors, or virtual methods as this is part of a
+ // union in shared memory and will not get called properly.
+
+ // These fields should both be size_t, but since they are located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+
+ // This struct information is stored in a single state queue to communicate the
+ // static AudioTrack server state to the client while data is consumed.
+ // It is smaller than StaticAudioTrackState to prevent unnecessary information from
+ // being sent.
+
+ uint32_t mBufferPosition;
+ int32_t mLoopCount;
+};
+
+typedef SingleStateQueue<StaticAudioTrackPosLoop> StaticAudioTrackPosLoopQueue;
+
struct AudioTrackSharedStatic {
+ // client requests to the server for loop or position changes.
StaticAudioTrackSingleStateQueue::Shared
mSingleStateQueue;
- // This field should be a size_t, but since it is located in shared memory we
- // force to 32-bit. The client and server may have different typedefs for size_t.
- uint32_t mBufferPosition; // updated asynchronously by server,
- // "for entertainment purposes only"
+ // position info updated asynchronously by server and read by client,
+ // "for entertainment purposes only"
+ StaticAudioTrackPosLoopQueue::Shared
+ mPosLoopQueue;
};
// ----------------------------------------------------------------------------
@@ -96,7 +137,8 @@
uint32_t mServer; // Number of filled frames consumed by server (mIsOut),
// or filled frames provided by server (!mIsOut).
// It is updated asynchronously by server without a barrier.
- // The value should be used "for entertainment purposes only",
+ // The value should be used
+ // "for entertainment purposes only",
// which means don't make important decisions based on it.
uint32_t mPad1; // unused
@@ -313,8 +355,28 @@
virtual void flush();
#define MIN_LOOP 16 // minimum length of each loop iteration in frames
+
+ // setLoop(), setBufferPosition(), and setBufferPositionAndLoop() set the
+ // static buffer position and looping parameters. These commands are not
+ // synchronous (they do not wait or block); instead they take effect at the
+ // next buffer data read from the server side. However, the client side
+ // getters will read a cached version of the position and loop variables
+ // until the setting takes effect.
+ //
+ // setBufferPositionAndLoop() is equivalent to calling, in order, setLoop() and
+ // setBufferPosition().
+ //
+ // The functions should not be relied upon to do parameter or state checking.
+ // That is done at the AudioTrack level.
+
void setLoop(size_t loopStart, size_t loopEnd, int loopCount);
+ void setBufferPosition(size_t position);
+ void setBufferPositionAndLoop(size_t position, size_t loopStart, size_t loopEnd,
+ int loopCount);
size_t getBufferPosition();
+ // getBufferPositionAndLoopCount() provides the proper snapshot of
+ // position and loopCount together.
+ void getBufferPositionAndLoopCount(size_t *position, int *loopCount);
virtual size_t getMisalignment() {
return 0;
@@ -326,7 +388,9 @@
private:
StaticAudioTrackSingleStateQueue::Mutator mMutator;
- size_t mBufferPosition; // so that getBufferPosition() appears to be synchronous
+ StaticAudioTrackPosLoopQueue::Observer mPosLoopObserver;
+ StaticAudioTrackState mState; // last communicated state to server
+ StaticAudioTrackPosLoop mPosLoop; // snapshot of position and loop.
};
// ----------------------------------------------------------------------------
@@ -447,10 +511,13 @@
virtual uint32_t getUnderrunFrames() const { return 0; }
private:
+ status_t updateStateWithLoop(StaticAudioTrackState *localState,
+ const StaticAudioTrackState &update) const;
+ status_t updateStateWithPosition(StaticAudioTrackState *localState,
+ const StaticAudioTrackState &update) const;
ssize_t pollPosition(); // poll for state queue update, and return current position
StaticAudioTrackSingleStateQueue::Observer mObserver;
- size_t mPosition; // server's current play position in frames, relative to 0
-
+ StaticAudioTrackPosLoopQueue::Mutator mPosLoopMutator;
size_t mFramesReadySafe; // Assuming size_t read/writes are atomic on 32 / 64 bit
// processors, this is a thread-safe version of
// mFramesReady.
@@ -459,7 +526,8 @@
// can cause a track to appear to have a large number
// of frames. INT64_MAX means an infinite loop.
bool mFramesReadyIsCalledByMultipleThreads;
- StaticAudioTrackState mState;
+ StaticAudioTrackState mState; // Server side state. Any updates from client must be
+ // passed by the mObserver SingleStateQueue.
};
// Proxy used by AudioFlinger for servicing AudioRecord
diff --git a/include/private/media/StaticAudioTrackState.h b/include/private/media/StaticAudioTrackState.h
deleted file mode 100644
index d483061..0000000
--- a/include/private/media/StaticAudioTrackState.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (C) 2012 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 STATIC_AUDIO_TRACK_STATE_H
-#define STATIC_AUDIO_TRACK_STATE_H
-
-namespace android {
-
-// Represents a single state of an AudioTrack that was created in static mode (shared memory buffer
-// supplied by the client). This state needs to be communicated from the client to server. As this
-// state is too large to be updated atomically without a mutex, and mutexes aren't allowed here, the
-// state is wrapped by a SingleStateQueue.
-struct StaticAudioTrackState {
- // do not define constructors, destructors, or virtual methods
-
- // These fields should both be size_t, but since they are located in shared memory we
- // force to 32-bit. The client and server may have different typedefs for size_t.
- uint32_t mLoopStart;
- uint32_t mLoopEnd;
-
- int mLoopCount;
-};
-
-} // namespace android
-
-#endif // STATIC_AUDIO_TRACK_STATE_H
diff --git a/include/radio/IRadio.h b/include/radio/IRadio.h
new file mode 100644
index 0000000..1877f8f
--- /dev/null
+++ b/include/radio/IRadio.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2015 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_IRADIO_H
+#define ANDROID_HARDWARE_IRADIO_H
+
+#include <utils/RefBase.h>
+#include <binder/IInterface.h>
+#include <binder/IMemory.h>
+#include <binder/Parcel.h>
+#include <system/radio.h>
+
+namespace android {
+
+class IRadio : public IInterface
+{
+public:
+
+ DECLARE_META_INTERFACE(Radio);
+
+ virtual void detach() = 0;
+
+ virtual status_t setConfiguration(const struct radio_band_config *config) = 0;
+
+ virtual status_t getConfiguration(struct radio_band_config *config) = 0;
+
+ virtual status_t setMute(bool mute) = 0;
+
+ virtual status_t getMute(bool *mute) = 0;
+
+ virtual status_t step(radio_direction_t direction, bool skipSubChannel) = 0;
+
+ virtual status_t scan(radio_direction_t direction, bool skipSubChannel) = 0;
+
+ virtual status_t tune(unsigned int channel, unsigned int subChannel) = 0;
+
+ virtual status_t cancel() = 0;
+
+ virtual status_t getProgramInformation(struct radio_program_info *info) = 0;
+
+ virtual status_t hasControl(bool *hasControl) = 0;
+};
+
+// ----------------------------------------------------------------------------
+
+class BnRadio: public BnInterface<IRadio>
+{
+public:
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif //ANDROID_HARDWARE_IRADIO_H
diff --git a/include/radio/IRadioClient.h b/include/radio/IRadioClient.h
new file mode 100644
index 0000000..9062ad6
--- /dev/null
+++ b/include/radio/IRadioClient.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (C) 2015 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_IRADIO_CLIENT_H
+#define ANDROID_HARDWARE_IRADIO_CLIENT_H
+
+#include <utils/RefBase.h>
+#include <binder/IInterface.h>
+#include <binder/IMemory.h>
+#include <binder/Parcel.h>
+
+namespace android {
+
+class IRadioClient : public IInterface
+{
+public:
+
+ DECLARE_META_INTERFACE(RadioClient);
+
+ virtual void onEvent(const sp<IMemory>& eventMemory) = 0;
+
+};
+
+// ----------------------------------------------------------------------------
+
+class BnRadioClient : public BnInterface<IRadioClient>
+{
+public:
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif //ANDROID_HARDWARE_IRADIO_CLIENT_H
diff --git a/include/radio/IRadioService.h b/include/radio/IRadioService.h
new file mode 100644
index 0000000..a946dd5
--- /dev/null
+++ b/include/radio/IRadioService.h
@@ -0,0 +1,59 @@
+/*
+ * Copyright (C) 2015 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_IRADIO_SERVICE_H
+#define ANDROID_HARDWARE_IRADIO_SERVICE_H
+
+#include <utils/RefBase.h>
+#include <binder/IInterface.h>
+#include <binder/Parcel.h>
+#include <system/radio.h>
+
+namespace android {
+
+class IRadio;
+class IRadioClient;
+
+class IRadioService : public IInterface
+{
+public:
+
+ DECLARE_META_INTERFACE(RadioService);
+
+ virtual status_t listModules(struct radio_properties *properties,
+ uint32_t *numModules) = 0;
+
+ virtual status_t attach(const radio_handle_t handle,
+ const sp<IRadioClient>& client,
+ const struct radio_band_config *config,
+ bool withAudio,
+ sp<IRadio>& radio) = 0;
+};
+
+// ----------------------------------------------------------------------------
+
+class BnRadioService: public BnInterface<IRadioService>
+{
+public:
+ virtual status_t onTransact( uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif //ANDROID_HARDWARE_IRADIO_SERVICE_H
diff --git a/include/radio/Radio.h b/include/radio/Radio.h
new file mode 100644
index 0000000..302bf16
--- /dev/null
+++ b/include/radio/Radio.h
@@ -0,0 +1,88 @@
+/*
+ * Copyright (C) 2015 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_RADIO_H
+#define ANDROID_HARDWARE_RADIO_H
+
+#include <binder/IBinder.h>
+#include <utils/threads.h>
+#include <radio/RadioCallback.h>
+#include <radio/IRadio.h>
+#include <radio/IRadioService.h>
+#include <radio/IRadioClient.h>
+#include <system/radio.h>
+
+namespace android {
+
+class MemoryDealer;
+
+class Radio : public BnRadioClient,
+ public IBinder::DeathRecipient
+{
+public:
+
+ virtual ~Radio();
+
+ static status_t listModules(struct radio_properties *properties,
+ uint32_t *numModules);
+ static sp<Radio> attach(radio_handle_t handle,
+ const struct radio_band_config *config,
+ bool withAudio,
+ const sp<RadioCallback>& callback);
+
+
+ void detach();
+
+ status_t setConfiguration(const struct radio_band_config *config);
+
+ status_t getConfiguration(struct radio_band_config *config);
+
+ status_t setMute(bool mute);
+
+ status_t getMute(bool *mute);
+
+ status_t step(radio_direction_t direction, bool skipSubChannel);
+
+ status_t scan(radio_direction_t direction, bool skipSubChannel);
+
+ status_t tune(unsigned int channel, unsigned int subChannel);
+
+ status_t cancel();
+
+ status_t getProgramInformation(struct radio_program_info *info);
+
+ status_t hasControl(bool *hasControl);
+
+ // BpRadioClient
+ virtual void onEvent(const sp<IMemory>& eventMemory);
+
+ //IBinder::DeathRecipient
+ virtual void binderDied(const wp<IBinder>& who);
+
+private:
+ Radio(radio_handle_t handle,
+ const sp<RadioCallback>&);
+ static const sp<IRadioService>& getRadioService();
+
+ Mutex mLock;
+ sp<IRadio> mIRadio;
+ const radio_handle_t mHandle;
+ sp<RadioCallback> mCallback;
+};
+
+}; // namespace android
+
+#endif //ANDROID_HARDWARE_RADIO_H
diff --git a/include/radio/RadioCallback.h b/include/radio/RadioCallback.h
new file mode 100644
index 0000000..4a7f1a6
--- /dev/null
+++ b/include/radio/RadioCallback.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2015 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_RADIO_CALLBACK_H
+#define ANDROID_HARDWARE_RADIO_CALLBACK_H
+
+#include <utils/RefBase.h>
+#include <system/radio.h>
+
+namespace android {
+
+class RadioCallback : public RefBase
+{
+public:
+
+ RadioCallback() {}
+ virtual ~RadioCallback() {}
+
+ virtual void onEvent(struct radio_event *event) = 0;
+
+};
+
+}; // namespace android
+
+#endif //ANDROID_HARDWARE_RADIO_CALLBACK_H
diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c
index 6d30d64..c310fe2 100644
--- a/media/libeffects/factory/EffectsFactory.c
+++ b/media/libeffects/factory/EffectsFactory.c
@@ -28,6 +28,7 @@
static list_elem_t *gEffectList; // list of effect_entry_t: all currently created effects
static list_elem_t *gLibraryList; // list of lib_entry_t: all currently loaded libraries
+static list_elem_t *gSkippedEffects; // list of effects skipped because of duplicate uuid
// list of effect_descriptor and list of sub effects : all currently loaded
// It does not contain effects without sub effects.
static list_sub_elem_t *gSubEffectList;
@@ -63,10 +64,10 @@
lib_entry_t **lib,
effect_descriptor_t **desc);
// To search a subeffect in the gSubEffectList
-int findSubEffect(const effect_uuid_t *uuid,
+static int findSubEffect(const effect_uuid_t *uuid,
lib_entry_t **lib,
effect_descriptor_t **desc);
-static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len);
+static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len, int indent);
static int stringToUuid(const char *str, effect_uuid_t *uuid);
static int uuidToString(const effect_uuid_t *uuid, char *str, size_t maxLen);
@@ -237,8 +238,8 @@
}
#if (LOG_NDEBUG == 0)
- char str[256];
- dumpEffectDescriptor(pDescriptor, str, 256);
+ char str[512];
+ dumpEffectDescriptor(pDescriptor, str, sizeof(str), 0 /* indent */);
ALOGV("EffectQueryEffect() desc:%s", str);
#endif
pthread_mutex_unlock(&gLibLock);
@@ -503,15 +504,31 @@
audio_effect_library_t *desc;
list_elem_t *e;
lib_entry_t *l;
+ char path[PATH_MAX];
+ char *str;
+ size_t len;
node = config_find(root, PATH_TAG);
if (node == NULL) {
return -EINVAL;
}
+ // audio_effects.conf always specifies 32 bit lib path: convert to 64 bit path if needed
+ strlcpy(path, node->value, PATH_MAX);
+#ifdef __LP64__
+ str = strstr(path, "/lib/");
+ if (str == NULL)
+ return -EINVAL;
+ len = str - path;
+ path[len] = '\0';
+ strlcat(path, "/lib64/", PATH_MAX);
+ strlcat(path, node->value + len + strlen("/lib/"), PATH_MAX);
+#endif
+ if (strlen(path) >= PATH_MAX - 1)
+ return -EINVAL;
- hdl = dlopen(node->value, RTLD_NOW);
+ hdl = dlopen(path, RTLD_NOW);
if (hdl == NULL) {
- ALOGW("loadLibrary() failed to open %s", node->value);
+ ALOGW("loadLibrary() failed to open %s", path);
goto error;
}
@@ -535,7 +552,7 @@
// add entry for library in gLibraryList
l = malloc(sizeof(lib_entry_t));
l->name = strndup(name, PATH_MAX);
- l->path = strndup(node->value, PATH_MAX);
+ l->path = strndup(path, PATH_MAX);
l->handle = hdl;
l->desc = desc;
l->effects = NULL;
@@ -547,7 +564,7 @@
e->next = gLibraryList;
gLibraryList = e;
pthread_mutex_unlock(&gLibLock);
- ALOGV("getLibrary() linked library %p for path %s", l, node->value);
+ ALOGV("getLibrary() linked library %p for path %s", l, path);
return 0;
@@ -595,8 +612,8 @@
return -EINVAL;
}
#if (LOG_NDEBUG==0)
- char s[256];
- dumpEffectDescriptor(d, s, 256);
+ char s[512];
+ dumpEffectDescriptor(d, s, sizeof(s), 0 /* indent */);
ALOGV("addSubEffect() read descriptor %p:%s",d, s);
#endif
if (EFFECT_API_VERSION_MAJOR(d->apiVersion) !=
@@ -660,6 +677,13 @@
ALOGW("loadEffect() invalid uuid %s", node->value);
return -EINVAL;
}
+ lib_entry_t *tmp;
+ bool skip = false;
+ if (findEffect(NULL, &uuid, &tmp, NULL) == 0) {
+ ALOGW("skipping duplicate uuid %s %s", node->value,
+ node->next ? "and its sub-effects" : "");
+ skip = true;
+ }
d = malloc(sizeof(effect_descriptor_t));
if (l->desc->get_descriptor(&uuid, d) != 0) {
@@ -670,8 +694,8 @@
return -EINVAL;
}
#if (LOG_NDEBUG==0)
- char s[256];
- dumpEffectDescriptor(d, s, 256);
+ char s[512];
+ dumpEffectDescriptor(d, s, sizeof(s), 0 /* indent */);
ALOGV("loadEffect() read descriptor %p:%s",d, s);
#endif
if (EFFECT_API_VERSION_MAJOR(d->apiVersion) !=
@@ -682,8 +706,14 @@
}
e = malloc(sizeof(list_elem_t));
e->object = d;
- e->next = l->effects;
- l->effects = e;
+ if (skip) {
+ e->next = gSkippedEffects;
+ gSkippedEffects = e;
+ return -EINVAL;
+ } else {
+ e->next = l->effects;
+ l->effects = e;
+ }
// After the UUID node in the config_tree, if node->next is valid,
// that would be sub effect node.
@@ -876,22 +906,30 @@
return ret;
}
-void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len) {
+void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len, int indent) {
char s[256];
+ char ss[256];
+ char idt[indent + 1];
- snprintf(str, len, "\nEffect Descriptor %p:\n", desc);
- strncat(str, "- TYPE: ", len);
- uuidToString(&desc->uuid, s, 256);
- snprintf(str, len, "- UUID: %s\n", s);
- uuidToString(&desc->type, s, 256);
- snprintf(str, len, "- TYPE: %s\n", s);
- sprintf(s, "- apiVersion: %08X\n- flags: %08X\n",
- desc->apiVersion, desc->flags);
- strncat(str, s, len);
- sprintf(s, "- name: %s\n", desc->name);
- strncat(str, s, len);
- sprintf(s, "- implementor: %s\n", desc->implementor);
- strncat(str, s, len);
+ memset(idt, ' ', indent);
+ idt[indent] = 0;
+
+ str[0] = 0;
+
+ snprintf(s, sizeof(s), "%s%s / %s\n", idt, desc->name, desc->implementor);
+ strlcat(str, s, len);
+
+ uuidToString(&desc->uuid, s, sizeof(s));
+ snprintf(ss, sizeof(ss), "%s UUID: %s\n", idt, s);
+ strlcat(str, ss, len);
+
+ uuidToString(&desc->type, s, sizeof(s));
+ snprintf(ss, sizeof(ss), "%s TYPE: %s\n", idt, s);
+ strlcat(str, ss, len);
+
+ sprintf(s, "%s apiVersion: %08X\n%s flags: %08X\n", idt,
+ desc->apiVersion, idt, desc->flags);
+ strlcat(str, s, len);
}
int stringToUuid(const char *str, effect_uuid_t *uuid)
@@ -934,3 +972,40 @@
return 0;
}
+int EffectDumpEffects(int fd) {
+ char s[512];
+ list_elem_t *e = gLibraryList;
+ lib_entry_t *l = NULL;
+ effect_descriptor_t *d = NULL;
+ int found = 0;
+ int ret = 0;
+
+ while (e) {
+ l = (lib_entry_t *)e->object;
+ list_elem_t *efx = l->effects;
+ dprintf(fd, "Library %s\n", l->name);
+ if (!efx) {
+ dprintf(fd, " (no effects)\n");
+ }
+ while (efx) {
+ d = (effect_descriptor_t *)efx->object;
+ dumpEffectDescriptor(d, s, sizeof(s), 2);
+ dprintf(fd, "%s", s);
+ efx = efx->next;
+ }
+ e = e->next;
+ }
+
+ e = gSkippedEffects;
+ if (e) {
+ dprintf(fd, "Skipped effects\n");
+ while(e) {
+ d = (effect_descriptor_t *)e->object;
+ dumpEffectDescriptor(d, s, sizeof(s), 2 /* indent */);
+ dprintf(fd, "%s", s);
+ e = e->next;
+ }
+ }
+ return ret;
+}
+
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 6c585fb..3b260d6 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -36,6 +36,8 @@
IMediaRecorder.cpp \
IRemoteDisplay.cpp \
IRemoteDisplayClient.cpp \
+ IResourceManagerClient.cpp \
+ IResourceManagerService.cpp \
IStreamSource.cpp \
MediaCodecInfo.cpp \
Metadata.cpp \
@@ -53,6 +55,8 @@
CharacterEncodingDetector.cpp \
IMediaDeathNotifier.cpp \
MediaProfiles.cpp \
+ MediaResource.cpp \
+ MediaResourcePolicy.cpp \
IEffect.cpp \
IEffectClient.cpp \
AudioEffect.cpp \
@@ -61,15 +65,11 @@
StringArray.cpp \
AudioPolicy.cpp
-LOCAL_SRC_FILES += ../libnbaio/roundup.c
-
LOCAL_SHARED_LIBRARIES := \
libui liblog libcutils libutils libbinder libsonivox libicuuc libicui18n libexpat \
libcamera_client libstagefright_foundation \
libgui libdl libaudioutils libnbaio
-LOCAL_STATIC_LIBRARIES += libinstantssq
-
LOCAL_WHOLE_STATIC_LIBRARIES := libmedia_helper
LOCAL_MODULE:= libmedia
@@ -85,12 +85,3 @@
include $(BUILD_SHARED_LIBRARY)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES += SingleStateQueue.cpp
-LOCAL_CFLAGS += -DSINGLE_STATE_QUEUE_INSTANTIATIONS='"SingleStateQueueInstantiations.cpp"'
-
-LOCAL_MODULE := libinstantssq
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index af103c1..7d8222f 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -486,4 +486,4 @@
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/AudioParameter.cpp b/media/libmedia/AudioParameter.cpp
index 33dbf0b..8c8cf45 100644
--- a/media/libmedia/AudioParameter.cpp
+++ b/media/libmedia/AudioParameter.cpp
@@ -180,4 +180,4 @@
}
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/AudioPolicy.cpp b/media/libmedia/AudioPolicy.cpp
index d2d0971..c7dafcb 100644
--- a/media/libmedia/AudioPolicy.cpp
+++ b/media/libmedia/AudioPolicy.cpp
@@ -112,4 +112,4 @@
return NO_ERROR;
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 07ca14f..100a914 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -112,7 +112,9 @@
mCblkMemory.clear();
mBufferMemory.clear();
IPCThreadState::self()->flushCommands();
- AudioSystem::releaseAudioSessionId(mSessionId, -1);
+ ALOGV("~AudioRecord, releasing session id %d",
+ mSessionId);
+ AudioSystem::releaseAudioSessionId(mSessionId, -1 /*pid*/);
}
}
@@ -159,8 +161,6 @@
}
mTransfer = transferType;
- AutoMutex lock(mLock);
-
// invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
ALOGE("Track already in use");
@@ -233,6 +233,7 @@
if (cbf != NULL) {
mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
+ // thread begins in paused state, and will not reference us until start()
}
// create the IAudioRecord
@@ -286,7 +287,6 @@
status_t status = NO_ERROR;
if (!(flags & CBLK_INVALID)) {
- ALOGV("mAudioRecord->start()");
status = mAudioRecord->start(event, triggerSession);
if (status == DEAD_OBJECT) {
flags |= CBLK_INVALID;
@@ -352,6 +352,10 @@
mMarkerPosition = marker;
mMarkerReached = false;
+ sp<AudioRecordThread> t = mAudioRecordThread;
+ if (t != 0) {
+ t->wake();
+ }
return NO_ERROR;
}
@@ -378,6 +382,10 @@
mNewPosition = mProxy->getPosition() + updatePeriod;
mUpdatePeriod = updatePeriod;
+ sp<AudioRecordThread> t = mAudioRecordThread;
+ if (t != 0) {
+ t->wake();
+ }
return NO_ERROR;
}
@@ -408,7 +416,7 @@
uint32_t AudioRecord::getInputFramesLost() const
{
// no need to check mActive, because if inactive this will return 0, which is what we want
- return AudioSystem::getInputFramesLost(getInput());
+ return AudioSystem::getInputFramesLost(getInputPrivate());
}
// -------------------------------------------------------------------------
@@ -416,7 +424,6 @@
// must be called with mLock held
status_t AudioRecord::openRecord_l(size_t epoch)
{
- status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
ALOGE("Could not get audioflinger");
@@ -431,12 +438,16 @@
}
// Client can only express a preference for FAST. Server will perform additional tests.
- if ((mFlags & AUDIO_INPUT_FLAG_FAST) && !(
- // use case: callback transfer mode
- (mTransfer == TRANSFER_CALLBACK) &&
+ if ((mFlags & AUDIO_INPUT_FLAG_FAST) && !((
+ // either of these use cases:
+ // use case 1: callback transfer mode
+ (mTransfer == TRANSFER_CALLBACK) ||
+ // use case 2: obtain/release mode
+ (mTransfer == TRANSFER_OBTAIN)) &&
// matching sample rate
(mSampleRate == afSampleRate))) {
- ALOGW("AUDIO_INPUT_FLAG_FAST denied by client");
+ ALOGW("AUDIO_INPUT_FLAG_FAST denied by client; transfer %d, track %u Hz, primary %u Hz",
+ mTransfer, mSampleRate, afSampleRate);
// once denied, do not request again if IAudioRecord is re-created
mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
}
@@ -452,7 +463,8 @@
}
audio_io_handle_t input;
- status = AudioSystem::getInputForAttr(&mAttributes, &input, (audio_session_t)mSessionId,
+ status_t status = AudioSystem::getInputForAttr(&mAttributes, &input,
+ (audio_session_t)mSessionId,
mSampleRate, mFormat, mChannelMask, mFlags);
if (status != NO_ERROR) {
@@ -684,9 +696,9 @@
return status;
}
-void AudioRecord::releaseBuffer(Buffer* audioBuffer)
+void AudioRecord::releaseBuffer(const Buffer* audioBuffer)
{
- // all TRANSFER_* are valid
+ // FIXME add error checking on mode, by adding an internal version
size_t stepCount = audioBuffer->size / mFrameSize;
if (stepCount == 0) {
@@ -704,7 +716,7 @@
// the server does not automatically disable recorder on overrun, so no need to restart
}
-audio_io_handle_t AudioRecord::getInput() const
+audio_io_handle_t AudioRecord::getInputPrivate() const
{
AutoMutex lock(mLock);
return mInput;
@@ -712,7 +724,7 @@
// -------------------------------------------------------------------------
-ssize_t AudioRecord::read(void* buffer, size_t userSize)
+ssize_t AudioRecord::read(void* buffer, size_t userSize, bool blocking)
{
if (mTransfer != TRANSFER_SYNC) {
return INVALID_OPERATION;
@@ -731,7 +743,8 @@
while (userSize >= mFrameSize) {
audioBuffer.frameCount = userSize / mFrameSize;
- status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever);
+ status_t err = obtainBuffer(&audioBuffer,
+ blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
if (err < 0) {
if (read > 0) {
break;
@@ -863,8 +876,11 @@
if (!markerReached && position < markerPosition) {
minFrames = markerPosition - position;
}
- if (updatePeriod > 0 && updatePeriod < minFrames) {
- minFrames = updatePeriod;
+ if (updatePeriod > 0) {
+ uint32_t remaining = newPosition - position;
+ if (remaining < minFrames) {
+ minFrames = remaining;
+ }
}
// If > 0, poll periodically to recover from a stuck server. A good value is 2.
@@ -990,14 +1006,13 @@
{
ALOGW("dead IAudioRecord, creating a new one from %s()", from);
++mSequence;
- status_t result;
// if the new IAudioRecord is created, openRecord_l() will modify the
// following member variables: mAudioRecord, mCblkMemory, mCblk, mBufferMemory.
// It will also delete the strong references on previous IAudioRecord and IMemory
size_t position = mProxy->getPosition();
mNewPosition = position + mUpdatePeriod;
- result = openRecord_l(position);
+ status_t result = openRecord_l(position);
if (result == NO_ERROR) {
if (mActive) {
// callback thread or sync event hasn't changed
@@ -1069,8 +1084,8 @@
case NS_NEVER:
return false;
case NS_WHENEVER:
- // FIXME increase poll interval, or make event-driven
- ns = 1000000000LL;
+ // Event driven: call wake() when callback notifications conditions change.
+ ns = INT64_MAX;
// fall through
default:
LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
@@ -1103,6 +1118,17 @@
}
}
+void AudioRecord::AudioRecordThread::wake()
+{
+ AutoMutex _l(mMyLock);
+ if (!mPaused && mPausedInt && mPausedNs > 0) {
+ // audio record is active and internally paused with timeout.
+ mIgnoreNextPausedInt = true;
+ mPausedInt = false;
+ mMyCond.signal();
+ }
+}
+
void AudioRecord::AudioRecordThread::pauseInternal(nsecs_t ns)
{
AutoMutex _l(mMyLock);
@@ -1112,4 +1138,4 @@
// -------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 9cae21c..9150a94 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -34,7 +34,6 @@
Mutex AudioSystem::gLock;
Mutex AudioSystem::gLockCache;
Mutex AudioSystem::gLockAPS;
-Mutex AudioSystem::gLockAPC;
sp<IAudioFlinger> AudioSystem::gAudioFlinger;
sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
@@ -48,8 +47,6 @@
audio_channel_mask_t AudioSystem::gPrevInChannelMask;
size_t AudioSystem::gInBuffSize = 0; // zero indicates cache is invalid
-sp<AudioSystem::AudioPortCallback> AudioSystem::gAudioPortCallback;
-
// establish binder interface to AudioFlinger service
const sp<IAudioFlinger> AudioSystem::get_audio_flinger()
{
@@ -499,8 +496,8 @@
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
gOutputs.add(ioHandle, outputDesc);
- ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %zu "
- "latency %d",
+ ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x "
+ "frameCount %zu latency %d",
outputDesc->samplingRate, outputDesc->format, outputDesc->channelMask,
outputDesc->frameCount, outputDesc->latency);
} break;
@@ -523,8 +520,8 @@
if (param2 == NULL) break;
desc = (const OutputDescriptor *)param2;
- ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x channel mask %#x "
- "frameCount %zu latency %d",
+ ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x "
+ "channel mask %#x frameCount %zu latency %d",
ioHandle, desc->samplingRate, desc->format,
desc->channelMask, desc->frameCount, desc->latency);
OutputDescriptor *outputDesc = gOutputs.valueAt(index);
@@ -590,18 +587,22 @@
status_t AudioSystem::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address)
+ const char *device_address,
+ const char *device_name)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
const char *address = "";
+ const char *name = "";
if (aps == 0) return PERMISSION_DENIED;
if (device_address != NULL) {
address = device_address;
}
-
- return aps->setDeviceConnectionState(device, state, address);
+ if (device_name != NULL) {
+ name = device_name;
+ }
+ return aps->setDeviceConnectionState(device, state, address, name);
}
audio_policy_dev_state_t AudioSystem::getDeviceConnectionState(audio_devices_t device,
@@ -869,7 +870,6 @@
Mutex::Autolock _l(gLockAPS);
gAudioPolicyService.clear();
}
- // Do not clear gAudioPortCallback
}
bool AudioSystem::isOffloadSupported(const audio_offload_info_t& info)
@@ -929,12 +929,31 @@
return aps->setAudioPortConfig(config);
}
-void AudioSystem::setAudioPortCallback(sp<AudioPortCallback> callBack)
+status_t AudioSystem::addAudioPortCallback(const sp<AudioPortCallback>& callBack)
{
- Mutex::Autolock _l(gLockAPC);
- gAudioPortCallback = callBack;
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) return PERMISSION_DENIED;
+
+ Mutex::Autolock _l(gLockAPS);
+ if (gAudioPolicyServiceClient == 0) {
+ return NO_INIT;
+ }
+ return gAudioPolicyServiceClient->addAudioPortCallback(callBack);
}
+status_t AudioSystem::removeAudioPortCallback(const sp<AudioPortCallback>& callBack)
+{
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) return PERMISSION_DENIED;
+
+ Mutex::Autolock _l(gLockAPS);
+ if (gAudioPolicyServiceClient == 0) {
+ return NO_INIT;
+ }
+ return gAudioPolicyServiceClient->removeAudioPortCallback(callBack);
+}
+
+
status_t AudioSystem::acquireSoundTriggerSession(audio_session_t *session,
audio_io_handle_t *ioHandle,
audio_devices_t *device)
@@ -967,12 +986,58 @@
// ---------------------------------------------------------------------------
+status_t AudioSystem::AudioPolicyServiceClient::addAudioPortCallback(
+ const sp<AudioPortCallback>& callBack)
+{
+ Mutex::Autolock _l(mLock);
+ for (size_t i = 0; i < mAudioPortCallbacks.size(); i++) {
+ if (mAudioPortCallbacks[i] == callBack) {
+ return INVALID_OPERATION;
+ }
+ }
+ mAudioPortCallbacks.add(callBack);
+ return NO_ERROR;
+}
+
+status_t AudioSystem::AudioPolicyServiceClient::removeAudioPortCallback(
+ const sp<AudioPortCallback>& callBack)
+{
+ Mutex::Autolock _l(mLock);
+ size_t i;
+ for (i = 0; i < mAudioPortCallbacks.size(); i++) {
+ if (mAudioPortCallbacks[i] == callBack) {
+ break;
+ }
+ }
+ if (i == mAudioPortCallbacks.size()) {
+ return INVALID_OPERATION;
+ }
+ mAudioPortCallbacks.removeAt(i);
+ return NO_ERROR;
+}
+
+void AudioSystem::AudioPolicyServiceClient::onAudioPortListUpdate()
+{
+ Mutex::Autolock _l(mLock);
+ for (size_t i = 0; i < mAudioPortCallbacks.size(); i++) {
+ mAudioPortCallbacks[i]->onAudioPortListUpdate();
+ }
+}
+
+void AudioSystem::AudioPolicyServiceClient::onAudioPatchListUpdate()
+{
+ Mutex::Autolock _l(mLock);
+ for (size_t i = 0; i < mAudioPortCallbacks.size(); i++) {
+ mAudioPortCallbacks[i]->onAudioPatchListUpdate();
+ }
+}
+
void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who __unused)
{
{
- Mutex::Autolock _l(gLockAPC);
- if (gAudioPortCallback != 0) {
- gAudioPortCallback->onServiceDied();
+ Mutex::Autolock _l(mLock);
+ for (size_t i = 0; i < mAudioPortCallbacks.size(); i++) {
+ mAudioPortCallbacks[i]->onServiceDied();
}
}
{
@@ -983,20 +1048,4 @@
ALOGW("AudioPolicyService server died!");
}
-void AudioSystem::AudioPolicyServiceClient::onAudioPortListUpdate()
-{
- Mutex::Autolock _l(gLockAPC);
- if (gAudioPortCallback != 0) {
- gAudioPortCallback->onAudioPortListUpdate();
- }
-}
-
-void AudioSystem::AudioPolicyServiceClient::onAudioPatchListUpdate()
-{
- Mutex::Autolock _l(gLockAPC);
- if (gAudioPortCallback != 0) {
- gAudioPortCallback->onAudioPatchListUpdate();
- }
-}
-
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 735db5c..ce30c62 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -33,11 +33,16 @@
#define WAIT_PERIOD_MS 10
#define WAIT_STREAM_END_TIMEOUT_SEC 120
-
+static const int kMaxLoopCountNotifications = 32;
namespace android {
// ---------------------------------------------------------------------------
+template <typename T>
+const T &min(const T &x, const T &y) {
+ return x < y ? x : y;
+}
+
static int64_t convertTimespecToUs(const struct timespec &tv)
{
return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000;
@@ -61,12 +66,11 @@
return BAD_VALUE;
}
- // FIXME merge with similar code in createTrack_l(), except we're missing
- // some information here that is available in createTrack_l():
+ // FIXME handle in server, like createTrack_l(), possible missing info:
// audio_io_handle_t output
// audio_format_t format
// audio_channel_mask_t channelMask
- // audio_output_flags_t flags
+ // audio_output_flags_t flags (FAST)
uint32_t afSampleRate;
status_t status;
status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
@@ -96,16 +100,16 @@
minBufCount = 2;
}
- *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
- afFrameCount * minBufCount * uint64_t(sampleRate) / afSampleRate;
- // The formula above should always produce a non-zero value, but return an error
- // in the unlikely event that it does not, as that's part of the API contract.
+ *frameCount = minBufCount * sourceFramesNeeded(sampleRate, afFrameCount, afSampleRate);
+ // The formula above should always produce a non-zero value under normal circumstances:
+ // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX.
+ // Return error in the unlikely event that it does not, as that's part of the API contract.
if (*frameCount == 0) {
- ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d",
+ ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %u",
streamType, sampleRate);
return BAD_VALUE;
}
- ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d",
+ ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%u, afSampleRate=%u, afLatency=%u",
*frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
return NO_ERROR;
}
@@ -199,8 +203,8 @@
mCblkMemory.clear();
mSharedBuffer.clear();
IPCThreadState::self()->flushCommands();
- ALOGV("~AudioTrack, releasing session id from %d on behalf of %d",
- IPCThreadState::self()->getCallingPid(), mClientPid);
+ ALOGV("~AudioTrack, releasing session id %d from %d on behalf of %d",
+ mSessionId, IPCThreadState::self()->getCallingPid(), mClientPid);
AudioSystem::releaseAudioSessionId(mSessionId, mClientPid);
}
}
@@ -225,9 +229,9 @@
const audio_attributes_t* pAttributes)
{
ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
- "flags #%x, notificationFrames %u, sessionId %d, transferType %d",
+ "flags #%x, notificationFrames %u, sessionId %d, transferType %d, uid %d, pid %d",
streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
- sessionId, transferType);
+ sessionId, transferType, uid, pid);
switch (transferType) {
case TRANSFER_DEFAULT:
@@ -270,8 +274,6 @@
ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags);
- AutoMutex lock(mLock);
-
// invariant that mAudioTrack != 0 is true only after set() returns successfully
if (mAudioTrack != 0) {
ALOGE("Track already in use");
@@ -295,6 +297,9 @@
ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]",
mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
mStreamType = AUDIO_STREAM_DEFAULT;
+ if ((mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
+ }
}
// these below should probably come from the audioFlinger too...
@@ -317,12 +322,6 @@
uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
- // AudioFlinger does not currently support 8-bit data in shared memory
- if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
- ALOGE("8-bit data in shared memory is not supported");
- return BAD_VALUE;
- }
-
// force direct flag if format is not linear PCM
// or offload was requested
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
@@ -346,12 +345,9 @@
} else {
mFrameSize = sizeof(uint8_t);
}
- mFrameSizeAF = mFrameSize;
} else {
ALOG_ASSERT(audio_is_linear_pcm(format));
mFrameSize = channelCount * audio_bytes_per_sample(format);
- mFrameSizeAF = channelCount * audio_bytes_per_sample(
- format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format);
// createTrack will return an error if PCM format is not supported by server,
// so no need to check for specific PCM formats here
}
@@ -403,6 +399,7 @@
if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
+ // thread begins in paused state, and will not reference us until start()
}
// create the IAudioTrack
@@ -420,7 +417,10 @@
mStatus = NO_ERROR;
mState = STATE_STOPPED;
mUserData = user;
- mLoopPeriod = 0;
+ mLoopCount = 0;
+ mLoopStart = 0;
+ mLoopEnd = 0;
+ mLoopCountNotified = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
@@ -531,14 +531,12 @@
// the playback head position will reset to 0, so if a marker is set, we need
// to activate it again
mMarkerReached = false;
-#if 0
- // Force flush if a shared buffer is used otherwise audioflinger
- // will not stop before end of buffer is reached.
- // It may be needed to make sure that we stop playback, likely in case looping is on.
+
if (mSharedBuffer != 0) {
- flush_l();
+ // clear buffer position and loop count.
+ mStaticProxy->setBufferPositionAndLoop(0 /* position */,
+ 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
}
-#endif
sp<AudioTrackThread> t = mAudioTrackThread;
if (t != 0) {
@@ -669,14 +667,18 @@
status_t AudioTrack::setSampleRate(uint32_t rate)
{
- if (mIsTimed || isOffloadedOrDirect()) {
+ AutoMutex lock(mLock);
+ if (rate == mSampleRate) {
+ return NO_ERROR;
+ }
+ if (mIsTimed || isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
return INVALID_OPERATION;
}
-
- AutoMutex lock(mLock);
if (mOutput == AUDIO_IO_HANDLE_NONE) {
return NO_INIT;
}
+ // NOTE: it is theoretically possible, but highly unlikely, that a device change
+ // could mean a previously allowed sampling rate is no longer allowed.
uint32_t afSamplingRate;
if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
return NO_INIT;
@@ -740,10 +742,15 @@
void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
- // Setting the loop will reset next notification update period (like setPosition).
- mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
- mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0;
+ // We do not update the periodic notification point.
+ // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
+ mLoopCount = loopCount;
+ mLoopEnd = loopEnd;
+ mLoopStart = loopStart;
+ mLoopCountNotified = loopCount;
mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
+
+ // Waking the AudioTrackThread is not needed as this cannot be called when active.
}
status_t AudioTrack::setMarkerPosition(uint32_t marker)
@@ -757,6 +764,10 @@
mMarkerPosition = marker;
mMarkerReached = false;
+ sp<AudioTrackThread> t = mAudioTrackThread;
+ if (t != 0) {
+ t->wake();
+ }
return NO_ERROR;
}
@@ -786,6 +797,10 @@
mNewPosition = updateAndGetPosition_l() + updatePeriod;
mUpdatePeriod = updatePeriod;
+ sp<AudioTrackThread> t = mAudioTrackThread;
+ if (t != 0) {
+ t->wake();
+ }
return NO_ERROR;
}
@@ -823,12 +838,11 @@
if (mState == STATE_ACTIVE) {
return INVALID_OPERATION;
}
+ // After setting the position, use full update period before notification.
mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
- mLoopPeriod = 0;
- // FIXME Check whether loops and setting position are incompatible in old code.
- // If we use setLoop for both purposes we lose the capability to set the position while looping.
- mStaticProxy->setLoop(position, mFrameCount, 0);
+ mStaticProxy->setBufferPosition(position);
+ // Waking the AudioTrackThread is not needed as this cannot be called when active.
return NO_ERROR;
}
@@ -893,10 +907,18 @@
return INVALID_OPERATION;
}
mNewPosition = mUpdatePeriod;
- mLoopPeriod = 0;
- // FIXME The new code cannot reload while keeping a loop specified.
- // Need to check how the old code handled this, and whether it's a significant change.
- mStaticProxy->setLoop(0, mFrameCount, 0);
+ (void) updateAndGetPosition_l();
+ mPosition = 0;
+#if 0
+ // The documentation is not clear on the behavior of reload() and the restoration
+ // of loop count. Historically we have not restored loop count, start, end,
+ // but it makes sense if one desires to repeat playing a particular sound.
+ if (mLoopCount != 0) {
+ mLoopCountNotified = mLoopCount;
+ mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
+ }
+#endif
+ mStaticProxy->setBufferPosition(0);
return NO_ERROR;
}
@@ -945,9 +967,9 @@
if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) {
- ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x,"
+ ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x,"
" channel mask %#x, flags %#x",
- streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
+ mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
return BAD_VALUE;
}
{
@@ -962,6 +984,7 @@
ALOGE("getLatency(%d) failed status %d", output, status);
goto release;
}
+ ALOGV("createTrack_l() output %d afLatency %u", output, afLatency);
size_t afFrameCount;
status = AudioSystem::getFrameCount(output, &afFrameCount);
@@ -986,23 +1009,23 @@
// use case 1: shared buffer
(mSharedBuffer != 0) ||
// use case 2: callback transfer mode
- (mTransfer == TRANSFER_CALLBACK)) &&
+ (mTransfer == TRANSFER_CALLBACK) ||
+ // use case 3: obtain/release mode
+ (mTransfer == TRANSFER_OBTAIN)) &&
// matching sample rate
(mSampleRate == afSampleRate))) {
- ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
+ ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, track %u Hz, output %u Hz",
+ mTransfer, mSampleRate, afSampleRate);
// once denied, do not request again if IAudioTrack is re-created
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
- ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
// The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
// n = 1 fast track with single buffering; nBuffering is ignored
// n = 2 fast track with double buffering
- // n = 2 normal track, no sample rate conversion
- // n = 3 normal track, with sample rate conversion
- // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)
- // n > 3 very high latency or very small notification interval; nBuffering is ignored
- const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3;
+ // n = 2 normal track, (including those with sample rate conversion)
+ // n >= 3 very high latency or very small notification interval (unused).
+ const uint32_t nBuffering = 2;
mNotificationFramesAct = mNotificationFramesReq;
@@ -1019,12 +1042,12 @@
mNotificationFramesAct = frameCount;
}
} else if (mSharedBuffer != 0) {
-
- // Ensure that buffer alignment matches channel count
- // 8-bit data in shared memory is not currently supported by AudioFlinger
- size_t alignment = audio_bytes_per_sample(
- mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat);
+ // 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) {
@@ -1042,40 +1065,10 @@
// 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() / mFrameSizeAF;
-
- } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
-
- // FIXME move these calculations and associated checks to server
-
- // Ensure that buffer depth covers at least audio hardware latency
- uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
- ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d",
- afFrameCount, minBufCount, afSampleRate, afLatency);
- if (minBufCount <= nBuffering) {
- minBufCount = nBuffering;
- }
-
- size_t minFrameCount = afFrameCount * minBufCount * uint64_t(mSampleRate) / afSampleRate;
- ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
- ", afLatency=%d",
- minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency);
-
- if (frameCount == 0) {
- frameCount = minFrameCount;
- } else if (frameCount < minFrameCount) {
- // not ALOGW because it happens all the time when playing key clicks over A2DP
- ALOGV("Minimum buffer size corrected from %zu to %zu",
- frameCount, minFrameCount);
- frameCount = minFrameCount;
- }
- // Make sure that application is notified with sufficient margin before underrun
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
- mNotificationFramesAct = frameCount/nBuffering;
- }
-
+ frameCount = mSharedBuffer->size() / mFrameSize;
} else {
- // For fast tracks, the frame count calculations and checks are done by server
+ // For fast and normal streaming tracks,
+ // the frame count calculations and checks are done by server
}
IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT;
@@ -1101,12 +1094,10 @@
size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
// but we will still need the original value also
+ int originalSessionId = mSessionId;
sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
mSampleRate,
- // AudioFlinger only sees 16-bit PCM
- mFormat == AUDIO_FORMAT_PCM_8_BIT &&
- !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ?
- AUDIO_FORMAT_PCM_16_BIT : mFormat,
+ mFormat,
mChannelMask,
&temp,
&trackFlags,
@@ -1116,6 +1107,8 @@
&mSessionId,
mClientUid,
&status);
+ ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId,
+ "session ID changed from %d to %d", originalSessionId, mSessionId);
if (status != NO_ERROR) {
ALOGE("AudioFlinger could not create track, status: %d", status);
@@ -1161,23 +1154,10 @@
if (trackFlags & IAudioFlinger::TRACK_FAST) {
ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount);
mAwaitBoost = true;
- if (mSharedBuffer == 0) {
- // Theoretically double-buffering is not required for fast tracks,
- // due to tighter scheduling. But in practice, to accommodate kernels with
- // scheduling jitter, and apps with computation jitter, we use double-buffering.
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
- mNotificationFramesAct = frameCount/nBuffering;
- }
- }
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
// once denied, do not request again if IAudioTrack is re-created
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
- if (mSharedBuffer == 0) {
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
- mNotificationFramesAct = frameCount/nBuffering;
- }
- }
}
}
if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
@@ -1200,6 +1180,16 @@
//return NO_INIT;
}
}
+ // Make sure that application is notified with sufficient margin before underrun
+ if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) {
+ // Theoretically double-buffering is not required for fast tracks,
+ // due to tighter scheduling. But in practice, to accommodate kernels with
+ // scheduling jitter, and apps with computation jitter, we use double-buffering
+ // for fast tracks just like normal streaming tracks.
+ if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount / nBuffering) {
+ mNotificationFramesAct = frameCount / nBuffering;
+ }
+ }
// We retain a copy of the I/O handle, but don't own the reference
mOutput = output;
@@ -1211,9 +1201,13 @@
// address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
void* buffers;
if (mSharedBuffer == 0) {
- buffers = (char*)cblk + sizeof(audio_track_cblk_t);
+ buffers = cblk + 1;
} else {
buffers = mSharedBuffer->pointer();
+ if (buffers == NULL) {
+ ALOGE("Could not get buffer pointer");
+ return NO_INIT;
+ }
}
mAudioTrack->attachAuxEffect(mAuxEffectId);
@@ -1230,9 +1224,9 @@
// update proxy
if (mSharedBuffer == 0) {
mStaticProxy.clear();
- mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
+ mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
} else {
- mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
+ mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
mProxy = mStaticProxy;
}
@@ -1258,7 +1252,7 @@
return status;
}
-status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
+status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
{
if (audioBuffer == NULL) {
return BAD_VALUE;
@@ -1285,7 +1279,7 @@
ALOGE("%s invalid waitCount %d", __func__, waitCount);
requested = NULL;
}
- return obtainBuffer(audioBuffer, requested);
+ return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
@@ -1352,7 +1346,7 @@
} while ((status == DEAD_OBJECT) && (tryCounter-- > 0));
audioBuffer->frameCount = buffer.mFrameCount;
- audioBuffer->size = buffer.mFrameCount * mFrameSizeAF;
+ audioBuffer->size = buffer.mFrameCount * mFrameSize;
audioBuffer->raw = buffer.mRaw;
if (nonContig != NULL) {
*nonContig = buffer.mNonContig;
@@ -1360,13 +1354,14 @@
return status;
}
-void AudioTrack::releaseBuffer(Buffer* audioBuffer)
+void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
{
+ // FIXME add error checking on mode, by adding an internal version
if (mTransfer == TRANSFER_SHARED) {
return;
}
- size_t stepCount = audioBuffer->size / mFrameSizeAF;
+ size_t stepCount = audioBuffer->size / mFrameSize;
if (stepCount == 0) {
return;
}
@@ -1431,15 +1426,8 @@
return ssize_t(err);
}
- size_t toWrite;
- if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
- // Divide capacity by 2 to take expansion into account
- toWrite = audioBuffer.size >> 1;
- memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite);
- } else {
- toWrite = audioBuffer.size;
- memcpy(audioBuffer.i8, buffer, toWrite);
- }
+ size_t toWrite = audioBuffer.size;
+ memcpy(audioBuffer.i8, buffer, toWrite);
buffer = ((const char *) buffer) + toWrite;
userSize -= toWrite;
written += toWrite;
@@ -1559,9 +1547,8 @@
// that the upper layers can recreate the track
if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
status_t status = restoreTrack_l("processAudioBuffer");
- mLock.unlock();
- // Run again immediately, but with a new IAudioTrack
- return 0;
+ // after restoration, continue below to make sure that the loop and buffer events
+ // are notified because they have been cleared from mCblk->mFlags above.
}
}
@@ -1610,7 +1597,6 @@
}
// Cache other fields that will be needed soon
- uint32_t loopPeriod = mLoopPeriod;
uint32_t sampleRate = mSampleRate;
uint32_t notificationFrames = mNotificationFramesAct;
if (mRefreshRemaining) {
@@ -1622,8 +1608,30 @@
uint32_t sequence = mSequence;
sp<AudioTrackClientProxy> proxy = mProxy;
+ // Determine the number of new loop callback(s) that will be needed, while locked.
+ int loopCountNotifications = 0;
+ uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
+
+ if (mLoopCount > 0) {
+ int loopCount;
+ size_t bufferPosition;
+ mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
+ loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
+ loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
+ mLoopCountNotified = loopCount; // discard any excess notifications
+ } else if (mLoopCount < 0) {
+ // FIXME: We're not accurate with notification count and position with infinite looping
+ // since loopCount from server side will always return -1 (we could decrement it).
+ size_t bufferPosition = mStaticProxy->getBufferPosition();
+ loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
+ loopPeriod = mLoopEnd - bufferPosition;
+ } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
+ size_t bufferPosition = mStaticProxy->getBufferPosition();
+ loopPeriod = mFrameCount - bufferPosition;
+ }
+
// These fields don't need to be cached, because they are assigned only by set():
- // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags
+ // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags
// mFlags is also assigned by createTrack_l(), but not the bit we care about.
mLock.unlock();
@@ -1662,10 +1670,9 @@
if (newUnderrun) {
mCbf(EVENT_UNDERRUN, mUserData, NULL);
}
- // FIXME we will miss loops if loop cycle was signaled several times since last call
- // to processAudioBuffer()
- if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) {
+ while (loopCountNotifications > 0) {
mCbf(EVENT_LOOP_END, mUserData, NULL);
+ --loopCountNotifications;
}
if (flags & CBLK_BUFFER_END) {
mCbf(EVENT_BUFFER_END, mUserData, NULL);
@@ -1701,10 +1708,11 @@
minFrames = markerPosition - position;
}
if (loopPeriod > 0 && loopPeriod < minFrames) {
+ // loopPeriod is already adjusted for actual position.
minFrames = loopPeriod;
}
- if (updatePeriod > 0 && updatePeriod < minFrames) {
- minFrames = updatePeriod;
+ if (updatePeriod > 0) {
+ minFrames = min(minFrames, uint32_t(newPosition - position));
}
// If > 0, poll periodically to recover from a stuck server. A good value is 2.
@@ -1767,13 +1775,6 @@
}
}
- // Divide buffer size by 2 to take into account the expansion
- // due to 8 to 16 bit conversion: the callback must fill only half
- // of the destination buffer
- if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
- audioBuffer.size >>= 1;
- }
-
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
size_t writtenSize = audioBuffer.size;
@@ -1793,13 +1794,7 @@
return WAIT_PERIOD_MS * 1000000LL;
}
- if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
- // 8 to 16 bit conversion, note that source and destination are the same address
- memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
- audioBuffer.size <<= 1;
- }
-
- size_t releasedFrames = audioBuffer.size / mFrameSizeAF;
+ size_t releasedFrames = writtenSize / mFrameSize;
audioBuffer.frameCount = releasedFrames;
mRemainingFrames -= releasedFrames;
if (misalignment >= releasedFrames) {
@@ -1844,7 +1839,6 @@
ALOGW("dead IAudioTrack, %s, creating a new one from %s()",
isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
++mSequence;
- status_t result;
// refresh the audio configuration cache in this process to make sure we get new
// output parameters and new IAudioFlinger in createTrack_l()
@@ -1856,39 +1850,39 @@
}
// save the old static buffer position
- size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0;
+ size_t bufferPosition = 0;
+ int loopCount = 0;
+ if (mStaticProxy != 0) {
+ mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
+ }
// If a new IAudioTrack is successfully created, createTrack_l() will modify the
// following member variables: mAudioTrack, mCblkMemory and mCblk.
// It will also delete the strong references on previous IAudioTrack and IMemory.
// If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
- result = createTrack_l();
+ status_t result = createTrack_l();
// 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).
(void) updateAndGetPosition_l();
- mPosition = mReleased;
+ if (mStaticProxy == 0) {
+ mPosition = mReleased;
+ }
if (result == NO_ERROR) {
- // continue playback from last known position, but
- // don't attempt to restore loop after invalidation; it's difficult and not worthwhile
- if (mStaticProxy != NULL) {
- mLoopPeriod = 0;
- mStaticProxy->setLoop(bufferPosition, mFrameCount, 0);
- }
- // FIXME How do we simulate the fact that all frames present in the buffer at the time of
- // track destruction have been played? This is critical for SoundPool implementation
- // This must be broken, and needs to be tested/debugged.
-#if 0
- // restore write index and set other indexes to reflect empty buffer status
- if (!strcmp(from, "start")) {
- // Make sure that a client relying on callback events indicating underrun or
- // the actual amount of audio frames played (e.g SoundPool) receives them.
- if (mSharedBuffer == 0) {
- // restart playback even if buffer is not completely filled.
- android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags);
+ // Continue playback from last known position and restore loop.
+ if (mStaticProxy != 0) {
+ if (loopCount != 0) {
+ mStaticProxy->setBufferPositionAndLoop(bufferPosition,
+ mLoopStart, mLoopEnd, loopCount);
+ } else {
+ mStaticProxy->setBufferPosition(bufferPosition);
+ if (bufferPosition == mFrameCount) {
+ ALOGD("restoring track at end of static buffer");
+ }
}
}
-#endif
if (mState == STATE_ACTIVE) {
result = mAudioTrack->start();
}
@@ -2148,8 +2142,8 @@
case NS_NEVER:
return false;
case NS_WHENEVER:
- // FIXME increase poll interval, or make event-driven
- ns = 1000000000LL;
+ // Event driven: call wake() when callback notifications conditions change.
+ ns = INT64_MAX;
// fall through
default:
LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
@@ -2182,6 +2176,17 @@
}
}
+void AudioTrack::AudioTrackThread::wake()
+{
+ AutoMutex _l(mMyLock);
+ if (!mPaused && mPausedInt && mPausedNs > 0) {
+ // audio track is active and internally paused with timeout.
+ mIgnoreNextPausedInt = true;
+ mPausedInt = false;
+ mMyCond.signal();
+ }
+}
+
void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
{
AutoMutex _l(mMyLock);
@@ -2189,4 +2194,4 @@
mPausedNs = ns;
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index ff24475..6d5f1af 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -28,7 +28,21 @@
// used to clamp a value to size_t. TODO: move to another file.
template <typename T>
size_t clampToSize(T x) {
- return x > SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x;
+ return sizeof(T) > sizeof(size_t) && x > (T) SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x;
+}
+
+// incrementSequence is used to determine the next sequence value
+// for the loop and position sequence counters. It should return
+// a value between "other" + 1 and "other" + INT32_MAX, the choice of
+// which needs to be the "least recently used" sequence value for "self".
+// In general, this means (new_self) returned is max(self, other) + 1.
+
+static uint32_t incrementSequence(uint32_t self, uint32_t other) {
+ int32_t diff = self - other;
+ if (diff >= 0 && diff < INT32_MAX) {
+ return self + 1; // we're already ahead of other.
+ }
+ return other + 1; // we're behind, so move just ahead of other.
}
audio_track_cblk_t::audio_track_cblk_t()
@@ -409,7 +423,6 @@
goto end;
}
// check for obtainBuffer interrupted by client
- // check for obtainBuffer interrupted by client
if (flags & CBLK_INTERRUPT) {
ALOGV("waitStreamEndDone() interrupted by client");
status = -EINTR;
@@ -485,8 +498,11 @@
StaticAudioTrackClientProxy::StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers,
size_t frameCount, size_t frameSize)
: AudioTrackClientProxy(cblk, buffers, frameCount, frameSize),
- mMutator(&cblk->u.mStatic.mSingleStateQueue), mBufferPosition(0)
+ mMutator(&cblk->u.mStatic.mSingleStateQueue),
+ mPosLoopObserver(&cblk->u.mStatic.mPosLoopQueue)
{
+ memset(&mState, 0, sizeof(mState));
+ memset(&mPosLoop, 0, sizeof(mPosLoop));
}
void StaticAudioTrackClientProxy::flush()
@@ -501,30 +517,72 @@
// FIXME Should return an error status
return;
}
- StaticAudioTrackState newState;
- newState.mLoopStart = (uint32_t) loopStart;
- newState.mLoopEnd = (uint32_t) loopEnd;
- newState.mLoopCount = loopCount;
- size_t bufferPosition;
- if (loopCount == 0 || (bufferPosition = getBufferPosition()) >= loopEnd) {
- bufferPosition = loopStart;
+ mState.mLoopStart = (uint32_t) loopStart;
+ mState.mLoopEnd = (uint32_t) loopEnd;
+ mState.mLoopCount = loopCount;
+ mState.mLoopSequence = incrementSequence(mState.mLoopSequence, mState.mPositionSequence);
+ // set patch-up variables until the mState is acknowledged by the ServerProxy.
+ // observed buffer position and loop count will freeze until then to give the
+ // illusion of a synchronous change.
+ getBufferPositionAndLoopCount(NULL, NULL);
+ // preserve behavior to restart at mState.mLoopStart if position exceeds mState.mLoopEnd.
+ if (mState.mLoopCount != 0 && mPosLoop.mBufferPosition >= mState.mLoopEnd) {
+ mPosLoop.mBufferPosition = mState.mLoopStart;
}
- mBufferPosition = bufferPosition; // snapshot buffer position until loop is acknowledged.
- (void) mMutator.push(newState);
+ mPosLoop.mLoopCount = mState.mLoopCount;
+ (void) mMutator.push(mState);
+}
+
+void StaticAudioTrackClientProxy::setBufferPosition(size_t position)
+{
+ // This can only happen on a 64-bit client
+ if (position > UINT32_MAX) {
+ // FIXME Should return an error status
+ return;
+ }
+ mState.mPosition = (uint32_t) position;
+ mState.mPositionSequence = incrementSequence(mState.mPositionSequence, mState.mLoopSequence);
+ // set patch-up variables until the mState is acknowledged by the ServerProxy.
+ // observed buffer position and loop count will freeze until then to give the
+ // illusion of a synchronous change.
+ if (mState.mLoopCount > 0) { // only check if loop count is changing
+ getBufferPositionAndLoopCount(NULL, NULL); // get last position
+ }
+ mPosLoop.mBufferPosition = position;
+ if (position >= mState.mLoopEnd) {
+ // no ongoing loop is possible if position is greater than loopEnd.
+ mPosLoop.mLoopCount = 0;
+ }
+ (void) mMutator.push(mState);
+}
+
+void StaticAudioTrackClientProxy::setBufferPositionAndLoop(size_t position, size_t loopStart,
+ size_t loopEnd, int loopCount)
+{
+ setLoop(loopStart, loopEnd, loopCount);
+ setBufferPosition(position);
}
size_t StaticAudioTrackClientProxy::getBufferPosition()
{
- size_t bufferPosition;
- if (mMutator.ack()) {
- bufferPosition = (size_t) mCblk->u.mStatic.mBufferPosition;
- if (bufferPosition > mFrameCount) {
- bufferPosition = mFrameCount;
- }
- } else {
- bufferPosition = mBufferPosition;
+ getBufferPositionAndLoopCount(NULL, NULL);
+ return mPosLoop.mBufferPosition;
+}
+
+void StaticAudioTrackClientProxy::getBufferPositionAndLoopCount(
+ size_t *position, int *loopCount)
+{
+ if (mMutator.ack() == StaticAudioTrackSingleStateQueue::SSQ_DONE) {
+ if (mPosLoopObserver.poll(mPosLoop)) {
+ ; // a valid mPosLoop should be available if ackDone is true.
+ }
}
- return bufferPosition;
+ if (position != NULL) {
+ *position = mPosLoop.mBufferPosition;
+ }
+ if (loopCount != NULL) {
+ *loopCount = mPosLoop.mLoopCount;
+ }
}
// ---------------------------------------------------------------------------
@@ -560,7 +618,8 @@
ssize_t filled = rear - newFront;
// Rather than shutting down on a corrupt flush, just treat it as a full flush
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
- ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, filled %d=%#x",
+ ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, "
+ "filled %d=%#x",
mFlush, flush, front, rear, mask, newFront, filled, filled);
newFront = rear;
}
@@ -739,13 +798,12 @@
StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers,
size_t frameCount, size_t frameSize)
: AudioTrackServerProxy(cblk, buffers, frameCount, frameSize),
- mObserver(&cblk->u.mStatic.mSingleStateQueue), mPosition(0),
+ mObserver(&cblk->u.mStatic.mSingleStateQueue),
+ mPosLoopMutator(&cblk->u.mStatic.mPosLoopQueue),
mFramesReadySafe(frameCount), mFramesReady(frameCount),
mFramesReadyIsCalledByMultipleThreads(false)
{
- mState.mLoopStart = 0;
- mState.mLoopEnd = 0;
- mState.mLoopCount = 0;
+ memset(&mState, 0, sizeof(mState));
}
void StaticAudioTrackServerProxy::framesReadyIsCalledByMultipleThreads()
@@ -762,55 +820,97 @@
return mFramesReadySafe;
}
-ssize_t StaticAudioTrackServerProxy::pollPosition()
+status_t StaticAudioTrackServerProxy::updateStateWithLoop(
+ StaticAudioTrackState *localState, const StaticAudioTrackState &update) const
{
- size_t position = mPosition;
- StaticAudioTrackState state;
- if (mObserver.poll(state)) {
+ if (localState->mLoopSequence != update.mLoopSequence) {
bool valid = false;
- size_t loopStart = state.mLoopStart;
- size_t loopEnd = state.mLoopEnd;
- if (state.mLoopCount == 0) {
- if (loopStart > mFrameCount) {
- loopStart = mFrameCount;
- }
- // ignore loopEnd
- mPosition = position = loopStart;
- mFramesReady = mFrameCount - mPosition;
- mState.mLoopCount = 0;
+ const size_t loopStart = update.mLoopStart;
+ const size_t loopEnd = update.mLoopEnd;
+ size_t position = localState->mPosition;
+ if (update.mLoopCount == 0) {
valid = true;
- } else if (state.mLoopCount >= -1) {
+ } else if (update.mLoopCount >= -1) {
if (loopStart < loopEnd && loopEnd <= mFrameCount &&
loopEnd - loopStart >= MIN_LOOP) {
// If the current position is greater than the end of the loop
// we "wrap" to the loop start. This might cause an audible pop.
if (position >= loopEnd) {
- mPosition = position = loopStart;
+ position = loopStart;
}
- if (state.mLoopCount == -1) {
- mFramesReady = INT64_MAX;
- } else {
- // mFramesReady is 64 bits to handle the effective number of frames
- // that the static audio track contains, including loops.
- // TODO: Later consider fixing overflow, but does not seem needed now
- // as will not overflow if loopStart and loopEnd are Java "ints".
- mFramesReady = int64_t(state.mLoopCount) * (loopEnd - loopStart)
- + mFrameCount - mPosition;
- }
- mState = state;
valid = true;
}
}
- if (!valid || mPosition > mFrameCount) {
+ if (!valid || position > mFrameCount) {
+ return NO_INIT;
+ }
+ localState->mPosition = position;
+ localState->mLoopCount = update.mLoopCount;
+ localState->mLoopEnd = loopEnd;
+ localState->mLoopStart = loopStart;
+ localState->mLoopSequence = update.mLoopSequence;
+ }
+ return OK;
+}
+
+status_t StaticAudioTrackServerProxy::updateStateWithPosition(
+ StaticAudioTrackState *localState, const StaticAudioTrackState &update) const
+{
+ if (localState->mPositionSequence != update.mPositionSequence) {
+ if (update.mPosition > mFrameCount) {
+ return NO_INIT;
+ } else if (localState->mLoopCount != 0 && update.mPosition >= localState->mLoopEnd) {
+ localState->mLoopCount = 0; // disable loop count if position is beyond loop end.
+ }
+ localState->mPosition = update.mPosition;
+ localState->mPositionSequence = update.mPositionSequence;
+ }
+ return OK;
+}
+
+ssize_t StaticAudioTrackServerProxy::pollPosition()
+{
+ StaticAudioTrackState state;
+ if (mObserver.poll(state)) {
+ StaticAudioTrackState trystate = mState;
+ bool result;
+ const int32_t diffSeq = state.mLoopSequence - state.mPositionSequence;
+
+ if (diffSeq < 0) {
+ result = updateStateWithLoop(&trystate, state) == OK &&
+ updateStateWithPosition(&trystate, state) == OK;
+ } else {
+ result = updateStateWithPosition(&trystate, state) == OK &&
+ updateStateWithLoop(&trystate, state) == OK;
+ }
+ if (!result) {
+ mObserver.done();
+ // caution: no update occurs so server state will be inconsistent with client state.
ALOGE("%s client pushed an invalid state, shutting down", __func__);
mIsShutdown = true;
return (ssize_t) NO_INIT;
}
+ mState = trystate;
+ if (mState.mLoopCount == -1) {
+ mFramesReady = INT64_MAX;
+ } else if (mState.mLoopCount == 0) {
+ mFramesReady = mFrameCount - mState.mPosition;
+ } else if (mState.mLoopCount > 0) {
+ // TODO: Later consider fixing overflow, but does not seem needed now
+ // as will not overflow if loopStart and loopEnd are Java "ints".
+ mFramesReady = int64_t(mState.mLoopCount) * (mState.mLoopEnd - mState.mLoopStart)
+ + mFrameCount - mState.mPosition;
+ }
mFramesReadySafe = clampToSize(mFramesReady);
// This may overflow, but client is not supposed to rely on it
- mCblk->u.mStatic.mBufferPosition = (uint32_t) position;
+ StaticAudioTrackPosLoop posLoop;
+
+ posLoop.mLoopCount = (int32_t) mState.mLoopCount;
+ posLoop.mBufferPosition = (uint32_t) mState.mPosition;
+ mPosLoopMutator.push(posLoop);
+ mObserver.done(); // safe to read mStatic variables.
}
- return (ssize_t) position;
+ return (ssize_t) mState.mPosition;
}
status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush __unused)
@@ -849,7 +949,7 @@
}
// As mFramesReady is the total remaining frames in the static audio track,
// it is always larger or equal to avail.
- LOG_ALWAYS_FATAL_IF(mFramesReady < avail);
+ LOG_ALWAYS_FATAL_IF(mFramesReady < (int64_t) avail);
buffer->mNonContig = mFramesReady == INT64_MAX ? SIZE_MAX : clampToSize(mFramesReady - avail);
mUnreleased = avail;
return NO_ERROR;
@@ -858,7 +958,7 @@
void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer)
{
size_t stepCount = buffer->mFrameCount;
- LOG_ALWAYS_FATAL_IF(!(stepCount <= mFramesReady));
+ LOG_ALWAYS_FATAL_IF(!((int64_t) stepCount <= mFramesReady));
LOG_ALWAYS_FATAL_IF(!(stepCount <= mUnreleased));
if (stepCount == 0) {
// prevent accidental re-use of buffer
@@ -868,11 +968,12 @@
}
mUnreleased -= stepCount;
audio_track_cblk_t* cblk = mCblk;
- size_t position = mPosition;
+ size_t position = mState.mPosition;
size_t newPosition = position + stepCount;
int32_t setFlags = 0;
if (!(position <= newPosition && newPosition <= mFrameCount)) {
- ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position, mFrameCount);
+ ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position,
+ mFrameCount);
newPosition = mFrameCount;
} else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) {
newPosition = mState.mLoopStart;
@@ -885,7 +986,7 @@
if (newPosition == mFrameCount) {
setFlags |= CBLK_BUFFER_END;
}
- mPosition = newPosition;
+ mState.mPosition = newPosition;
if (mFramesReady != INT64_MAX) {
mFramesReady -= stepCount;
}
@@ -893,7 +994,10 @@
cblk->mServer += stepCount;
// This may overflow, but client is not supposed to rely on it
- cblk->u.mStatic.mBufferPosition = (uint32_t) newPosition;
+ StaticAudioTrackPosLoop posLoop;
+ posLoop.mBufferPosition = mState.mPosition;
+ posLoop.mLoopCount = mState.mLoopCount;
+ mPosLoopMutator.push(posLoop);
if (setFlags != 0) {
(void) android_atomic_or(setFlags, &cblk->mFlags);
// this would be a good place to wake a futex
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 8e3b633..38055f9 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -83,6 +83,8 @@
GET_AUDIO_HW_SYNC
};
+#define MAX_ITEMS_PER_LIST 1024
+
class BpAudioFlinger : public BpInterface<IAudioFlinger>
{
public:
@@ -1289,15 +1291,27 @@
} break;
case LIST_AUDIO_PORTS: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- unsigned int num_ports = data.readInt32();
+ unsigned int numPortsReq = data.readInt32();
+ if (numPortsReq > MAX_ITEMS_PER_LIST) {
+ numPortsReq = MAX_ITEMS_PER_LIST;
+ }
+ unsigned int numPorts = numPortsReq;
struct audio_port *ports =
- (struct audio_port *)calloc(num_ports,
+ (struct audio_port *)calloc(numPortsReq,
sizeof(struct audio_port));
- status_t status = listAudioPorts(&num_ports, ports);
+ if (ports == NULL) {
+ reply->writeInt32(NO_MEMORY);
+ reply->writeInt32(0);
+ return NO_ERROR;
+ }
+ status_t status = listAudioPorts(&numPorts, ports);
reply->writeInt32(status);
+ reply->writeInt32(numPorts);
if (status == NO_ERROR) {
- reply->writeInt32(num_ports);
- reply->write(&ports, num_ports * sizeof(struct audio_port));
+ if (numPortsReq > numPorts) {
+ numPortsReq = numPorts;
+ }
+ reply->write(ports, numPortsReq * sizeof(struct audio_port));
}
free(ports);
return NO_ERROR;
@@ -1336,15 +1350,27 @@
} break;
case LIST_AUDIO_PATCHES: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- unsigned int num_patches = data.readInt32();
+ unsigned int numPatchesReq = data.readInt32();
+ if (numPatchesReq > MAX_ITEMS_PER_LIST) {
+ numPatchesReq = MAX_ITEMS_PER_LIST;
+ }
+ unsigned int numPatches = numPatchesReq;
struct audio_patch *patches =
- (struct audio_patch *)calloc(num_patches,
+ (struct audio_patch *)calloc(numPatchesReq,
sizeof(struct audio_patch));
- status_t status = listAudioPatches(&num_patches, patches);
+ if (patches == NULL) {
+ reply->writeInt32(NO_MEMORY);
+ reply->writeInt32(0);
+ return NO_ERROR;
+ }
+ status_t status = listAudioPatches(&numPatches, patches);
reply->writeInt32(status);
+ reply->writeInt32(numPatches);
if (status == NO_ERROR) {
- reply->writeInt32(num_patches);
- reply->write(&patches, num_patches * sizeof(struct audio_patch));
+ if (numPatchesReq > numPatches) {
+ numPatchesReq = numPatches;
+ }
+ reply->write(patches, numPatchesReq * sizeof(struct audio_patch));
}
free(patches);
return NO_ERROR;
@@ -1369,4 +1395,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IAudioFlingerClient.cpp b/media/libmedia/IAudioFlingerClient.cpp
index 1c299f7..641e6c1 100644
--- a/media/libmedia/IAudioFlingerClient.cpp
+++ b/media/libmedia/IAudioFlingerClient.cpp
@@ -99,4 +99,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index cfb28a9..39374d8 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -86,13 +86,15 @@
virtual status_t setDeviceConnectionState(
audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address)
+ const char *device_address,
+ const char *device_name)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(static_cast <uint32_t>(device));
data.writeInt32(static_cast <uint32_t>(state));
data.writeCString(device_address);
+ data.writeCString(device_name);
remote()->transact(SET_DEVICE_CONNECTION_STATE, data, &reply);
return static_cast <status_t> (reply.readInt32());
}
@@ -728,9 +730,11 @@
audio_policy_dev_state_t state =
static_cast <audio_policy_dev_state_t>(data.readInt32());
const char *device_address = data.readCString();
+ const char *device_name = data.readCString();
reply->writeInt32(static_cast<uint32_t> (setDeviceConnectionState(device,
state,
- device_address)));
+ device_address,
+ device_name)));
return NO_ERROR;
} break;
@@ -1224,4 +1228,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IAudioPolicyServiceClient.cpp b/media/libmedia/IAudioPolicyServiceClient.cpp
index e802277..7c65878 100644
--- a/media/libmedia/IAudioPolicyServiceClient.cpp
+++ b/media/libmedia/IAudioPolicyServiceClient.cpp
@@ -80,4 +80,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IAudioRecord.cpp b/media/libmedia/IAudioRecord.cpp
index 8a4a383..9d80753 100644
--- a/media/libmedia/IAudioRecord.cpp
+++ b/media/libmedia/IAudioRecord.cpp
@@ -91,4 +91,4 @@
}
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index df209fd..651cb61 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -292,4 +292,4 @@
}
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IDrm.cpp b/media/libmedia/IDrm.cpp
index b08fa82..714a0b3 100644
--- a/media/libmedia/IDrm.cpp
+++ b/media/libmedia/IDrm.cpp
@@ -125,7 +125,8 @@
Vector<uint8_t> const &initData,
String8 const &mimeType, DrmPlugin::KeyType keyType,
KeyedVector<String8, String8> const &optionalParameters,
- Vector<uint8_t> &request, String8 &defaultUrl) {
+ Vector<uint8_t> &request, String8 &defaultUrl,
+ DrmPlugin::KeyRequestType *keyRequestType) {
Parcel data, reply;
data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
@@ -143,6 +144,7 @@
readVector(reply, request);
defaultUrl = reply.readString8();
+ *keyRequestType = static_cast<DrmPlugin::KeyRequestType>(reply.readInt32());
return reply.readInt32();
}
@@ -562,13 +564,15 @@
Vector<uint8_t> request;
String8 defaultUrl;
+ DrmPlugin::KeyRequestType keyRequestType;
- status_t result = getKeyRequest(sessionId, initData,
- mimeType, keyType,
- optionalParameters,
- request, defaultUrl);
+ status_t result = getKeyRequest(sessionId, initData, mimeType,
+ keyType, optionalParameters, request, defaultUrl,
+ &keyRequestType);
+
writeVector(reply, request);
reply->writeString8(defaultUrl);
+ reply->writeInt32(static_cast<int32_t>(keyRequestType));
reply->writeInt32(result);
return OK;
}
diff --git a/media/libmedia/IDrmClient.cpp b/media/libmedia/IDrmClient.cpp
index f50715e..490c6ed 100644
--- a/media/libmedia/IDrmClient.cpp
+++ b/media/libmedia/IDrmClient.cpp
@@ -78,4 +78,4 @@
}
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IEffect.cpp b/media/libmedia/IEffect.cpp
index c2fff78..eb4b098 100644
--- a/media/libmedia/IEffect.cpp
+++ b/media/libmedia/IEffect.cpp
@@ -201,4 +201,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IEffectClient.cpp b/media/libmedia/IEffectClient.cpp
index aef4371..1322e72 100644
--- a/media/libmedia/IEffectClient.cpp
+++ b/media/libmedia/IEffectClient.cpp
@@ -141,4 +141,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IMediaCodecList.cpp b/media/libmedia/IMediaCodecList.cpp
index bf7c5ca..80020db 100644
--- a/media/libmedia/IMediaCodecList.cpp
+++ b/media/libmedia/IMediaCodecList.cpp
@@ -160,4 +160,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IMediaDeathNotifier.cpp b/media/libmedia/IMediaDeathNotifier.cpp
index 38e9ca0..d4360ea 100644
--- a/media/libmedia/IMediaDeathNotifier.cpp
+++ b/media/libmedia/IMediaDeathNotifier.cpp
@@ -108,4 +108,4 @@
}
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IMediaHTTPConnection.cpp b/media/libmedia/IMediaHTTPConnection.cpp
index 7e26ee6..2ff7658 100644
--- a/media/libmedia/IMediaHTTPConnection.cpp
+++ b/media/libmedia/IMediaHTTPConnection.cpp
@@ -178,5 +178,4 @@
IMPLEMENT_META_INTERFACE(
MediaHTTPConnection, "android.media.IMediaHTTPConnection");
-} // namespace android
-
+} // namespace android
diff --git a/media/libmedia/IMediaHTTPService.cpp b/media/libmedia/IMediaHTTPService.cpp
index 1260582..f30d0f3 100644
--- a/media/libmedia/IMediaHTTPService.cpp
+++ b/media/libmedia/IMediaHTTPService.cpp
@@ -54,5 +54,4 @@
IMPLEMENT_META_INTERFACE(
MediaHTTPService, "android.media.IMediaHTTPService");
-} // namespace android
-
+} // namespace android
diff --git a/media/libmedia/IMediaLogService.cpp b/media/libmedia/IMediaLogService.cpp
index a4af7b7..230749e 100644
--- a/media/libmedia/IMediaLogService.cpp
+++ b/media/libmedia/IMediaLogService.cpp
@@ -91,4 +91,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IMediaMetadataRetriever.cpp b/media/libmedia/IMediaMetadataRetriever.cpp
index aa2665a..551cffe 100644
--- a/media/libmedia/IMediaMetadataRetriever.cpp
+++ b/media/libmedia/IMediaMetadataRetriever.cpp
@@ -297,4 +297,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IMediaPlayer.cpp b/media/libmedia/IMediaPlayer.cpp
index 7f3e5cc..ce3009a 100644
--- a/media/libmedia/IMediaPlayer.cpp
+++ b/media/libmedia/IMediaPlayer.cpp
@@ -39,6 +39,7 @@
START,
STOP,
IS_PLAYING,
+ SET_PLAYBACK_RATE,
PAUSE,
SEEK_TO,
GET_CURRENT_POSITION,
@@ -164,6 +165,15 @@
return reply.readInt32();
}
+ status_t setPlaybackRate(float rate)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
+ data.writeFloat(rate);
+ remote()->transact(SET_PLAYBACK_RATE, data, &reply);
+ return reply.readInt32();
+ }
+
status_t pause()
{
Parcel data, reply;
@@ -426,6 +436,11 @@
reply->writeInt32(ret);
return NO_ERROR;
} break;
+ case SET_PLAYBACK_RATE: {
+ CHECK_INTERFACE(IMediaPlayer, data, reply);
+ reply->writeInt32(setPlaybackRate(data.readFloat()));
+ return NO_ERROR;
+ } break;
case PAUSE: {
CHECK_INTERFACE(IMediaPlayer, data, reply);
reply->writeInt32(pause());
@@ -559,4 +574,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IMediaPlayerClient.cpp b/media/libmedia/IMediaPlayerClient.cpp
index a670c96..d608386 100644
--- a/media/libmedia/IMediaPlayerClient.cpp
+++ b/media/libmedia/IMediaPlayerClient.cpp
@@ -75,4 +75,4 @@
}
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index feea267..aa7b2e1 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -234,4 +234,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IMediaRecorder.cpp b/media/libmedia/IMediaRecorder.cpp
index a733b68..8ca256c 100644
--- a/media/libmedia/IMediaRecorder.cpp
+++ b/media/libmedia/IMediaRecorder.cpp
@@ -46,7 +46,6 @@
SET_OUTPUT_FORMAT,
SET_VIDEO_ENCODER,
SET_AUDIO_ENCODER,
- SET_OUTPUT_FILE_PATH,
SET_OUTPUT_FILE_FD,
SET_VIDEO_SIZE,
SET_VIDEO_FRAMERATE,
@@ -158,16 +157,6 @@
return reply.readInt32();
}
- status_t setOutputFile(const char* path)
- {
- ALOGV("setOutputFile(%s)", path);
- Parcel data, reply;
- data.writeInterfaceToken(IMediaRecorder::getInterfaceDescriptor());
- data.writeCString(path);
- remote()->transact(SET_OUTPUT_FILE_PATH, data, &reply);
- return reply.readInt32();
- }
-
status_t setOutputFile(int fd, int64_t offset, int64_t length) {
ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
Parcel data, reply;
@@ -300,7 +289,8 @@
// ----------------------------------------------------------------------
status_t BnMediaRecorder::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+ uint32_t code, const Parcel& data, Parcel* reply,
+ uint32_t flags)
{
switch (code) {
case RELEASE: {
@@ -390,13 +380,6 @@
return NO_ERROR;
} break;
- case SET_OUTPUT_FILE_PATH: {
- ALOGV("SET_OUTPUT_FILE_PATH");
- CHECK_INTERFACE(IMediaRecorder, data, reply);
- const char* path = data.readCString();
- reply->writeInt32(setOutputFile(path));
- return NO_ERROR;
- } break;
case SET_OUTPUT_FILE_FD: {
ALOGV("SET_OUTPUT_FILE_FD");
CHECK_INTERFACE(IMediaRecorder, data, reply);
@@ -445,7 +428,8 @@
case SET_PREVIEW_SURFACE: {
ALOGV("SET_PREVIEW_SURFACE");
CHECK_INTERFACE(IMediaRecorder, data, reply);
- sp<IGraphicBufferProducer> surface = interface_cast<IGraphicBufferProducer>(data.readStrongBinder());
+ sp<IGraphicBufferProducer> surface = interface_cast<IGraphicBufferProducer>(
+ data.readStrongBinder());
reply->writeInt32(setPreviewSurface(surface));
return NO_ERROR;
} break;
@@ -479,4 +463,4 @@
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IMediaRecorderClient.cpp b/media/libmedia/IMediaRecorderClient.cpp
index e7907e3..6795d23 100644
--- a/media/libmedia/IMediaRecorderClient.cpp
+++ b/media/libmedia/IMediaRecorderClient.cpp
@@ -67,4 +67,4 @@
}
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IRemoteDisplay.cpp b/media/libmedia/IRemoteDisplay.cpp
index 1e15434..869d11a 100644
--- a/media/libmedia/IRemoteDisplay.cpp
+++ b/media/libmedia/IRemoteDisplay.cpp
@@ -91,4 +91,4 @@
}
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IRemoteDisplayClient.cpp b/media/libmedia/IRemoteDisplayClient.cpp
index 9d63bc9..bedeb6c 100644
--- a/media/libmedia/IRemoteDisplayClient.cpp
+++ b/media/libmedia/IRemoteDisplayClient.cpp
@@ -101,4 +101,4 @@
}
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/IResourceManagerClient.cpp b/media/libmedia/IResourceManagerClient.cpp
new file mode 100644
index 0000000..6fa56fc
--- /dev/null
+++ b/media/libmedia/IResourceManagerClient.cpp
@@ -0,0 +1,70 @@
+/*
+**
+** Copyright 2015, 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 <utils/RefBase.h>
+#include <binder/IInterface.h>
+#include <binder/Parcel.h>
+
+#include <media/IResourceManagerClient.h>
+
+namespace android {
+
+enum {
+ RECLAIM_RESOURCE = IBinder::FIRST_CALL_TRANSACTION,
+};
+
+class BpResourceManagerClient: public BpInterface<IResourceManagerClient>
+{
+public:
+ BpResourceManagerClient(const sp<IBinder> &impl)
+ : BpInterface<IResourceManagerClient>(impl)
+ {
+ }
+
+ virtual bool reclaimResource() {
+ Parcel data, reply;
+ data.writeInterfaceToken(IResourceManagerClient::getInterfaceDescriptor());
+
+ bool ret = false;
+ status_t status = remote()->transact(RECLAIM_RESOURCE, data, &reply);
+ if (status == NO_ERROR) {
+ ret = (bool)reply.readInt32();
+ }
+ return ret;
+ }
+};
+
+IMPLEMENT_META_INTERFACE(ResourceManagerClient, "android.media.IResourceManagerClient");
+
+// ----------------------------------------------------------------------
+
+status_t BnResourceManagerClient::onTransact(
+ uint32_t code, const Parcel &data, Parcel *reply, uint32_t flags)
+{
+ switch (code) {
+ case RECLAIM_RESOURCE: {
+ CHECK_INTERFACE(IResourceManagerClient, data, reply);
+ bool ret = reclaimResource();
+ reply->writeInt32(ret);
+ return NO_ERROR;
+ } break;
+ default:
+ return BBinder::onTransact(code, data, reply, flags);
+ }
+}
+
+}; // namespace android
diff --git a/media/libmedia/IResourceManagerService.cpp b/media/libmedia/IResourceManagerService.cpp
new file mode 100644
index 0000000..95a2d1c
--- /dev/null
+++ b/media/libmedia/IResourceManagerService.cpp
@@ -0,0 +1,169 @@
+/*
+**
+** Copyright 2015, 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 "IResourceManagerService"
+#include <utils/Log.h>
+
+#include "media/IResourceManagerService.h"
+
+#include <binder/Parcel.h>
+
+#include <stdint.h>
+#include <sys/types.h>
+
+namespace android {
+
+enum {
+ CONFIG = IBinder::FIRST_CALL_TRANSACTION,
+ ADD_RESOURCE,
+ REMOVE_RESOURCE,
+ RECLAIM_RESOURCE,
+};
+
+template <typename T>
+static void writeToParcel(Parcel *data, const Vector<T> &items) {
+ size_t size = items.size();
+ size_t sizePosition = data->dataPosition();
+ // truncates size, but should be okay for this usecase
+ data->writeUint32(static_cast<uint32_t>(size));
+ for (size_t i = 0; i < size; i++) {
+ size_t position = data->dataPosition();
+ items[i].writeToParcel(data);
+ }
+}
+
+template <typename T>
+static void readFromParcel(const Parcel &data, Vector<T> *items) {
+ size_t size = (size_t)data.readUint32();
+ for (size_t i = 0; i < size; i++) {
+ T item;
+ item.readFromParcel(data);
+ items->add(item);
+ }
+}
+
+class BpResourceManagerService : public BpInterface<IResourceManagerService>
+{
+public:
+ BpResourceManagerService(const sp<IBinder> &impl)
+ : BpInterface<IResourceManagerService>(impl)
+ {
+ }
+
+ virtual void config(const Vector<MediaResourcePolicy> &policies) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IResourceManagerService::getInterfaceDescriptor());
+ writeToParcel(&data, policies);
+ remote()->transact(CONFIG, data, &reply);
+ }
+
+ virtual void addResource(
+ int pid,
+ int64_t clientId,
+ const sp<IResourceManagerClient> client,
+ const Vector<MediaResource> &resources) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IResourceManagerService::getInterfaceDescriptor());
+ data.writeInt32(pid);
+ data.writeInt64(clientId);
+ data.writeStrongBinder(IInterface::asBinder(client));
+ writeToParcel(&data, resources);
+
+ remote()->transact(ADD_RESOURCE, data, &reply);
+ }
+
+ virtual void removeResource(int64_t clientId) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IResourceManagerService::getInterfaceDescriptor());
+ data.writeInt64(clientId);
+
+ remote()->transact(REMOVE_RESOURCE, data, &reply);
+ }
+
+ virtual bool reclaimResource(int callingPid, const Vector<MediaResource> &resources) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IResourceManagerService::getInterfaceDescriptor());
+ data.writeInt32(callingPid);
+ writeToParcel(&data, resources);
+
+ bool ret = false;
+ status_t status = remote()->transact(RECLAIM_RESOURCE, data, &reply);
+ if (status == NO_ERROR) {
+ ret = (bool)reply.readInt32();
+ }
+ return ret;
+ }
+};
+
+IMPLEMENT_META_INTERFACE(ResourceManagerService, "android.media.IResourceManagerService");
+
+// ----------------------------------------------------------------------
+
+
+status_t BnResourceManagerService::onTransact(
+ uint32_t code, const Parcel &data, Parcel *reply, uint32_t flags)
+{
+ switch (code) {
+ case CONFIG: {
+ CHECK_INTERFACE(IResourceManagerService, data, reply);
+ int pid = data.readInt32();
+ sp<IResourceManagerClient> client(
+ interface_cast<IResourceManagerClient>(data.readStrongBinder()));
+ Vector<MediaResourcePolicy> policies;
+ readFromParcel(data, &policies);
+ config(policies);
+ return NO_ERROR;
+ } break;
+
+ case ADD_RESOURCE: {
+ CHECK_INTERFACE(IResourceManagerService, data, reply);
+ int pid = data.readInt32();
+ int64_t clientId = data.readInt64();
+ sp<IResourceManagerClient> client(
+ interface_cast<IResourceManagerClient>(data.readStrongBinder()));
+ Vector<MediaResource> resources;
+ readFromParcel(data, &resources);
+ addResource(pid, clientId, client, resources);
+ return NO_ERROR;
+ } break;
+
+ case REMOVE_RESOURCE: {
+ CHECK_INTERFACE(IResourceManagerService, data, reply);
+ int64_t clientId = data.readInt64();
+ removeResource(clientId);
+ return NO_ERROR;
+ } break;
+
+ case RECLAIM_RESOURCE: {
+ CHECK_INTERFACE(IResourceManagerService, data, reply);
+ int callingPid = data.readInt32();
+ Vector<MediaResource> resources;
+ readFromParcel(data, &resources);
+ bool ret = reclaimResource(callingPid, resources);
+ reply->writeInt32(ret);
+ return NO_ERROR;
+ } break;
+
+ default:
+ return BBinder::onTransact(code, data, reply, flags);
+ }
+}
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
diff --git a/media/libmedia/IStreamSource.cpp b/media/libmedia/IStreamSource.cpp
index d480aef..840e453 100644
--- a/media/libmedia/IStreamSource.cpp
+++ b/media/libmedia/IStreamSource.cpp
@@ -35,6 +35,9 @@
// static
const char *const IStreamListener::kKeyMediaTimeUs = "media-time-us";
+// static
+const char *const IStreamListener::kKeyRecentMediaTimeUs = "recent-media-time-us";
+
enum {
// IStreamSource
SET_LISTENER = IBinder::FIRST_CALL_TRANSACTION,
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index 721d8d7..271be0c 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -408,7 +408,8 @@
ALOGV("JetPlayer::queueSegment segmentNum=%d, libNum=%d, repeatCount=%d, transpose=%d",
segmentNum, libNum, repeatCount, transpose);
Mutex::Autolock lock(mMutex);
- return JET_QueueSegment(mEasData, segmentNum, libNum, repeatCount, transpose, muteFlags, userID);
+ return JET_QueueSegment(mEasData, segmentNum, libNum, repeatCount, transpose, muteFlags,
+ userID);
}
//-------------------------------------------------------------------------------------------------
@@ -449,7 +450,8 @@
void JetPlayer::dumpJetStatus(S_JET_STATUS* pJetStatus)
{
if (pJetStatus!=NULL)
- ALOGV(">> current JET player status: userID=%d segmentRepeatCount=%d numQueuedSegments=%d paused=%d",
+ ALOGV(">> current JET player status: userID=%d segmentRepeatCount=%d numQueuedSegments=%d "
+ "paused=%d",
pJetStatus->currentUserID, pJetStatus->segmentRepeatCount,
pJetStatus->numQueuedSegments, pJetStatus->paused);
else
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index e2e6042..47f9258 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -163,7 +163,8 @@
}
/*static*/ int
-MediaProfiles::findTagForName(const MediaProfiles::NameToTagMap *map, size_t nMappings, const char *name)
+MediaProfiles::findTagForName(const MediaProfiles::NameToTagMap *map, size_t nMappings,
+ const char *name)
{
int tag = -1;
for (size_t i = 0; i < nMappings; ++i) {
@@ -295,9 +296,8 @@
CHECK(codec != -1);
MediaProfiles::AudioEncoderCap *cap =
- new MediaProfiles::AudioEncoderCap(static_cast<audio_encoder>(codec), atoi(atts[5]), atoi(atts[7]),
- atoi(atts[9]), atoi(atts[11]), atoi(atts[13]),
- atoi(atts[15]));
+ new MediaProfiles::AudioEncoderCap(static_cast<audio_encoder>(codec), atoi(atts[5]),
+ atoi(atts[7]), atoi(atts[9]), atoi(atts[11]), atoi(atts[13]), atoi(atts[15]));
logAudioEncoderCap(*cap);
return cap;
}
@@ -330,7 +330,8 @@
!strcmp("fileFormat", atts[2]) &&
!strcmp("duration", atts[4]));
- const size_t nProfileMappings = sizeof(sCamcorderQualityNameMap)/sizeof(sCamcorderQualityNameMap[0]);
+ const size_t nProfileMappings = sizeof(sCamcorderQualityNameMap)/
+ sizeof(sCamcorderQualityNameMap[0]);
const int quality = findTagForName(sCamcorderQualityNameMap, nProfileMappings, atts[1]);
CHECK(quality != -1);
@@ -722,16 +723,20 @@
MediaProfiles::createDefaultCamcorderTimeLapseLowProfiles(
MediaProfiles::CamcorderProfile **lowTimeLapseProfile,
MediaProfiles::CamcorderProfile **lowSpecificTimeLapseProfile) {
- *lowTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(CAMCORDER_QUALITY_TIME_LAPSE_LOW);
- *lowSpecificTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(CAMCORDER_QUALITY_TIME_LAPSE_QCIF);
+ *lowTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(
+ CAMCORDER_QUALITY_TIME_LAPSE_LOW);
+ *lowSpecificTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(
+ CAMCORDER_QUALITY_TIME_LAPSE_QCIF);
}
/*static*/ void
MediaProfiles::createDefaultCamcorderTimeLapseHighProfiles(
MediaProfiles::CamcorderProfile **highTimeLapseProfile,
MediaProfiles::CamcorderProfile **highSpecificTimeLapseProfile) {
- *highTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(CAMCORDER_QUALITY_TIME_LAPSE_HIGH);
- *highSpecificTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(CAMCORDER_QUALITY_TIME_LAPSE_480P);
+ *highTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(
+ CAMCORDER_QUALITY_TIME_LAPSE_HIGH);
+ *highSpecificTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(
+ CAMCORDER_QUALITY_TIME_LAPSE_480P);
}
/*static*/ MediaProfiles::CamcorderProfile*
@@ -809,7 +814,8 @@
// high camcorder time lapse profiles.
MediaProfiles::CamcorderProfile *highTimeLapseProfile, *highSpecificTimeLapseProfile;
- createDefaultCamcorderTimeLapseHighProfiles(&highTimeLapseProfile, &highSpecificTimeLapseProfile);
+ createDefaultCamcorderTimeLapseHighProfiles(&highTimeLapseProfile,
+ &highSpecificTimeLapseProfile);
profiles->mCamcorderProfiles.add(highTimeLapseProfile);
profiles->mCamcorderProfiles.add(highSpecificTimeLapseProfile);
diff --git a/media/libmedia/MediaResource.cpp b/media/libmedia/MediaResource.cpp
new file mode 100644
index 0000000..8be01bc
--- /dev/null
+++ b/media/libmedia/MediaResource.cpp
@@ -0,0 +1,65 @@
+/*
+ * Copyright 2015 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 "MediaResource"
+#include <utils/Log.h>
+#include <media/MediaResource.h>
+
+namespace android {
+
+const char kResourceSecureCodec[] = "secure-codec";
+const char kResourceNonSecureCodec[] = "non-secure-codec";
+const char kResourceGraphicMemory[] = "graphic-memory";
+
+MediaResource::MediaResource() : mValue(0) {}
+
+MediaResource::MediaResource(String8 type, uint64_t value)
+ : mType(type),
+ mValue(value) {}
+
+MediaResource::MediaResource(String8 type, String8 subType, uint64_t value)
+ : mType(type),
+ mSubType(subType),
+ mValue(value) {}
+
+void MediaResource::readFromParcel(const Parcel &parcel) {
+ mType = parcel.readString8();
+ mSubType = parcel.readString8();
+ mValue = parcel.readUint64();
+}
+
+void MediaResource::writeToParcel(Parcel *parcel) const {
+ parcel->writeString8(mType);
+ parcel->writeString8(mSubType);
+ parcel->writeUint64(mValue);
+}
+
+String8 MediaResource::toString() const {
+ String8 str;
+ str.appendFormat("%s/%s:%llu", mType.string(), mSubType.string(), mValue);
+ return str;
+}
+
+bool MediaResource::operator==(const MediaResource &other) const {
+ return (other.mType == mType) && (other.mSubType == mSubType) && (other.mValue == mValue);
+}
+
+bool MediaResource::operator!=(const MediaResource &other) const {
+ return !(*this == other);
+}
+
+}; // namespace android
diff --git a/media/libmedia/MediaResourcePolicy.cpp b/media/libmedia/MediaResourcePolicy.cpp
new file mode 100644
index 0000000..2bb996a
--- /dev/null
+++ b/media/libmedia/MediaResourcePolicy.cpp
@@ -0,0 +1,49 @@
+/*
+ * Copyright 2015 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 "MediaResourcePolicy"
+#include <utils/Log.h>
+#include <media/MediaResourcePolicy.h>
+
+namespace android {
+
+const char kPolicySupportsMultipleSecureCodecs[] = "supports-multiple-secure-codecs";
+const char kPolicySupportsSecureWithNonSecureCodec[] = "supports-secure-with-non-secure-codec";
+
+MediaResourcePolicy::MediaResourcePolicy() : mValue(0) {}
+
+MediaResourcePolicy::MediaResourcePolicy(String8 type, uint64_t value)
+ : mType(type),
+ mValue(value) {}
+
+void MediaResourcePolicy::readFromParcel(const Parcel &parcel) {
+ mType = parcel.readString8();
+ mValue = parcel.readUint64();
+}
+
+void MediaResourcePolicy::writeToParcel(Parcel *parcel) const {
+ parcel->writeString8(mType);
+ parcel->writeUint64(mValue);
+}
+
+String8 MediaResourcePolicy::toString() const {
+ String8 str;
+ str.appendFormat("%s:%llu", mType.string(), mValue);
+ return str;
+}
+
+}; // namespace android
diff --git a/media/libmedia/SingleStateQueue.cpp b/media/libmedia/SingleStateQueue.cpp
deleted file mode 100644
index c241184..0000000
--- a/media/libmedia/SingleStateQueue.cpp
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- * Copyright (C) 2012 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 <new>
-#include <cutils/atomic.h>
-#include <media/SingleStateQueue.h>
-
-namespace android {
-
-template<typename T> SingleStateQueue<T>::Mutator::Mutator(Shared *shared)
- : mSequence(0), mShared((Shared *) shared)
-{
- // exactly one of Mutator and Observer must initialize, currently it is Observer
- //shared->init();
-}
-
-template<typename T> int32_t SingleStateQueue<T>::Mutator::push(const T& value)
-{
- Shared *shared = mShared;
- int32_t sequence = mSequence;
- sequence++;
- android_atomic_acquire_store(sequence, &shared->mSequence);
- shared->mValue = value;
- sequence++;
- android_atomic_release_store(sequence, &shared->mSequence);
- mSequence = sequence;
- // consider signalling a futex here, if we know that observer is waiting
- return sequence;
-}
-
-template<typename T> bool SingleStateQueue<T>::Mutator::ack()
-{
- return mShared->mAck - mSequence == 0;
-}
-
-template<typename T> bool SingleStateQueue<T>::Mutator::ack(int32_t sequence)
-{
- // this relies on 2's complement rollover to detect an ancient sequence number
- return mShared->mAck - sequence >= 0;
-}
-
-template<typename T> SingleStateQueue<T>::Observer::Observer(Shared *shared)
- : mSequence(0), mSeed(1), mShared((Shared *) shared)
-{
- // exactly one of Mutator and Observer must initialize, currently it is Observer
- shared->init();
-}
-
-template<typename T> bool SingleStateQueue<T>::Observer::poll(T& value)
-{
- Shared *shared = mShared;
- int32_t before = shared->mSequence;
- if (before == mSequence) {
- return false;
- }
- for (int tries = 0; ; ) {
- const int MAX_TRIES = 5;
- if (before & 1) {
- if (++tries >= MAX_TRIES) {
- return false;
- }
- before = shared->mSequence;
- } else {
- android_memory_barrier();
- T temp = shared->mValue;
- int32_t after = android_atomic_release_load(&shared->mSequence);
- if (after == before) {
- value = temp;
- shared->mAck = before;
- mSequence = before;
- return true;
- }
- if (++tries >= MAX_TRIES) {
- return false;
- }
- before = after;
- }
- }
-}
-
-#if 0
-template<typename T> SingleStateQueue<T>::SingleStateQueue(void /*Shared*/ *shared)
-{
- ((Shared *) shared)->init();
-}
-#endif
-
-} // namespace android
-
-// hack for gcc
-#ifdef SINGLE_STATE_QUEUE_INSTANTIATIONS
-#include SINGLE_STATE_QUEUE_INSTANTIATIONS
-#endif
diff --git a/media/libmedia/SingleStateQueueInstantiations.cpp b/media/libmedia/SingleStateQueueInstantiations.cpp
deleted file mode 100644
index 0265c8c..0000000
--- a/media/libmedia/SingleStateQueueInstantiations.cpp
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (C) 2012 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/SingleStateQueue.h>
-#include <private/media/StaticAudioTrackState.h>
-#include <media/AudioTimestamp.h>
-
-// FIXME hack for gcc
-
-namespace android {
-
-template class SingleStateQueue<StaticAudioTrackState>; // typedef StaticAudioTrackSingleStateQueue
-template class SingleStateQueue<AudioTimestamp>; // typedef AudioTimestampSingleStateQueue
-
-}
diff --git a/media/libmedia/StringArray.cpp b/media/libmedia/StringArray.cpp
index 5f5b57a..b2e5907 100644
--- a/media/libmedia/StringArray.cpp
+++ b/media/libmedia/StringArray.cpp
@@ -16,7 +16,7 @@
//
// Sortable array of strings. STL-ish, but STL-free.
-//
+//
#include <stdlib.h>
#include <string.h>
@@ -110,4 +110,4 @@
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
index f91e3e4..9d69b6a 100644
--- a/media/libmedia/Visualizer.cpp
+++ b/media/libmedia/Visualizer.cpp
@@ -429,4 +429,4 @@
return false;
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/docs/Makefile b/media/libmedia/docs/Makefile
new file mode 100644
index 0000000..bddbc9b
--- /dev/null
+++ b/media/libmedia/docs/Makefile
@@ -0,0 +1,2 @@
+paused.png : paused.dot
+ dot -Tpng < $< > $@
diff --git a/media/libmedia/docs/paused.dot b/media/libmedia/docs/paused.dot
new file mode 100644
index 0000000..11e1777
--- /dev/null
+++ b/media/libmedia/docs/paused.dot
@@ -0,0 +1,85 @@
+digraph paused {
+initial [label="INITIAL\n\
+mIgnoreNextPausedInt = false\n\
+mPaused = false\n\
+mPausedInt = false"];
+
+resume_body [label="mIgnoreNextPausedInt = true\nif (mPaused || mPausedInt)"];
+resume_paused [label="mPaused = false\nmPausedInt = false\nsignal()"];
+resume_paused -> resume_merged;
+resume_merged [label="return"];
+
+Application -> ATstop;
+ATstop [label="AudioTrack::stop()"];
+ATstop -> pause;
+Application -> ATpause;
+ATpause [label="AudioTrack::pause()"];
+ATpause -> pause;
+ATstart -> resume;
+ATstart [label="AudioTrack::start()"];
+destructor [label="~AudioTrack()"];
+destructor -> requestExit;
+requestExit [label="AudioTrackThread::requestExit()"];
+requestExit -> resume;
+Application -> ATsetMarkerPosition
+ATsetMarkerPosition [label="AudioTrack::setMarkerPosition()\n[sets marker variables]"];
+ATsetMarkerPosition -> ATTwake
+Application -> ATsetPositionUpdatePeriod
+ATsetPositionUpdatePeriod [label="AudioTrack::setPositionUpdatePeriod()\n[sets update period variables]"];
+ATsetPositionUpdatePeriod -> ATTwake
+Application -> ATstart;
+
+resume [label="AudioTrackThread::resume()"];
+resume -> resume_body;
+
+resume_body -> resume_paused [label="true"];
+resume_body -> resume_merged [label="false"];
+
+ATTwake [label="AudioTrackThread::wake()\nif (!mPaused && mPausedInt && mPausedNs > 0)"];
+ATTwake-> ATTWake_wakeable [label="true"];
+ATTWake_wakeable [label="mIgnoreNextPausedInt = true\nmPausedInt = false\nsignal()"];
+ATTwake-> ATTWake_cannotwake [label="false"]
+ATTWake_cannotwake [label="ignore"];
+
+pause [label="mPaused = true"];
+pause -> return;
+
+threadLoop [label="AudioTrackThread::threadLoop()\nENTRY"];
+threadLoop -> threadLoop_1;
+threadLoop_1 [label="if (mPaused)"];
+threadLoop_1 -> threadLoop_1_true [label="true"];
+threadLoop_1 -> threadLoop_2 [label="false"];
+threadLoop_1_true [label="wait()\nreturn true"];
+threadLoop_2 [label="if (mIgnoreNextPausedInt)"];
+threadLoop_2 -> threadLoop_2_true [label="true"];
+threadLoop_2 -> threadLoop_3 [label="false"];
+threadLoop_2_true [label="mIgnoreNextPausedInt = false\nmPausedInt = false"];
+threadLoop_2_true -> threadLoop_3;
+threadLoop_3 [label="if (mPausedInt)"];
+threadLoop_3 -> threadLoop_3_true [label="true"];
+threadLoop_3 -> threadLoop_4 [label="false"];
+threadLoop_3_true [label="wait()\nmPausedInt = false\nreturn true"];
+threadLoop_4 [label="if (exitPending)"];
+threadLoop_4 -> threadLoop_4_true [label="true"];
+threadLoop_4 -> threadLoop_5 [label="false"];
+threadLoop_4_true [label="return false"];
+threadLoop_5 [label="ns = processAudioBuffer()"];
+threadLoop_5 -> threadLoop_6;
+threadLoop_6 [label="case ns"];
+threadLoop_6 -> threadLoop_6_0 [label="0"];
+threadLoop_6 -> threadLoop_6_NS_INACTIVE [label="NS_INACTIVE"];
+threadLoop_6 -> threadLoop_6_NS_NEVER [label="NS_NEVER"];
+threadLoop_6 -> threadLoop_6_NS_WHENEVER [label="NS_WHENEVER"];
+threadLoop_6 -> threadLoop_6_default [label="default"];
+threadLoop_6_default [label="if (ns < 0)"];
+threadLoop_6_default -> threadLoop_6_default_true [label="true"];
+threadLoop_6_default -> threadLoop_6_default_false [label="false"];
+threadLoop_6_default_true [label="FATAL"];
+threadLoop_6_default_false [label="pauseInternal(ns) [wake()-able]\nmPausedInternal = true\nmPausedNs = ns\nreturn true"];
+threadLoop_6_0 [label="return true"];
+threadLoop_6_NS_INACTIVE [label="pauseInternal()\nmPausedInternal = true\nmPausedNs = 0\nreturn true"];
+threadLoop_6_NS_NEVER [label="return false"];
+threadLoop_6_NS_WHENEVER [label="ns = 1s"];
+threadLoop_6_NS_WHENEVER -> threadLoop_6_default_false;
+
+}
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 8e8a1ed..873808a 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -176,4 +176,4 @@
}
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 05c89ed..5dd8c02 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -59,6 +59,7 @@
mLoop = false;
mLeftVolume = mRightVolume = 1.0;
mVideoWidth = mVideoHeight = 0;
+ mPlaybackRate = 1.0;
mLockThreadId = 0;
mAudioSessionId = AudioSystem::newAudioUniqueId();
AudioSystem::acquireAudioSessionId(mAudioSessionId, -1);
@@ -240,7 +241,7 @@
// must call with lock held
status_t MediaPlayer::prepareAsync_l()
{
- if ( (mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_INITIALIZED | MEDIA_PLAYER_STOPPED) ) ) {
+ if ( (mPlayer != 0) && ( mCurrentState & (MEDIA_PLAYER_INITIALIZED | MEDIA_PLAYER_STOPPED) ) ) {
mPlayer->setAudioStreamType(mStreamType);
if (mAudioAttributesParcel != NULL) {
mPlayer->setParameter(KEY_PARAMETER_AUDIO_ATTRIBUTES, *mAudioAttributesParcel);
@@ -378,6 +379,24 @@
return false;
}
+status_t MediaPlayer::setPlaybackRate(float rate)
+{
+ ALOGV("setPlaybackRate: %f", rate);
+ if (rate <= 0.0) {
+ return BAD_VALUE;
+ }
+ Mutex::Autolock _l(mLock);
+ if (mPlayer != 0) {
+ if (mPlaybackRate == rate) {
+ return NO_ERROR;
+ }
+ mPlaybackRate = rate;
+ return mPlayer->setPlaybackRate(rate);
+ }
+ ALOGV("setPlaybackRate: no active player");
+ return INVALID_OPERATION;
+}
+
status_t MediaPlayer::getVideoWidth(int *w)
{
ALOGV("getVideoWidth");
@@ -414,7 +433,8 @@
status_t MediaPlayer::getDuration_l(int *msec)
{
ALOGV("getDuration_l");
- bool isValidState = (mCurrentState & (MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_STOPPED | MEDIA_PLAYER_PLAYBACK_COMPLETE));
+ bool isValidState = (mCurrentState & (MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_STARTED |
+ MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_STOPPED | MEDIA_PLAYER_PLAYBACK_COMPLETE));
if (mPlayer != 0 && isValidState) {
int durationMs;
status_t ret = mPlayer->getDuration(&durationMs);
@@ -443,7 +463,8 @@
status_t MediaPlayer::seekTo_l(int msec)
{
ALOGV("seekTo %d", msec);
- if ((mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_PLAYBACK_COMPLETE) ) ) {
+ if ((mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PREPARED |
+ MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_PLAYBACK_COMPLETE) ) ) {
if ( msec < 0 ) {
ALOGW("Attempt to seek to invalid position: %d", msec);
msec = 0;
@@ -477,7 +498,8 @@
return NO_ERROR;
}
}
- ALOGE("Attempt to perform seekTo in wrong state: mPlayer=%p, mCurrentState=%u", mPlayer.get(), mCurrentState);
+ ALOGE("Attempt to perform seekTo in wrong state: mPlayer=%p, mCurrentState=%u", mPlayer.get(),
+ mCurrentState);
return INVALID_OPERATION;
}
@@ -855,4 +877,4 @@
return mPlayer->setNextPlayer(next == NULL ? NULL : next->mPlayer);
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp
index 1952b86..a2d6e53 100644
--- a/media/libmedia/mediarecorder.cpp
+++ b/media/libmedia/mediarecorder.cpp
@@ -264,32 +264,6 @@
return ret;
}
-status_t MediaRecorder::setOutputFile(const char* path)
-{
- ALOGV("setOutputFile(%s)", path);
- if (mMediaRecorder == NULL) {
- ALOGE("media recorder is not initialized yet");
- return INVALID_OPERATION;
- }
- if (mIsOutputFileSet) {
- ALOGE("output file has already been set");
- return INVALID_OPERATION;
- }
- if (!(mCurrentState & MEDIA_RECORDER_DATASOURCE_CONFIGURED)) {
- ALOGE("setOutputFile called in an invalid state(%d)", mCurrentState);
- return INVALID_OPERATION;
- }
-
- status_t ret = mMediaRecorder->setOutputFile(path);
- if (OK != ret) {
- ALOGV("setOutputFile failed: %d", ret);
- mCurrentState = MEDIA_RECORDER_ERROR;
- return ret;
- }
- mIsOutputFileSet = true;
- return ret;
-}
-
status_t MediaRecorder::setOutputFile(int fd, int64_t offset, int64_t length)
{
ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
@@ -706,4 +680,4 @@
notify(MEDIA_RECORDER_EVENT_ERROR, MEDIA_ERROR_SERVER_DIED, 0);
}
-}; // namespace android
+} // namespace android
diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk
index 9d8fe62..4b31715 100644
--- a/media/libmediaplayerservice/Android.mk
+++ b/media/libmediaplayerservice/Android.mk
@@ -10,6 +10,7 @@
ActivityManager.cpp \
Crypto.cpp \
Drm.cpp \
+ DrmSessionManager.cpp \
HDCP.cpp \
MediaPlayerFactory.cpp \
MediaPlayerService.cpp \
diff --git a/media/libmediaplayerservice/Drm.cpp b/media/libmediaplayerservice/Drm.cpp
index 73f1a2a..49e01d1 100644
--- a/media/libmediaplayerservice/Drm.cpp
+++ b/media/libmediaplayerservice/Drm.cpp
@@ -23,6 +23,8 @@
#include "Drm.h"
+#include "DrmSessionClientInterface.h"
+#include "DrmSessionManager.h"
#include <media/drm/DrmAPI.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AString.h>
@@ -33,6 +35,10 @@
namespace android {
+static inline int getCallingPid() {
+ return IPCThreadState::self()->getCallingPid();
+}
+
static bool checkPermission(const char* permissionString) {
#ifndef HAVE_ANDROID_OS
return true;
@@ -57,14 +63,41 @@
return memcmp((void *)lhs.array(), (void *)rhs.array(), rhs.size()) < 0;
}
+struct DrmSessionClient : public DrmSessionClientInterface {
+ DrmSessionClient(Drm* drm) : mDrm(drm) {}
+
+ virtual bool reclaimSession(const Vector<uint8_t>& sessionId) {
+ sp<Drm> drm = mDrm.promote();
+ if (drm == NULL) {
+ return true;
+ }
+ status_t err = drm->closeSession(sessionId);
+ if (err != OK) {
+ return false;
+ }
+ drm->sendEvent(DrmPlugin::kDrmPluginEventSessionReclaimed, 0, &sessionId, NULL);
+ return true;
+ }
+
+protected:
+ virtual ~DrmSessionClient() {}
+
+private:
+ wp<Drm> mDrm;
+
+ DISALLOW_EVIL_CONSTRUCTORS(DrmSessionClient);
+};
+
Drm::Drm()
: mInitCheck(NO_INIT),
+ mDrmSessionClient(new DrmSessionClient(this)),
mListener(NULL),
mFactory(NULL),
mPlugin(NULL) {
}
Drm::~Drm() {
+ DrmSessionManager::Instance()->removeDrm(mDrmSessionClient);
delete mPlugin;
mPlugin = NULL;
closeFactory();
@@ -289,7 +322,18 @@
return -EINVAL;
}
- return mPlugin->openSession(sessionId);
+ status_t err = mPlugin->openSession(sessionId);
+ if (err == ERROR_DRM_RESOURCE_BUSY) {
+ bool retry = false;
+ retry = DrmSessionManager::Instance()->reclaimSession(getCallingPid());
+ if (retry) {
+ err = mPlugin->openSession(sessionId);
+ }
+ }
+ if (err == OK) {
+ DrmSessionManager::Instance()->addSession(getCallingPid(), mDrmSessionClient, sessionId);
+ }
+ return err;
}
status_t Drm::closeSession(Vector<uint8_t> const &sessionId) {
@@ -303,14 +347,19 @@
return -EINVAL;
}
- return mPlugin->closeSession(sessionId);
+ status_t err = mPlugin->closeSession(sessionId);
+ if (err == OK) {
+ DrmSessionManager::Instance()->removeSession(sessionId);
+ }
+ return err;
}
status_t Drm::getKeyRequest(Vector<uint8_t> const &sessionId,
Vector<uint8_t> const &initData,
String8 const &mimeType, DrmPlugin::KeyType keyType,
KeyedVector<String8, String8> const &optionalParameters,
- Vector<uint8_t> &request, String8 &defaultUrl) {
+ Vector<uint8_t> &request, String8 &defaultUrl,
+ DrmPlugin::KeyRequestType *keyRequestType) {
Mutex::Autolock autoLock(mLock);
if (mInitCheck != OK) {
@@ -321,8 +370,11 @@
return -EINVAL;
}
+ DrmSessionManager::Instance()->useSession(sessionId);
+
return mPlugin->getKeyRequest(sessionId, initData, mimeType, keyType,
- optionalParameters, request, defaultUrl);
+ optionalParameters, request, defaultUrl,
+ keyRequestType);
}
status_t Drm::provideKeyResponse(Vector<uint8_t> const &sessionId,
@@ -338,6 +390,8 @@
return -EINVAL;
}
+ DrmSessionManager::Instance()->useSession(sessionId);
+
return mPlugin->provideKeyResponse(sessionId, response, keySetId);
}
@@ -367,6 +421,8 @@
return -EINVAL;
}
+ DrmSessionManager::Instance()->useSession(sessionId);
+
return mPlugin->restoreKeys(sessionId, keySetId);
}
@@ -382,6 +438,8 @@
return -EINVAL;
}
+ DrmSessionManager::Instance()->useSession(sessionId);
+
return mPlugin->queryKeyStatus(sessionId, infoMap);
}
@@ -561,6 +619,8 @@
return -EINVAL;
}
+ DrmSessionManager::Instance()->useSession(sessionId);
+
return mPlugin->setCipherAlgorithm(sessionId, algorithm);
}
@@ -576,6 +636,8 @@
return -EINVAL;
}
+ DrmSessionManager::Instance()->useSession(sessionId);
+
return mPlugin->setMacAlgorithm(sessionId, algorithm);
}
@@ -594,6 +656,8 @@
return -EINVAL;
}
+ DrmSessionManager::Instance()->useSession(sessionId);
+
return mPlugin->encrypt(sessionId, keyId, input, iv, output);
}
@@ -612,6 +676,8 @@
return -EINVAL;
}
+ DrmSessionManager::Instance()->useSession(sessionId);
+
return mPlugin->decrypt(sessionId, keyId, input, iv, output);
}
@@ -629,6 +695,8 @@
return -EINVAL;
}
+ DrmSessionManager::Instance()->useSession(sessionId);
+
return mPlugin->sign(sessionId, keyId, message, signature);
}
@@ -647,6 +715,8 @@
return -EINVAL;
}
+ DrmSessionManager::Instance()->useSession(sessionId);
+
return mPlugin->verify(sessionId, keyId, message, signature, match);
}
@@ -669,6 +739,8 @@
return -EPERM;
}
+ DrmSessionManager::Instance()->useSession(sessionId);
+
return mPlugin->signRSA(sessionId, algorithm, message, wrappedKey, signature);
}
diff --git a/media/libmediaplayerservice/Drm.h b/media/libmediaplayerservice/Drm.h
index 0e1eb2c..7e8f246 100644
--- a/media/libmediaplayerservice/Drm.h
+++ b/media/libmediaplayerservice/Drm.h
@@ -28,6 +28,7 @@
struct DrmFactory;
struct DrmPlugin;
+struct DrmSessionClientInterface;
struct Drm : public BnDrm,
public IBinder::DeathRecipient,
@@ -52,7 +53,8 @@
Vector<uint8_t> const &initData,
String8 const &mimeType, DrmPlugin::KeyType keyType,
KeyedVector<String8, String8> const &optionalParameters,
- Vector<uint8_t> &request, String8 &defaultUrl);
+ Vector<uint8_t> &request, String8 &defaultUrl,
+ DrmPlugin::KeyRequestType *keyRequestType);
virtual status_t provideKeyResponse(Vector<uint8_t> const &sessionId,
Vector<uint8_t> const &response,
@@ -138,6 +140,8 @@
status_t mInitCheck;
+ sp<DrmSessionClientInterface> mDrmSessionClient;
+
sp<IDrmClient> mListener;
mutable Mutex mEventLock;
mutable Mutex mNotifyLock;
diff --git a/media/libmediaplayerservice/DrmSessionClientInterface.h b/media/libmediaplayerservice/DrmSessionClientInterface.h
new file mode 100644
index 0000000..17faf08
--- /dev/null
+++ b/media/libmediaplayerservice/DrmSessionClientInterface.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (C) 2015 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 DRM_PROXY_INTERFACE_H_
+#define DRM_PROXY_INTERFACE_H_
+
+#include <utils/RefBase.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+struct DrmSessionClientInterface : public RefBase {
+ virtual bool reclaimSession(const Vector<uint8_t>& sessionId) = 0;
+
+protected:
+ virtual ~DrmSessionClientInterface() {}
+};
+
+} // namespace android
+
+#endif // DRM_PROXY_INTERFACE_H_
diff --git a/media/libmediaplayerservice/DrmSessionManager.cpp b/media/libmediaplayerservice/DrmSessionManager.cpp
new file mode 100644
index 0000000..641f881
--- /dev/null
+++ b/media/libmediaplayerservice/DrmSessionManager.cpp
@@ -0,0 +1,240 @@
+/*
+ * Copyright (C) 2015 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 "DrmSessionManager"
+#include <utils/Log.h>
+
+#include "DrmSessionManager.h"
+
+#include "DrmSessionClientInterface.h"
+#include <binder/IPCThreadState.h>
+#include <binder/IProcessInfoService.h>
+#include <binder/IServiceManager.h>
+#include <media/stagefright/ProcessInfo.h>
+#include <unistd.h>
+#include <utils/String8.h>
+
+namespace android {
+
+static String8 GetSessionIdString(const Vector<uint8_t> &sessionId) {
+ String8 sessionIdStr;
+ for (size_t i = 0; i < sessionId.size(); ++i) {
+ sessionIdStr.appendFormat("%u ", sessionId[i]);
+ }
+ return sessionIdStr;
+}
+
+bool isEqualSessionId(const Vector<uint8_t> &sessionId1, const Vector<uint8_t> &sessionId2) {
+ if (sessionId1.size() != sessionId2.size()) {
+ return false;
+ }
+ for (size_t i = 0; i < sessionId1.size(); ++i) {
+ if (sessionId1[i] != sessionId2[i]) {
+ return false;
+ }
+ }
+ return true;
+}
+
+sp<DrmSessionManager> DrmSessionManager::Instance() {
+ static sp<DrmSessionManager> drmSessionManager = new DrmSessionManager();
+ return drmSessionManager;
+}
+
+DrmSessionManager::DrmSessionManager()
+ : mProcessInfo(new ProcessInfo()),
+ mTime(0) {}
+
+DrmSessionManager::DrmSessionManager(sp<ProcessInfoInterface> processInfo)
+ : mProcessInfo(processInfo),
+ mTime(0) {}
+
+DrmSessionManager::~DrmSessionManager() {}
+
+void DrmSessionManager::addSession(
+ int pid, sp<DrmSessionClientInterface> drm, const Vector<uint8_t> &sessionId) {
+ ALOGV("addSession(pid %d, drm %p, sessionId %s)", pid, drm.get(),
+ GetSessionIdString(sessionId).string());
+
+ Mutex::Autolock lock(mLock);
+ SessionInfo info;
+ info.drm = drm;
+ info.sessionId = sessionId;
+ info.timeStamp = getTime_l();
+ ssize_t index = mSessionMap.indexOfKey(pid);
+ if (index < 0) {
+ // new pid
+ SessionInfos infosForPid;
+ infosForPid.push_back(info);
+ mSessionMap.add(pid, infosForPid);
+ } else {
+ mSessionMap.editValueAt(index).push_back(info);
+ }
+}
+
+void DrmSessionManager::useSession(const Vector<uint8_t> &sessionId) {
+ ALOGV("useSession(%s)", GetSessionIdString(sessionId).string());
+
+ Mutex::Autolock lock(mLock);
+ for (size_t i = 0; i < mSessionMap.size(); ++i) {
+ SessionInfos& infos = mSessionMap.editValueAt(i);
+ for (size_t j = 0; j < infos.size(); ++j) {
+ SessionInfo& info = infos.editItemAt(j);
+ if (isEqualSessionId(sessionId, info.sessionId)) {
+ info.timeStamp = getTime_l();
+ return;
+ }
+ }
+ }
+}
+
+void DrmSessionManager::removeSession(const Vector<uint8_t> &sessionId) {
+ ALOGV("removeSession(%s)", GetSessionIdString(sessionId).string());
+
+ Mutex::Autolock lock(mLock);
+ for (size_t i = 0; i < mSessionMap.size(); ++i) {
+ SessionInfos& infos = mSessionMap.editValueAt(i);
+ for (size_t j = 0; j < infos.size(); ++j) {
+ if (isEqualSessionId(sessionId, infos[j].sessionId)) {
+ infos.removeAt(j);
+ return;
+ }
+ }
+ }
+}
+
+void DrmSessionManager::removeDrm(sp<DrmSessionClientInterface> drm) {
+ ALOGV("removeDrm(%p)", drm.get());
+
+ Mutex::Autolock lock(mLock);
+ bool found = false;
+ for (size_t i = 0; i < mSessionMap.size(); ++i) {
+ SessionInfos& infos = mSessionMap.editValueAt(i);
+ for (size_t j = 0; j < infos.size();) {
+ if (infos[j].drm == drm) {
+ ALOGV("removed session (%s)", GetSessionIdString(infos[j].sessionId).string());
+ j = infos.removeAt(j);
+ found = true;
+ } else {
+ ++j;
+ }
+ }
+ if (found) {
+ break;
+ }
+ }
+}
+
+bool DrmSessionManager::reclaimSession(int callingPid) {
+ ALOGV("reclaimSession(%d)", callingPid);
+
+ sp<DrmSessionClientInterface> drm;
+ Vector<uint8_t> sessionId;
+ int lowestPriorityPid;
+ int lowestPriority;
+ {
+ Mutex::Autolock lock(mLock);
+ int callingPriority;
+ if (!mProcessInfo->getPriority(callingPid, &callingPriority)) {
+ return false;
+ }
+ if (!getLowestPriority_l(&lowestPriorityPid, &lowestPriority)) {
+ return false;
+ }
+ if (lowestPriority <= callingPriority) {
+ return false;
+ }
+
+ if (!getLeastUsedSession_l(lowestPriorityPid, &drm, &sessionId)) {
+ return false;
+ }
+ }
+
+ if (drm == NULL) {
+ return false;
+ }
+
+ ALOGV("reclaim session(%s) opened by pid %d",
+ GetSessionIdString(sessionId).string(), lowestPriorityPid);
+
+ return drm->reclaimSession(sessionId);
+}
+
+int64_t DrmSessionManager::getTime_l() {
+ return mTime++;
+}
+
+bool DrmSessionManager::getLowestPriority_l(int* lowestPriorityPid, int* lowestPriority) {
+ int pid = -1;
+ int priority = -1;
+ for (size_t i = 0; i < mSessionMap.size(); ++i) {
+ if (mSessionMap.valueAt(i).size() == 0) {
+ // no opened session by this process.
+ continue;
+ }
+ int tempPid = mSessionMap.keyAt(i);
+ int tempPriority;
+ if (!mProcessInfo->getPriority(tempPid, &tempPriority)) {
+ // shouldn't happen.
+ return false;
+ }
+ if (pid == -1) {
+ pid = tempPid;
+ priority = tempPriority;
+ } else {
+ if (tempPriority > priority) {
+ pid = tempPid;
+ priority = tempPriority;
+ }
+ }
+ }
+ if (pid != -1) {
+ *lowestPriorityPid = pid;
+ *lowestPriority = priority;
+ }
+ return (pid != -1);
+}
+
+bool DrmSessionManager::getLeastUsedSession_l(
+ int pid, sp<DrmSessionClientInterface>* drm, Vector<uint8_t>* sessionId) {
+ ssize_t index = mSessionMap.indexOfKey(pid);
+ if (index < 0) {
+ return false;
+ }
+
+ int leastUsedIndex = -1;
+ int64_t minTs = LLONG_MAX;
+ const SessionInfos& infos = mSessionMap.valueAt(index);
+ for (size_t j = 0; j < infos.size(); ++j) {
+ if (leastUsedIndex == -1) {
+ leastUsedIndex = j;
+ minTs = infos[j].timeStamp;
+ } else {
+ if (infos[j].timeStamp < minTs) {
+ leastUsedIndex = j;
+ minTs = infos[j].timeStamp;
+ }
+ }
+ }
+ if (leastUsedIndex != -1) {
+ *drm = infos[leastUsedIndex].drm;
+ *sessionId = infos[leastUsedIndex].sessionId;
+ }
+ return (leastUsedIndex != -1);
+}
+
+} // namespace android
diff --git a/media/libmediaplayerservice/DrmSessionManager.h b/media/libmediaplayerservice/DrmSessionManager.h
new file mode 100644
index 0000000..ba5c268
--- /dev/null
+++ b/media/libmediaplayerservice/DrmSessionManager.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2015 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 DRM_SESSION_MANAGER_H_
+
+#define DRM_SESSION_MANAGER_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/RefBase.h>
+#include <utils/KeyedVector.h>
+#include <utils/threads.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+class DrmSessionManagerTest;
+struct DrmSessionClientInterface;
+struct ProcessInfoInterface;
+
+bool isEqualSessionId(const Vector<uint8_t> &sessionId1, const Vector<uint8_t> &sessionId2);
+
+struct SessionInfo {
+ sp<DrmSessionClientInterface> drm;
+ Vector<uint8_t> sessionId;
+ int64_t timeStamp;
+};
+
+typedef Vector<SessionInfo > SessionInfos;
+typedef KeyedVector<int, SessionInfos > PidSessionInfosMap;
+
+struct DrmSessionManager : public RefBase {
+ static sp<DrmSessionManager> Instance();
+
+ DrmSessionManager();
+ DrmSessionManager(sp<ProcessInfoInterface> processInfo);
+
+ void addSession(int pid, sp<DrmSessionClientInterface> drm, const Vector<uint8_t>& sessionId);
+ void useSession(const Vector<uint8_t>& sessionId);
+ void removeSession(const Vector<uint8_t>& sessionId);
+ void removeDrm(sp<DrmSessionClientInterface> drm);
+ bool reclaimSession(int callingPid);
+
+protected:
+ virtual ~DrmSessionManager();
+
+private:
+ friend class DrmSessionManagerTest;
+
+ int64_t getTime_l();
+ bool getLowestPriority_l(int* lowestPriorityPid, int* lowestPriority);
+ bool getLeastUsedSession_l(
+ int pid, sp<DrmSessionClientInterface>* drm, Vector<uint8_t>* sessionId);
+
+ sp<ProcessInfoInterface> mProcessInfo;
+ mutable Mutex mLock;
+ PidSessionInfosMap mSessionMap;
+ int64_t mTime;
+
+ DISALLOW_EVIL_CONSTRUCTORS(DrmSessionManager);
+};
+
+} // namespace android
+
+#endif // DRM_SESSION_MANAGER_H_
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 694f1a4..f113e21 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -290,8 +290,9 @@
const sp<IServiceManager> sm(defaultServiceManager());
if (sm != NULL) {
const String16 name("batterystats");
+ // use checkService() to avoid blocking if service is not up yet
sp<IBatteryStats> batteryStats =
- interface_cast<IBatteryStats>(sm->getService(name));
+ interface_cast<IBatteryStats>(sm->checkService(name));
if (batteryStats != NULL) {
batteryStats->noteResetVideo();
batteryStats->noteResetAudio();
@@ -441,6 +442,9 @@
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
+ SortedVector< sp<Client> > clients; //to serialise the mutex unlock & client destruction.
+ SortedVector< sp<MediaRecorderClient> > mediaRecorderClients;
+
if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
snprintf(buffer, SIZE, "Permission Denial: "
"can't dump MediaPlayerService from pid=%d, uid=%d\n",
@@ -452,6 +456,7 @@
for (int i = 0, n = mClients.size(); i < n; ++i) {
sp<Client> c = mClients[i].promote();
if (c != 0) c->dump(fd, args);
+ clients.add(c);
}
if (mMediaRecorderClients.size() == 0) {
result.append(" No media recorder client\n\n");
@@ -464,6 +469,7 @@
write(fd, result.string(), result.size());
result = "\n";
c->dump(fd, args);
+ mediaRecorderClients.add(c);
}
}
}
@@ -961,6 +967,14 @@
return NO_ERROR;
}
+status_t MediaPlayerService::Client::setPlaybackRate(float rate)
+{
+ ALOGV("[%d] setPlaybackRate(%f)", mConnId, rate);
+ sp<MediaPlayerBase> p = getPlayer();
+ if (p == 0) return UNKNOWN_ERROR;
+ return p->setPlaybackRate(rate);
+}
+
status_t MediaPlayerService::Client::getCurrentPosition(int *msec)
{
ALOGV("getCurrentPosition");
@@ -1664,13 +1678,13 @@
}
}
-ssize_t MediaPlayerService::AudioOutput::write(const void* buffer, size_t size)
+ssize_t MediaPlayerService::AudioOutput::write(const void* buffer, size_t size, bool blocking)
{
LOG_ALWAYS_FATAL_IF(mCallback != NULL, "Don't call write if supplying a callback.");
//ALOGV("write(%p, %u)", buffer, size);
if (mTrack != 0) {
- ssize_t ret = mTrack->write(buffer, size);
+ ssize_t ret = mTrack->write(buffer, size, blocking);
if (ret >= 0) {
mBytesWritten += ret;
}
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index fad3447..4ce4b81 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -97,7 +97,7 @@
const audio_offload_info_t *offloadInfo = NULL);
virtual status_t start();
- virtual ssize_t write(const void* buffer, size_t size);
+ virtual ssize_t write(const void* buffer, size_t size, bool blocking = true);
virtual void stop();
virtual void flush();
virtual void pause();
@@ -261,6 +261,7 @@
virtual status_t stop();
virtual status_t pause();
virtual status_t isPlaying(bool* state);
+ virtual status_t setPlaybackRate(float rate);
virtual status_t seekTo(int msec);
virtual status_t getCurrentPosition(int* msec);
virtual status_t getDuration(int* msec);
diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp
index 194abbb..4d4de9b 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.cpp
+++ b/media/libmediaplayerservice/MediaRecorderClient.cpp
@@ -154,17 +154,6 @@
return mRecorder->setAudioEncoder((audio_encoder)ae);
}
-status_t MediaRecorderClient::setOutputFile(const char* path)
-{
- ALOGV("setOutputFile(%s)", path);
- Mutex::Autolock lock(mLock);
- if (mRecorder == NULL) {
- ALOGE("recorder is not initialized");
- return NO_INIT;
- }
- return mRecorder->setOutputFile(path);
-}
-
status_t MediaRecorderClient::setOutputFile(int fd, int64_t offset, int64_t length)
{
ALOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
diff --git a/media/libmediaplayerservice/MediaRecorderClient.h b/media/libmediaplayerservice/MediaRecorderClient.h
index a65ec9f..a444b6c 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.h
+++ b/media/libmediaplayerservice/MediaRecorderClient.h
@@ -38,7 +38,6 @@
virtual status_t setOutputFormat(int of);
virtual status_t setVideoEncoder(int ve);
virtual status_t setAudioEncoder(int ae);
- virtual status_t setOutputFile(const char* path);
virtual status_t setOutputFile(int fd, int64_t offset,
int64_t length);
virtual status_t setVideoSize(int width, int height);
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 86639cb..55763f0 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -75,6 +75,7 @@
mAudioSource(AUDIO_SOURCE_CNT),
mVideoSource(VIDEO_SOURCE_LIST_END),
mCaptureTimeLapse(false),
+ mCaptureFps(0.0f),
mStarted(false) {
ALOGV("Constructor");
@@ -206,7 +207,7 @@
status_t StagefrightRecorder::setVideoFrameRate(int frames_per_second) {
ALOGV("setVideoFrameRate: %d", frames_per_second);
if ((frames_per_second <= 0 && frames_per_second != -1) ||
- frames_per_second > 120) {
+ frames_per_second > kMaxHighSpeedFps) {
ALOGE("Invalid video frame rate: %d", frames_per_second);
return BAD_VALUE;
}
@@ -241,14 +242,6 @@
return OK;
}
-status_t StagefrightRecorder::setOutputFile(const char * /* path */) {
- ALOGE("setOutputFile(const char*) must not be called");
- // We don't actually support this at all, as the media_server process
- // no longer has permissions to create files.
-
- return -EPERM;
-}
-
status_t StagefrightRecorder::setOutputFile(int fd, int64_t offset, int64_t length) {
ALOGV("setOutputFile: %d, %lld, %lld", fd, offset, length);
// These don't make any sense, do they?
@@ -271,6 +264,31 @@
return OK;
}
+// Attempt to parse an float literal optionally surrounded by whitespace,
+// returns true on success, false otherwise.
+static bool safe_strtof(const char *s, float *val) {
+ char *end;
+
+ // It is lame, but according to man page, we have to set errno to 0
+ // before calling strtof().
+ errno = 0;
+ *val = strtof(s, &end);
+
+ if (end == s || errno == ERANGE) {
+ return false;
+ }
+
+ // Skip trailing whitespace
+ while (isspace(*end)) {
+ ++end;
+ }
+
+ // For a successful return, the string must contain nothing but a valid
+ // float literal optionally surrounded by whitespace.
+
+ return *end == '\0';
+}
+
// Attempt to parse an int64 literal optionally surrounded by whitespace,
// returns true on success, false otherwise.
static bool safe_strtoi64(const char *s, int64_t *val) {
@@ -554,8 +572,10 @@
return OK;
}
-status_t StagefrightRecorder::setParamTimeBetweenTimeLapseFrameCapture(int64_t timeUs) {
- ALOGV("setParamTimeBetweenTimeLapseFrameCapture: %lld us", timeUs);
+status_t StagefrightRecorder::setParamTimeLapseFps(float fps) {
+ ALOGV("setParamTimeLapseFps: %.2f", fps);
+
+ int64_t timeUs = (int64_t) (1000000.0 / fps + 0.5f);
// Not allowing time more than a day
if (timeUs <= 0 || timeUs > 86400*1E6) {
@@ -563,6 +583,7 @@
return BAD_VALUE;
}
+ mCaptureFps = fps;
mTimeBetweenTimeLapseFrameCaptureUs = timeUs;
return OK;
}
@@ -690,11 +711,10 @@
if (safe_strtoi32(value.string(), &timeLapseEnable)) {
return setParamTimeLapseEnable(timeLapseEnable);
}
- } else if (key == "time-between-time-lapse-frame-capture") {
- int64_t timeBetweenTimeLapseFrameCaptureUs;
- if (safe_strtoi64(value.string(), &timeBetweenTimeLapseFrameCaptureUs)) {
- return setParamTimeBetweenTimeLapseFrameCapture(
- timeBetweenTimeLapseFrameCaptureUs);
+ } else if (key == "time-lapse-fps") {
+ float fps;
+ if (safe_strtof(value.string(), &fps)) {
+ return setParamTimeLapseFps(fps);
}
} else {
ALOGE("setParameter: failed to find key %s", key.string());
@@ -1589,10 +1609,11 @@
status_t err = OK;
sp<MediaWriter> writer;
+ sp<MPEG4Writer> mp4writer;
if (mOutputFormat == OUTPUT_FORMAT_WEBM) {
writer = new WebmWriter(mOutputFd);
} else {
- writer = new MPEG4Writer(mOutputFd);
+ writer = mp4writer = new MPEG4Writer(mOutputFd);
}
if (mVideoSource < VIDEO_SOURCE_LIST_END) {
@@ -1625,13 +1646,15 @@
mTotalBitRate += mAudioBitRate;
}
+ if (mCaptureTimeLapse) {
+ mp4writer->setCaptureRate(mCaptureFps);
+ }
+
if (mInterleaveDurationUs > 0) {
- reinterpret_cast<MPEG4Writer *>(writer.get())->
- setInterleaveDuration(mInterleaveDurationUs);
+ mp4writer->setInterleaveDuration(mInterleaveDurationUs);
}
if (mLongitudex10000 > -3600000 && mLatitudex10000 > -3600000) {
- reinterpret_cast<MPEG4Writer *>(writer.get())->
- setGeoData(mLatitudex10000, mLongitudex10000);
+ mp4writer->setGeoData(mLatitudex10000, mLongitudex10000);
}
}
if (mMaxFileDurationUs != 0) {
diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index 54c38d3..f34c229 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -53,7 +53,6 @@
virtual status_t setVideoFrameRate(int frames_per_second);
virtual status_t setCamera(const sp<ICamera>& camera, const sp<ICameraRecordingProxy>& proxy);
virtual status_t setPreviewSurface(const sp<IGraphicBufferProducer>& surface);
- virtual status_t setOutputFile(const char *path);
virtual status_t setOutputFile(int fd, int64_t offset, int64_t length);
virtual status_t setParameters(const String8& params);
virtual status_t setListener(const sp<IMediaRecorderClient>& listener);
@@ -110,6 +109,7 @@
int32_t mTotalBitRate;
bool mCaptureTimeLapse;
+ float mCaptureFps;
int64_t mTimeBetweenTimeLapseFrameCaptureUs;
sp<CameraSourceTimeLapse> mCameraSourceTimeLapse;
@@ -127,6 +127,8 @@
sp<IGraphicBufferProducer> mGraphicBufferProducer;
sp<ALooper> mLooper;
+ static const int kMaxHighSpeedFps = 1000;
+
status_t prepareInternal();
status_t setupMPEG4orWEBMRecording();
void setupMPEG4orWEBMMetaData(sp<MetaData> *meta);
@@ -154,7 +156,7 @@
status_t setParamAudioSamplingRate(int32_t sampleRate);
status_t setParamAudioTimeScale(int32_t timeScale);
status_t setParamTimeLapseEnable(int32_t timeLapseEnable);
- status_t setParamTimeBetweenTimeLapseFrameCapture(int64_t timeUs);
+ status_t setParamTimeLapseFps(float fps);
status_t setParamVideoEncodingBitRate(int32_t bitRate);
status_t setParamVideoIFramesInterval(int32_t seconds);
status_t setParamVideoEncoderProfile(int32_t profile);
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index 1b2fc5e..5a31b74 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -65,12 +65,12 @@
mUID(uid),
mFd(-1),
mDrmManagerClient(NULL),
- mMetaDataSize(-1ll),
mBitrate(-1ll),
mPollBufferingGeneration(0),
mPendingReadBufferTypes(0),
mBuffering(false),
- mPrepareBuffering(false) {
+ mPrepareBuffering(false),
+ mPrevBufferPercentage(-1) {
resetDataSource();
DataSource::RegisterDefaultSniffers();
}
@@ -130,23 +130,34 @@
status_t NuPlayer::GenericSource::initFromDataSource() {
sp<MediaExtractor> extractor;
+ String8 mimeType;
+ float confidence;
+ sp<AMessage> dummy;
+ bool isWidevineStreaming = false;
CHECK(mDataSource != NULL);
if (mIsWidevine) {
- String8 mimeType;
- float confidence;
- sp<AMessage> dummy;
- bool success;
-
- success = SniffWVM(mDataSource, &mimeType, &confidence, &dummy);
- if (!success
- || strcasecmp(
+ isWidevineStreaming = SniffWVM(
+ mDataSource, &mimeType, &confidence, &dummy);
+ if (!isWidevineStreaming ||
+ strcasecmp(
mimeType.string(), MEDIA_MIMETYPE_CONTAINER_WVM)) {
ALOGE("unsupported widevine mime: %s", mimeType.string());
return UNKNOWN_ERROR;
}
+ } else if (mIsStreaming) {
+ if (!mDataSource->sniff(&mimeType, &confidence, &dummy)) {
+ return UNKNOWN_ERROR;
+ }
+ isWidevineStreaming = !strcasecmp(
+ mimeType.string(), MEDIA_MIMETYPE_CONTAINER_WVM);
+ }
+ if (isWidevineStreaming) {
+ // we don't want cached source for widevine streaming.
+ mCachedSource.clear();
+ mDataSource = mHttpSource;
mWVMExtractor = new WVMExtractor(mDataSource);
mWVMExtractor->setAdaptiveStreamingMode(true);
if (mUIDValid) {
@@ -155,7 +166,7 @@
extractor = mWVMExtractor;
} else {
extractor = MediaExtractor::Create(mDataSource,
- mSniffedMIME.empty() ? NULL: mSniffedMIME.c_str());
+ mimeType.isEmpty() ? NULL : mimeType.string());
}
if (extractor == NULL) {
@@ -181,14 +192,6 @@
if (mFileMeta->findCString(kKeyMIMEType, &fileMime)
&& !strncasecmp(fileMime, "video/wvm", 9)) {
mIsWidevine = true;
- if (!mUri.empty()) {
- // streaming, but the app forgot to specify widevine:// url
- mWVMExtractor = static_cast<WVMExtractor *>(extractor.get());
- mWVMExtractor->setAdaptiveStreamingMode(true);
- if (mUIDValid) {
- mWVMExtractor->setUID(mUID);
- }
- }
}
}
}
@@ -332,7 +335,7 @@
mLooper->registerHandler(this);
}
- sp<AMessage> msg = new AMessage(kWhatPrepareAsync, id());
+ sp<AMessage> msg = new AMessage(kWhatPrepareAsync, this);
msg->post();
}
@@ -345,6 +348,7 @@
if (!mUri.empty()) {
const char* uri = mUri.c_str();
+ String8 contentType;
mIsWidevine = !strncasecmp(uri, "widevine://", 11);
if (!strncasecmp("http://", uri, 7)
@@ -359,7 +363,7 @@
}
mDataSource = DataSource::CreateFromURI(
- mHTTPService, uri, &mUriHeaders, &mContentType,
+ mHTTPService, uri, &mUriHeaders, &contentType,
static_cast<HTTPBase *>(mHttpSource.get()));
} else {
mIsWidevine = false;
@@ -387,20 +391,8 @@
mIsStreaming = (mIsWidevine || mCachedSource != NULL);
}
- // check initial caching status
- status_t err = prefillCacheIfNecessary();
- if (err != OK) {
- if (err == -EAGAIN) {
- (new AMessage(kWhatPrepareAsync, id()))->post(200000);
- } else {
- ALOGE("Failed to prefill data cache!");
- notifyPreparedAndCleanup(UNKNOWN_ERROR);
- }
- return;
- }
-
- // init extrator from data source
- err = initFromDataSource();
+ // init extractor from data source
+ status_t err = initFromDataSource();
if (err != OK) {
ALOGE("Failed to init from data source!");
@@ -429,7 +421,7 @@
if (mIsSecure) {
// secure decoders must be instantiated before starting widevine source
- sp<AMessage> reply = new AMessage(kWhatSecureDecodersInstantiated, id());
+ sp<AMessage> reply = new AMessage(kWhatSecureDecodersInstantiated, this);
notifyInstantiateSecureDecoders(reply);
} else {
finishPrepareAsync();
@@ -465,9 +457,6 @@
void NuPlayer::GenericSource::notifyPreparedAndCleanup(status_t err) {
if (err != OK) {
- mMetaDataSize = -1ll;
- mContentType = "";
- mSniffedMIME = "";
mDataSource.clear();
mCachedSource.clear();
mHttpSource.clear();
@@ -478,76 +467,6 @@
notifyPrepared(err);
}
-status_t NuPlayer::GenericSource::prefillCacheIfNecessary() {
- CHECK(mDataSource != NULL);
-
- if (mCachedSource == NULL) {
- // no prefill if the data source is not cached
- return OK;
- }
-
- // We're not doing this for streams that appear to be audio-only
- // streams to ensure that even low bandwidth streams start
- // playing back fairly instantly.
- if (!strncasecmp(mContentType.string(), "audio/", 6)) {
- return OK;
- }
-
- // We're going to prefill the cache before trying to instantiate
- // the extractor below, as the latter is an operation that otherwise
- // could block on the datasource for a significant amount of time.
- // During that time we'd be unable to abort the preparation phase
- // without this prefill.
-
- // Initially make sure we have at least 192 KB for the sniff
- // to complete without blocking.
- static const size_t kMinBytesForSniffing = 192 * 1024;
- static const size_t kDefaultMetaSize = 200000;
-
- status_t finalStatus;
-
- size_t cachedDataRemaining =
- mCachedSource->approxDataRemaining(&finalStatus);
-
- if (finalStatus != OK || (mMetaDataSize >= 0
- && (off64_t)cachedDataRemaining >= mMetaDataSize)) {
- ALOGV("stop caching, status %d, "
- "metaDataSize %lld, cachedDataRemaining %zu",
- finalStatus, mMetaDataSize, cachedDataRemaining);
- return OK;
- }
-
- ALOGV("now cached %zu bytes of data", cachedDataRemaining);
-
- if (mMetaDataSize < 0
- && cachedDataRemaining >= kMinBytesForSniffing) {
- String8 tmp;
- float confidence;
- sp<AMessage> meta;
- if (!mCachedSource->sniff(&tmp, &confidence, &meta)) {
- return UNKNOWN_ERROR;
- }
-
- // We successfully identified the file's extractor to
- // be, remember this mime type so we don't have to
- // sniff it again when we call MediaExtractor::Create()
- mSniffedMIME = tmp.string();
-
- if (meta == NULL
- || !meta->findInt64("meta-data-size",
- reinterpret_cast<int64_t*>(&mMetaDataSize))) {
- mMetaDataSize = kDefaultMetaSize;
- }
-
- if (mMetaDataSize < 0ll) {
- ALOGE("invalid metaDataSize = %lld bytes", mMetaDataSize);
- return UNKNOWN_ERROR;
- }
- }
-
- return -EAGAIN;
-}
-
void NuPlayer::GenericSource::start() {
ALOGI("start");
@@ -563,7 +482,7 @@
setDrmPlaybackStatusIfNeeded(Playback::START, getLastReadPosition() / 1000);
mStarted = true;
- (new AMessage(kWhatStart, id()))->post();
+ (new AMessage(kWhatStart, this))->post();
}
void NuPlayer::GenericSource::stop() {
@@ -572,7 +491,7 @@
mStarted = false;
if (mIsWidevine || mIsSecure) {
// For widevine or secure sources we need to prevent any further reads.
- sp<AMessage> msg = new AMessage(kWhatStopWidevine, id());
+ sp<AMessage> msg = new AMessage(kWhatStopWidevine, this);
sp<AMessage> response;
(void) msg->postAndAwaitResponse(&response);
}
@@ -589,7 +508,7 @@
setDrmPlaybackStatusIfNeeded(Playback::START, getLastReadPosition() / 1000);
mStarted = true;
- (new AMessage(kWhatResume, id()))->post();
+ (new AMessage(kWhatResume, this))->post();
}
void NuPlayer::GenericSource::disconnect() {
@@ -616,7 +535,7 @@
}
void NuPlayer::GenericSource::schedulePollBuffering() {
- sp<AMessage> msg = new AMessage(kWhatPollBuffering, id());
+ sp<AMessage> msg = new AMessage(kWhatPollBuffering, this);
msg->setInt32("generation", mPollBufferingGeneration);
msg->post(1000000ll);
}
@@ -624,6 +543,7 @@
void NuPlayer::GenericSource::cancelPollBuffering() {
mBuffering = false;
++mPollBufferingGeneration;
+ mPrevBufferPercentage = -1;
}
void NuPlayer::GenericSource::restartPollBuffering() {
@@ -633,7 +553,19 @@
}
}
-void NuPlayer::GenericSource::notifyBufferingUpdate(int percentage) {
+void NuPlayer::GenericSource::notifyBufferingUpdate(int32_t percentage) {
+ // Buffering percent could go backward as it's estimated from remaining
+ // data and last access time. This could cause the buffering position
+ // drawn on media control to jitter slightly. Remember previously reported
+ // percentage and don't allow it to go backward.
+ if (percentage < mPrevBufferPercentage) {
+ percentage = mPrevBufferPercentage;
+ } else if (percentage > 100) {
+ percentage = 100;
+ }
+
+ mPrevBufferPercentage = percentage;
+
ALOGV("notifyBufferingUpdate: buffering %d%%", percentage);
sp<AMessage> msg = dupNotify();
@@ -687,10 +619,10 @@
int32_t kbps = 0;
status_t err = UNKNOWN_ERROR;
- if (mCachedSource != NULL) {
- err = mCachedSource->getEstimatedBandwidthKbps(&kbps);
- } else if (mWVMExtractor != NULL) {
+ if (mWVMExtractor != NULL) {
err = mWVMExtractor->getEstimatedBandwidthKbps(&kbps);
+ } else if (mCachedSource != NULL) {
+ err = mCachedSource->getEstimatedBandwidthKbps(&kbps);
}
if (err == OK) {
@@ -712,7 +644,13 @@
int64_t cachedDurationUs = -1ll;
ssize_t cachedDataRemaining = -1;
- if (mCachedSource != NULL) {
+ ALOGW_IF(mWVMExtractor != NULL && mCachedSource != NULL,
+ "WVMExtractor and NuCachedSource both present");
+
+ if (mWVMExtractor != NULL) {
+ cachedDurationUs =
+ mWVMExtractor->getCachedDurationUs(&finalStatus);
+ } else if (mCachedSource != NULL) {
cachedDataRemaining =
mCachedSource->approxDataRemaining(&finalStatus);
@@ -728,9 +666,6 @@
cachedDurationUs = cachedDataRemaining * 8000000ll / bitrate;
}
}
- } else if (mWVMExtractor != NULL) {
- cachedDurationUs
- = mWVMExtractor->getCachedDurationUs(&finalStatus);
}
if (finalStatus != OK) {
@@ -918,7 +853,7 @@
mVideoTrack.mPackets->clear();
}
sp<AMessage> response = new AMessage;
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
response->postReply(replyID);
break;
@@ -958,7 +893,7 @@
const int64_t oneSecUs = 1000000ll;
delayUs -= oneSecUs;
}
- sp<AMessage> msg2 = new AMessage(sendWhat, id());
+ sp<AMessage> msg2 = new AMessage(sendWhat, this);
msg2->setInt32("generation", msgGeneration);
msg2->post(delayUs < 0 ? 0 : delayUs);
}
@@ -998,7 +933,7 @@
}
sp<MetaData> NuPlayer::GenericSource::getFormatMeta(bool audio) {
- sp<AMessage> msg = new AMessage(kWhatGetFormat, id());
+ sp<AMessage> msg = new AMessage(kWhatGetFormat, this);
msg->setInt32("audio", audio);
sp<AMessage> response;
@@ -1020,7 +955,7 @@
sp<MetaData> format = doGetFormatMeta(audio);
response->setPointer("format", format.get());
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
response->postReply(replyID);
}
@@ -1087,7 +1022,7 @@
if (mSubtitleTrack.mSource != NULL
&& !mSubtitleTrack.mPackets->hasBufferAvailable(&eosResult)) {
- sp<AMessage> msg = new AMessage(kWhatFetchSubtitleData, id());
+ sp<AMessage> msg = new AMessage(kWhatFetchSubtitleData, this);
msg->setInt64("timeUs", timeUs);
msg->setInt32("generation", mFetchSubtitleDataGeneration);
msg->post();
@@ -1095,7 +1030,7 @@
if (mTimedTextTrack.mSource != NULL
&& !mTimedTextTrack.mPackets->hasBufferAvailable(&eosResult)) {
- sp<AMessage> msg = new AMessage(kWhatFetchTimedTextData, id());
+ sp<AMessage> msg = new AMessage(kWhatFetchTimedTextData, this);
msg->setInt64("timeUs", timeUs);
msg->setInt32("generation", mFetchTimedTextDataGeneration);
msg->post();
@@ -1160,7 +1095,7 @@
}
ssize_t NuPlayer::GenericSource::getSelectedTrack(media_track_type type) const {
- sp<AMessage> msg = new AMessage(kWhatGetSelectedTrack, id());
+ sp<AMessage> msg = new AMessage(kWhatGetSelectedTrack, this);
msg->setInt32("type", type);
sp<AMessage> response;
@@ -1183,7 +1118,7 @@
ssize_t index = doGetSelectedTrack(type);
response->setInt32("index", index);
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
response->postReply(replyID);
}
@@ -1216,7 +1151,7 @@
status_t NuPlayer::GenericSource::selectTrack(size_t trackIndex, bool select, int64_t timeUs) {
ALOGV("%s track: %zu", select ? "select" : "deselect", trackIndex);
- sp<AMessage> msg = new AMessage(kWhatSelectTrack, id());
+ sp<AMessage> msg = new AMessage(kWhatSelectTrack, this);
msg->setInt32("trackIndex", trackIndex);
msg->setInt32("select", select);
msg->setInt64("timeUs", timeUs);
@@ -1241,7 +1176,7 @@
status_t err = doSelectTrack(trackIndex, select, timeUs);
response->setInt32("err", err);
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
response->postReply(replyID);
}
@@ -1302,7 +1237,7 @@
status_t eosResult; // ignored
if (mSubtitleTrack.mSource != NULL
&& !mSubtitleTrack.mPackets->hasBufferAvailable(&eosResult)) {
- sp<AMessage> msg = new AMessage(kWhatFetchSubtitleData, id());
+ sp<AMessage> msg = new AMessage(kWhatFetchSubtitleData, this);
msg->setInt64("timeUs", timeUs);
msg->setInt32("generation", mFetchSubtitleDataGeneration);
msg->post();
@@ -1310,7 +1245,7 @@
if (mTimedTextTrack.mSource != NULL
&& !mTimedTextTrack.mPackets->hasBufferAvailable(&eosResult)) {
- sp<AMessage> msg = new AMessage(kWhatFetchTimedTextData, id());
+ sp<AMessage> msg = new AMessage(kWhatFetchTimedTextData, this);
msg->setInt64("timeUs", timeUs);
msg->setInt32("generation", mFetchTimedTextDataGeneration);
msg->post();
@@ -1324,7 +1259,7 @@
return OK;
}
- sp<AMessage> msg = new AMessage(kWhatChangeAVSource, id());
+ sp<AMessage> msg = new AMessage(kWhatChangeAVSource, this);
msg->setInt32("trackIndex", trackIndex);
msg->post();
return OK;
@@ -1334,7 +1269,7 @@
}
status_t NuPlayer::GenericSource::seekTo(int64_t seekTimeUs) {
- sp<AMessage> msg = new AMessage(kWhatSeek, id());
+ sp<AMessage> msg = new AMessage(kWhatSeek, this);
msg->setInt64("seekTimeUs", seekTimeUs);
sp<AMessage> response;
@@ -1354,7 +1289,7 @@
status_t err = doSeek(seekTimeUs);
response->setInt32("err", err);
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
response->postReply(replyID);
}
@@ -1474,7 +1409,7 @@
if ((mPendingReadBufferTypes & (1 << trackType)) == 0) {
mPendingReadBufferTypes |= (1 << trackType);
- sp<AMessage> msg = new AMessage(kWhatReadBuffer, id());
+ sp<AMessage> msg = new AMessage(kWhatReadBuffer, this);
msg->setInt32("trackType", trackType);
msg->post();
}
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.h b/media/libmediaplayerservice/nuplayer/GenericSource.h
index 2d73ea9..862ee5f 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.h
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.h
@@ -140,14 +140,13 @@
sp<DecryptHandle> mDecryptHandle;
bool mStarted;
bool mStopRead;
- String8 mContentType;
- AString mSniffedMIME;
- off64_t mMetaDataSize;
int64_t mBitrate;
int32_t mPollBufferingGeneration;
uint32_t mPendingReadBufferTypes;
bool mBuffering;
bool mPrepareBuffering;
+ int32_t mPrevBufferPercentage;
+
mutable Mutex mReadBufferLock;
sp<ALooper> mLooper;
@@ -159,8 +158,6 @@
int64_t getLastReadPosition();
void setDrmPlaybackStatusIfNeeded(int playbackStatus, int64_t position);
- status_t prefillCacheIfNecessary();
-
void notifyPreparedAndCleanup(status_t err);
void onSecureDecodersInstantiated(status_t err);
void finishPrepareAsync();
@@ -204,7 +201,7 @@
void cancelPollBuffering();
void restartPollBuffering();
void onPollBuffering();
- void notifyBufferingUpdate(int percentage);
+ void notifyBufferingUpdate(int32_t percentage);
void startBufferingIfNecessary();
void stopBufferingIfNecessary();
void sendCacheStats();
diff --git a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
index a26ef9e..0476c9b 100644
--- a/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/HTTPLiveSource.cpp
@@ -81,7 +81,7 @@
mLiveLooper->registerHandler(this);
}
- sp<AMessage> notify = new AMessage(kWhatSessionNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatSessionNotify, this);
mLiveSession = new LiveSession(
notify,
@@ -153,7 +153,7 @@
if (err == OK) {
mFetchSubtitleDataGeneration++;
if (select) {
- sp<AMessage> msg = new AMessage(kWhatFetchSubtitleData, id());
+ sp<AMessage> msg = new AMessage(kWhatFetchSubtitleData, this);
msg->setInt32("generation", mFetchSubtitleDataGeneration);
msg->post();
}
@@ -281,6 +281,34 @@
break;
}
+ case LiveSession::kWhatBufferingStart:
+ {
+ sp<AMessage> notify = dupNotify();
+ notify->setInt32("what", kWhatPauseOnBufferingStart);
+ notify->post();
+ break;
+ }
+
+ case LiveSession::kWhatBufferingEnd:
+ {
+ sp<AMessage> notify = dupNotify();
+ notify->setInt32("what", kWhatResumeOnBufferingEnd);
+ notify->post();
+ break;
+ }
+
+
+ case LiveSession::kWhatBufferingUpdate:
+ {
+ sp<AMessage> notify = dupNotify();
+ int32_t percentage;
+ CHECK(msg->findInt32("percentage", &percentage));
+ notify->setInt32("what", kWhatBufferingUpdate);
+ notify->setInt32("percentage", percentage);
+ notify->post();
+ break;
+ }
+
case LiveSession::kWhatError:
{
break;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index aeea204..02d9f32 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -180,6 +180,7 @@
mFlushingVideo(NONE),
mResumePending(false),
mVideoScalingMode(NATIVE_WINDOW_SCALING_MODE_SCALE_TO_WINDOW),
+ mPlaybackRate(1.0),
mStarted(false),
mPaused(false),
mPausedByClient(false) {
@@ -199,9 +200,9 @@
}
void NuPlayer::setDataSourceAsync(const sp<IStreamSource> &source) {
- sp<AMessage> msg = new AMessage(kWhatSetDataSource, id());
+ sp<AMessage> msg = new AMessage(kWhatSetDataSource, this);
- sp<AMessage> notify = new AMessage(kWhatSourceNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatSourceNotify, this);
msg->setObject("source", new StreamingSource(notify, source));
msg->post();
@@ -229,10 +230,10 @@
const char *url,
const KeyedVector<String8, String8> *headers) {
- sp<AMessage> msg = new AMessage(kWhatSetDataSource, id());
+ sp<AMessage> msg = new AMessage(kWhatSetDataSource, this);
size_t len = strlen(url);
- sp<AMessage> notify = new AMessage(kWhatSourceNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatSourceNotify, this);
sp<Source> source;
if (IsHTTPLiveURL(url)) {
@@ -266,9 +267,9 @@
}
void NuPlayer::setDataSourceAsync(int fd, int64_t offset, int64_t length) {
- sp<AMessage> msg = new AMessage(kWhatSetDataSource, id());
+ sp<AMessage> msg = new AMessage(kWhatSetDataSource, this);
- sp<AMessage> notify = new AMessage(kWhatSourceNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatSourceNotify, this);
sp<GenericSource> source =
new GenericSource(notify, mUIDValid, mUID);
@@ -285,12 +286,12 @@
}
void NuPlayer::prepareAsync() {
- (new AMessage(kWhatPrepare, id()))->post();
+ (new AMessage(kWhatPrepare, this))->post();
}
void NuPlayer::setVideoSurfaceTextureAsync(
const sp<IGraphicBufferProducer> &bufferProducer) {
- sp<AMessage> msg = new AMessage(kWhatSetVideoNativeWindow, id());
+ sp<AMessage> msg = new AMessage(kWhatSetVideoNativeWindow, this);
if (bufferProducer == NULL) {
msg->setObject("native-window", NULL);
@@ -305,17 +306,23 @@
}
void NuPlayer::setAudioSink(const sp<MediaPlayerBase::AudioSink> &sink) {
- sp<AMessage> msg = new AMessage(kWhatSetAudioSink, id());
+ sp<AMessage> msg = new AMessage(kWhatSetAudioSink, this);
msg->setObject("sink", sink);
msg->post();
}
void NuPlayer::start() {
- (new AMessage(kWhatStart, id()))->post();
+ (new AMessage(kWhatStart, this))->post();
+}
+
+void NuPlayer::setPlaybackRate(float rate) {
+ sp<AMessage> msg = new AMessage(kWhatSetRate, this);
+ msg->setFloat("rate", rate);
+ msg->post();
}
void NuPlayer::pause() {
- (new AMessage(kWhatPause, id()))->post();
+ (new AMessage(kWhatPause, this))->post();
}
void NuPlayer::resetAsync() {
@@ -329,11 +336,11 @@
mSource->disconnect();
}
- (new AMessage(kWhatReset, id()))->post();
+ (new AMessage(kWhatReset, this))->post();
}
void NuPlayer::seekToAsync(int64_t seekTimeUs, bool needNotify) {
- sp<AMessage> msg = new AMessage(kWhatSeek, id());
+ sp<AMessage> msg = new AMessage(kWhatSeek, this);
msg->setInt64("seekTimeUs", seekTimeUs);
msg->setInt32("needNotify", needNotify);
msg->post();
@@ -401,7 +408,7 @@
case kWhatGetTrackInfo:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
Parcel* reply;
@@ -454,7 +461,7 @@
sp<AMessage> response = new AMessage;
response->setInt32("err", err);
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
response->postReply(replyID);
break;
@@ -462,7 +469,7 @@
case kWhatSelectTrack:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
size_t trackIndex;
@@ -604,6 +611,16 @@
break;
}
+ case kWhatSetRate:
+ {
+ ALOGV("kWhatSetRate");
+ CHECK(msg->findFloat("rate", &mPlaybackRate));
+ if (mRenderer != NULL) {
+ mRenderer->setPlaybackRate(mPlaybackRate);
+ }
+ break;
+ }
+
case kWhatScanSources:
{
int32_t generation;
@@ -1062,15 +1079,17 @@
flags |= Renderer::FLAG_OFFLOAD_AUDIO;
}
- sp<AMessage> notify = new AMessage(kWhatRendererNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatRendererNotify, this);
++mRendererGeneration;
notify->setInt32("generation", mRendererGeneration);
mRenderer = new Renderer(mAudioSink, notify, flags);
-
mRendererLooper = new ALooper;
mRendererLooper->setName("NuPlayerRenderer");
mRendererLooper->start(false, false, ANDROID_PRIORITY_AUDIO);
mRendererLooper->registerHandler(mRenderer);
+ if (mPlaybackRate != 1.0) {
+ mRenderer->setPlaybackRate(mPlaybackRate);
+ }
sp<MetaData> meta = getFileMeta();
int32_t rate;
@@ -1176,7 +1195,7 @@
return;
}
- sp<AMessage> msg = new AMessage(kWhatScanSources, id());
+ sp<AMessage> msg = new AMessage(kWhatScanSources, this);
msg->setInt32("generation", mScanSourcesGeneration);
msg->post();
@@ -1218,7 +1237,7 @@
AString mime;
CHECK(format->findString("mime", &mime));
- sp<AMessage> ccNotify = new AMessage(kWhatClosedCaptionNotify, id());
+ sp<AMessage> ccNotify = new AMessage(kWhatClosedCaptionNotify, this);
if (mCCDecoder == NULL) {
mCCDecoder = new CCDecoder(ccNotify);
}
@@ -1233,17 +1252,19 @@
}
if (audio) {
- sp<AMessage> notify = new AMessage(kWhatAudioNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatAudioNotify, this);
++mAudioDecoderGeneration;
notify->setInt32("generation", mAudioDecoderGeneration);
if (mOffloadAudio) {
+ const bool hasVideo = (mSource->getFormat(false /*audio */) != NULL);
+ format->setInt32("has-video", hasVideo);
*decoder = new DecoderPassThrough(notify, mSource, mRenderer);
} else {
*decoder = new Decoder(notify, mSource, mRenderer);
}
} else {
- sp<AMessage> notify = new AMessage(kWhatVideoNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatVideoNotify, this);
++mVideoDecoderGeneration;
notify->setInt32("generation", mVideoDecoderGeneration);
@@ -1434,7 +1455,7 @@
}
status_t NuPlayer::getTrackInfo(Parcel* reply) const {
- sp<AMessage> msg = new AMessage(kWhatGetTrackInfo, id());
+ sp<AMessage> msg = new AMessage(kWhatGetTrackInfo, this);
msg->setPointer("reply", reply);
sp<AMessage> response;
@@ -1443,7 +1464,7 @@
}
status_t NuPlayer::getSelectedTrack(int32_t type, Parcel* reply) const {
- sp<AMessage> msg = new AMessage(kWhatGetSelectedTrack, id());
+ sp<AMessage> msg = new AMessage(kWhatGetSelectedTrack, this);
msg->setPointer("reply", reply);
msg->setInt32("type", type);
@@ -1456,7 +1477,7 @@
}
status_t NuPlayer::selectTrack(size_t trackIndex, bool select, int64_t timeUs) {
- sp<AMessage> msg = new AMessage(kWhatSelectTrack, id());
+ sp<AMessage> msg = new AMessage(kWhatSelectTrack, this);
msg->setSize("trackIndex", trackIndex);
msg->setInt32("select", select);
msg->setInt64("timeUs", timeUs);
@@ -1499,7 +1520,7 @@
}
void NuPlayer::schedulePollDuration() {
- sp<AMessage> msg = new AMessage(kWhatPollDuration, id());
+ sp<AMessage> msg = new AMessage(kWhatPollDuration, this);
msg->setInt32("generation", mPollDurationGeneration);
msg->post();
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 30ede1a..2bc20d7 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -51,6 +51,7 @@
const sp<IGraphicBufferProducer> &bufferProducer);
void setAudioSink(const sp<MediaPlayerBase::AudioSink> &sink);
+ void setPlaybackRate(float rate);
void start();
void pause();
@@ -104,6 +105,7 @@
kWhatSetVideoNativeWindow = '=NaW',
kWhatSetAudioSink = '=AuS',
kWhatMoreDataQueued = 'more',
+ kWhatSetRate = 'setR',
kWhatStart = 'strt',
kWhatScanSources = 'scan',
kWhatVideoNotify = 'vidN',
@@ -175,6 +177,7 @@
int32_t mVideoScalingMode;
+ float mPlaybackRate;
bool mStarted;
// Actual pause state, either as requested by client or due to buffering.
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
index 9229704..cf3e8ad 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
@@ -19,6 +19,7 @@
#include <utils/Log.h>
#include <inttypes.h>
+#include "avc_utils.h"
#include "NuPlayerCCDecoder.h"
#include <media/stagefright/foundation/ABitReader.h>
@@ -185,17 +186,38 @@
// returns true if a new CC track is found
bool NuPlayer::CCDecoder::extractFromSEI(const sp<ABuffer> &accessUnit) {
- int64_t timeUs;
- CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
-
sp<ABuffer> sei;
if (!accessUnit->meta()->findBuffer("sei", &sei) || sei == NULL) {
return false;
}
+ int64_t timeUs;
+ CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
+
bool trackAdded = false;
- NALBitReader br(sei->data() + 1, sei->size() - 1);
+ const NALPosition *nal = (NALPosition *) sei->data();
+
+ for (size_t i = 0; i < sei->size() / sizeof(NALPosition); ++i, ++nal) {
+ trackAdded |= parseSEINalUnit(
+ timeUs, accessUnit->data() + nal->nalOffset, nal->nalSize);
+ }
+
+ return trackAdded;
+}
+
+// returns true if a new CC track is found
+bool NuPlayer::CCDecoder::parseSEINalUnit(
+ int64_t timeUs, const uint8_t *nalStart, size_t nalSize) {
+ unsigned nalType = nalStart[0] & 0x1f;
+
+ // the buffer should only have SEI in it
+ if (nalType != 6) {
+ return false;
+ }
+
+ bool trackAdded = false;
+ NALBitReader br(nalStart + 1, nalSize - 1);
// sei_message()
while (br.atLeastNumBitsLeft(16)) { // at least 16-bit for sei_message()
uint32_t payload_type = 0;
@@ -214,20 +236,25 @@
// sei_payload()
if (payload_type == 4) {
- // user_data_registered_itu_t_t35()
+ bool isCC = false;
+ if (payload_size > 1 + 2 + 4 + 1) {
+ // user_data_registered_itu_t_t35()
- // ATSC A/72: 6.4.2
- uint8_t itu_t_t35_country_code = br.getBits(8);
- uint16_t itu_t_t35_provider_code = br.getBits(16);
- uint32_t user_identifier = br.getBits(32);
- uint8_t user_data_type_code = br.getBits(8);
+ // ATSC A/72: 6.4.2
+ uint8_t itu_t_t35_country_code = br.getBits(8);
+ uint16_t itu_t_t35_provider_code = br.getBits(16);
+ uint32_t user_identifier = br.getBits(32);
+ uint8_t user_data_type_code = br.getBits(8);
- payload_size -= 1 + 2 + 4 + 1;
+ payload_size -= 1 + 2 + 4 + 1;
- if (itu_t_t35_country_code == 0xB5
- && itu_t_t35_provider_code == 0x0031
- && user_identifier == 'GA94'
- && user_data_type_code == 0x3) {
+ isCC = itu_t_t35_country_code == 0xB5
+ && itu_t_t35_provider_code == 0x0031
+ && user_identifier == 'GA94'
+ && user_data_type_code == 0x3;
+ }
+
+ if (isCC && payload_size > 2) {
// MPEG_cc_data()
// ATSC A/53 Part 4: 6.2.3.1
br.skipBits(1); //process_em_data_flag
@@ -243,7 +270,7 @@
sp<ABuffer> ccBuf = new ABuffer(cc_count * sizeof(CCData));
ccBuf->setRange(0, 0);
- for (size_t i = 0; i < cc_count; i++) {
+ for (size_t i = 0; i < cc_count && payload_size >= 3; i++) {
uint8_t marker = br.getBits(5);
CHECK_EQ(marker, 0x1f);
@@ -253,6 +280,8 @@
uint8_t cc_data_1 = br.getBits(8) & 0x7f;
uint8_t cc_data_2 = br.getBits(8) & 0x7f;
+ payload_size -= 3;
+
if (cc_valid
&& (cc_type == 0 || cc_type == 1)) {
CCData cc(cc_type, cc_data_1, cc_data_2);
@@ -269,7 +298,6 @@
}
}
}
- payload_size -= cc_count * 3;
mCCMap.add(timeUs, ccBuf);
break;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h
index 5e06f4e..77fb0fe 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h
@@ -49,6 +49,7 @@
bool isTrackValid(size_t index) const;
int32_t getTrackIndex(size_t channel) const;
bool extractFromSEI(const sp<ABuffer> &accessUnit);
+ bool parseSEINalUnit(int64_t timeUs, const uint8_t *nalStart, size_t nalSize);
sp<ABuffer> filterCCBuf(const sp<ABuffer> &ccBuf, size_t index);
DISALLOW_EVIL_CONSTRUCTORS(CCDecoder);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 5d98d98..04ac699 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -56,6 +56,7 @@
mIsVideoAVC(false),
mIsSecure(false),
mFormatChangePending(false),
+ mTimeChangePending(false),
mPaused(true),
mResumePending(false),
mComponentName("decoder") {
@@ -121,6 +122,7 @@
CHECK(mCodec == NULL);
mFormatChangePending = false;
+ mTimeChangePending = false;
++mBufferGeneration;
@@ -235,7 +237,7 @@
}
}
-void NuPlayer::Decoder::onFlush(bool notifyComplete) {
+void NuPlayer::Decoder::doFlush(bool notifyComplete) {
if (mCCDecoder != NULL) {
mCCDecoder->flush();
}
@@ -259,13 +261,22 @@
// we attempt to release the buffers even if flush fails.
}
releaseAndResetMediaBuffers();
+}
- if (notifyComplete) {
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatFlushCompleted);
- notify->post();
- mPaused = true;
+void NuPlayer::Decoder::onFlush() {
+ doFlush(true);
+
+ if (isDiscontinuityPending()) {
+ // This could happen if the client starts seeking/shutdown
+ // after we queued an EOS for discontinuities.
+ // We can consider discontinuity handled.
+ finishHandleDiscontinuity(false /* flushOnTimeChange */);
}
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatFlushCompleted);
+ notify->post();
+ mPaused = true;
}
void NuPlayer::Decoder::onShutdown(bool notifyComplete) {
@@ -309,16 +320,17 @@
}
void NuPlayer::Decoder::doRequestBuffers() {
- if (mFormatChangePending) {
+ if (isDiscontinuityPending()) {
return;
}
status_t err = OK;
- while (!mDequeuedInputBuffers.empty()) {
+ while (err == OK && !mDequeuedInputBuffers.empty()) {
size_t bufferIx = *mDequeuedInputBuffers.begin();
sp<AMessage> msg = new AMessage();
msg->setSize("buffer-ix", bufferIx);
err = fetchInputData(msg);
- if (err != OK) {
+ if (err != OK && err != ERROR_END_OF_STREAM) {
+ // if EOS, need to queue EOS buffer
break;
}
mDequeuedInputBuffers.erase(mDequeuedInputBuffers.begin());
@@ -336,7 +348,7 @@
}
bool NuPlayer::Decoder::handleAnInputBuffer() {
- if (mFormatChangePending) {
+ if (isDiscontinuityPending()) {
return false;
}
size_t bufferIx = -1;
@@ -391,9 +403,6 @@
}
bool NuPlayer::Decoder::handleAnOutputBuffer() {
- if (mFormatChangePending) {
- return false;
- }
size_t bufferIx = -1;
size_t offset;
size_t size;
@@ -474,17 +483,20 @@
buffer->setRange(offset, size);
buffer->meta()->clear();
buffer->meta()->setInt64("timeUs", timeUs);
- if (flags & MediaCodec::BUFFER_FLAG_EOS) {
- buffer->meta()->setInt32("eos", true);
- notifyResumeCompleteIfNecessary();
- }
+
+ bool eos = flags & MediaCodec::BUFFER_FLAG_EOS;
// we do not expect CODECCONFIG or SYNCFRAME for decoder
- sp<AMessage> reply = new AMessage(kWhatRenderBuffer, id());
+ sp<AMessage> reply = new AMessage(kWhatRenderBuffer, this);
reply->setSize("buffer-ix", bufferIx);
reply->setInt32("generation", mBufferGeneration);
- if (mSkipRenderingUntilMediaTimeUs >= 0) {
+ if (eos) {
+ ALOGI("[%s] saw output EOS", mIsAudio ? "audio" : "video");
+
+ buffer->meta()->setInt32("eos", true);
+ reply->setInt32("eos", true);
+ } else if (mSkipRenderingUntilMediaTimeUs >= 0) {
if (timeUs < mSkipRenderingUntilMediaTimeUs) {
ALOGV("[%s] dropping buffer at time %lld as requested.",
mComponentName.c_str(), (long long)timeUs);
@@ -502,7 +514,7 @@
if (mRenderer != NULL) {
// send the buffer to renderer.
mRenderer->queueBuffer(mIsAudio, buffer, reply);
- if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+ if (eos && !isDiscontinuityPending()) {
mRenderer->queueEOS(mIsAudio, ERROR_END_OF_STREAM);
}
}
@@ -533,11 +545,8 @@
}
void NuPlayer::Decoder::requestCodecNotification() {
- if (mFormatChangePending) {
- return;
- }
if (mCodec != NULL) {
- sp<AMessage> reply = new AMessage(kWhatCodecNotify, id());
+ sp<AMessage> reply = new AMessage(kWhatCodecNotify, this);
reply->setInt32("generation", mBufferGeneration);
mCodec->requestActivityNotification(reply);
}
@@ -582,39 +591,31 @@
formatChange = !seamlessFormatChange;
}
- if (formatChange || timeChange) {
- sp<AMessage> msg = mNotify->dup();
- msg->setInt32("what", kWhatInputDiscontinuity);
- msg->setInt32("formatChange", formatChange);
- msg->post();
- }
-
+ // For format or time change, return EOS to queue EOS input,
+ // then wait for EOS on output.
if (formatChange /* not seamless */) {
- // must change decoder
- // return EOS and wait to be killed
mFormatChangePending = true;
- return ERROR_END_OF_STREAM;
+ err = ERROR_END_OF_STREAM;
} else if (timeChange) {
- // need to flush
- // TODO: Ideally we shouldn't need a flush upon time
- // discontinuity, flushing will cause loss of frames.
- // We probably should queue a time change marker to the
- // output queue, and handles it in renderer instead.
rememberCodecSpecificData(newFormat);
- onFlush(false /* notifyComplete */);
- err = OK;
+ mTimeChangePending = true;
+ err = ERROR_END_OF_STREAM;
} else if (seamlessFormatChange) {
// reuse existing decoder and don't flush
rememberCodecSpecificData(newFormat);
- err = OK;
+ continue;
} else {
// This stream is unaffected by the discontinuity
return -EWOULDBLOCK;
}
}
+ // reply should only be returned without a buffer set
+ // when there is an error (including EOS)
+ CHECK(err != OK);
+
reply->setInt32("err", err);
- return OK;
+ return ERROR_END_OF_STREAM;
}
if (!mIsAudio) {
@@ -636,7 +637,7 @@
#if 0
int64_t mediaTimeUs;
CHECK(accessUnit->meta()->findInt64("timeUs", &mediaTimeUs));
- ALOGV("feeding %s input buffer at media time %.2f secs",
+ ALOGV("[%s] feeding input buffer at media time %" PRId64,
mIsAudio ? "audio" : "video",
mediaTimeUs / 1E6);
#endif
@@ -696,10 +697,7 @@
int32_t streamErr = ERROR_END_OF_STREAM;
CHECK(msg->findInt32("err", &streamErr) || !hasBuffer);
- if (streamErr == OK) {
- /* buffers are returned to hold on to */
- return true;
- }
+ CHECK(streamErr != OK);
// attempt to queue EOS
status_t err = mCodec->queueInputBuffer(
@@ -781,6 +779,7 @@
status_t err;
int32_t render;
size_t bufferIx;
+ int32_t eos;
CHECK(msg->findSize("buffer-ix", &bufferIx));
if (!mIsAudio) {
@@ -805,6 +804,42 @@
mComponentName.c_str(), err);
handleError(err);
}
+ if (msg->findInt32("eos", &eos) && eos
+ && isDiscontinuityPending()) {
+ finishHandleDiscontinuity(true /* flushOnTimeChange */);
+ }
+}
+
+bool NuPlayer::Decoder::isDiscontinuityPending() const {
+ return mFormatChangePending || mTimeChangePending;
+}
+
+void NuPlayer::Decoder::finishHandleDiscontinuity(bool flushOnTimeChange) {
+ ALOGV("finishHandleDiscontinuity: format %d, time %d, flush %d",
+ mFormatChangePending, mTimeChangePending, flushOnTimeChange);
+
+ // If we have format change, pause and wait to be killed;
+ // If we have time change only, flush and restart fetching.
+
+ if (mFormatChangePending) {
+ mPaused = true;
+ } else if (mTimeChangePending) {
+ if (flushOnTimeChange) {
+ doFlush(false /*notifyComplete*/);
+ }
+
+ // restart fetching input
+ scheduleRequestBuffers();
+ }
+
+ // Notify NuPlayer to either shutdown decoder, or rescan sources
+ sp<AMessage> msg = mNotify->dup();
+ msg->setInt32("what", kWhatInputDiscontinuity);
+ msg->setInt32("formatChange", mFormatChangePending);
+ msg->post();
+
+ mFormatChangePending = false;
+ mTimeChangePending = false;
}
bool NuPlayer::Decoder::supportsSeamlessAudioFormatChange(
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
index 1bfa94f..4aab2c6 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
@@ -43,7 +43,7 @@
virtual void onSetRenderer(const sp<Renderer> &renderer);
virtual void onGetInputBuffers(Vector<sp<ABuffer> > *dstBuffers);
virtual void onResume(bool notifyComplete);
- virtual void onFlush(bool notifyComplete);
+ virtual void onFlush();
virtual void onShutdown(bool notifyComplete);
virtual void doRequestBuffers();
@@ -81,6 +81,7 @@
bool mIsVideoAVC;
bool mIsSecure;
bool mFormatChangePending;
+ bool mTimeChangePending;
bool mPaused;
bool mResumePending;
@@ -93,6 +94,7 @@
void requestCodecNotification();
bool isStaleReply(const sp<AMessage> &msg);
+ void doFlush(bool notifyComplete);
status_t fetchInputData(sp<AMessage> &reply);
bool onInputBufferFetched(const sp<AMessage> &msg);
void onRenderBuffer(const sp<AMessage> &msg);
@@ -100,6 +102,8 @@
bool supportsSeamlessFormatChange(const sp<AMessage> &to) const;
bool supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const;
void rememberCodecSpecificData(const sp<AMessage> &format);
+ bool isDiscontinuityPending() const;
+ void finishHandleDiscontinuity(bool flushOnTimeChange);
void notifyResumeCompleteIfNecessary();
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.cpp
index d56fc4d..4636f0a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.cpp
@@ -61,7 +61,7 @@
}
void NuPlayer::DecoderBase::configure(const sp<AMessage> &format) {
- sp<AMessage> msg = new AMessage(kWhatConfigure, id());
+ sp<AMessage> msg = new AMessage(kWhatConfigure, this);
msg->setMessage("format", format);
msg->post();
}
@@ -71,13 +71,13 @@
}
void NuPlayer::DecoderBase::setRenderer(const sp<Renderer> &renderer) {
- sp<AMessage> msg = new AMessage(kWhatSetRenderer, id());
+ sp<AMessage> msg = new AMessage(kWhatSetRenderer, this);
msg->setObject("renderer", renderer);
msg->post();
}
status_t NuPlayer::DecoderBase::getInputBuffers(Vector<sp<ABuffer> > *buffers) const {
- sp<AMessage> msg = new AMessage(kWhatGetInputBuffers, id());
+ sp<AMessage> msg = new AMessage(kWhatGetInputBuffers, this);
msg->setPointer("buffers", buffers);
sp<AMessage> response;
@@ -85,17 +85,17 @@
}
void NuPlayer::DecoderBase::signalFlush() {
- (new AMessage(kWhatFlush, id()))->post();
+ (new AMessage(kWhatFlush, this))->post();
}
void NuPlayer::DecoderBase::signalResume(bool notifyComplete) {
- sp<AMessage> msg = new AMessage(kWhatResume, id());
+ sp<AMessage> msg = new AMessage(kWhatResume, this);
msg->setInt32("notifyComplete", notifyComplete);
msg->post();
}
void NuPlayer::DecoderBase::initiateShutdown() {
- (new AMessage(kWhatShutdown, id()))->post();
+ (new AMessage(kWhatShutdown, this))->post();
}
void NuPlayer::DecoderBase::onRequestInputBuffers() {
@@ -111,7 +111,7 @@
return;
}
mRequestInputBuffersPending = true;
- sp<AMessage> msg = new AMessage(kWhatRequestInputBuffers, id());
+ sp<AMessage> msg = new AMessage(kWhatRequestInputBuffers, this);
msg->post(10 * 1000ll);
}
@@ -136,7 +136,7 @@
case kWhatGetInputBuffers:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
Vector<sp<ABuffer> > *dstBuffers;
@@ -157,7 +157,7 @@
case kWhatFlush:
{
- onFlush(true);
+ onFlush();
break;
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.h
index 6732ff4..97e9269 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.h
@@ -65,7 +65,7 @@
virtual void onSetRenderer(const sp<Renderer> &renderer) = 0;
virtual void onGetInputBuffers(Vector<sp<ABuffer> > *dstBuffers) = 0;
virtual void onResume(bool notifyComplete) = 0;
- virtual void onFlush(bool notifyComplete) = 0;
+ virtual void onFlush() = 0;
virtual void onShutdown(bool notifyComplete) = 0;
void onRequestInputBuffers();
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp
index 9f7f09a..563de5e 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp
@@ -74,11 +74,14 @@
onRequestInputBuffers();
+ int32_t hasVideo = 0;
+ format->findInt32("has-video", &hasVideo);
+
// The audio sink is already opened before the PassThrough decoder is created.
// Opening again might be relevant if decoder is instantiated after shutdown and
// format is different.
status_t err = mRenderer->openAudioSink(
- format, true /* offloadOnly */, false /* hasVideo */,
+ format, true /* offloadOnly */, hasVideo,
AUDIO_OUTPUT_FLAG_NONE /* flags */, NULL /* isOffloaded */);
if (err != OK) {
handleError(err);
@@ -247,7 +250,7 @@
}
if (timeChange) {
- onFlush(false /* notifyComplete */);
+ doFlush(false /* notifyComplete */);
err = OK;
} else if (formatChange) {
// do seamless format change
@@ -333,7 +336,7 @@
return;
}
- sp<AMessage> reply = new AMessage(kWhatBufferConsumed, id());
+ sp<AMessage> reply = new AMessage(kWhatBufferConsumed, this);
reply->setInt32("generation", mBufferGeneration);
reply->setInt32("size", bufferSize);
@@ -364,7 +367,7 @@
}
}
-void NuPlayer::DecoderPassThrough::onFlush(bool notifyComplete) {
+void NuPlayer::DecoderPassThrough::doFlush(bool notifyComplete) {
++mBufferGeneration;
mSkipRenderingUntilMediaTimeUs = -1;
mPendingAudioAccessUnit.clear();
@@ -376,18 +379,21 @@
mRenderer->signalTimeDiscontinuity();
}
- if (notifyComplete) {
- mPaused = true;
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatFlushCompleted);
- notify->post();
- }
-
mPendingBuffersToDrain = 0;
mCachedBytes = 0;
mReachedEOS = false;
}
+void NuPlayer::DecoderPassThrough::onFlush() {
+ doFlush(true /* notifyComplete */);
+
+ mPaused = true;
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatFlushCompleted);
+ notify->post();
+
+}
+
void NuPlayer::DecoderPassThrough::onShutdown(bool notifyComplete) {
++mBufferGeneration;
mSkipRenderingUntilMediaTimeUs = -1;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.h
index a6e1faf..173cfbd 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.h
@@ -43,7 +43,7 @@
virtual void onSetRenderer(const sp<Renderer> &renderer);
virtual void onGetInputBuffers(Vector<sp<ABuffer> > *dstBuffers);
virtual void onResume(bool notifyComplete);
- virtual void onFlush(bool notifyComplete);
+ virtual void onFlush();
virtual void onShutdown(bool notifyComplete);
virtual void doRequestBuffers();
@@ -77,6 +77,7 @@
status_t dequeueAccessUnit(sp<ABuffer> *accessUnit);
sp<ABuffer> aggregateBuffer(const sp<ABuffer> &accessUnit);
status_t fetchInputData(sp<AMessage> &reply);
+ void doFlush(bool notifyComplete);
void onInputBufferFetched(const sp<AMessage> &msg);
void onBufferConsumed(int32_t size);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index bc79fdb..1fa9cef 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -341,6 +341,11 @@
return mState == STATE_RUNNING && !mAtEOS;
}
+status_t NuPlayerDriver::setPlaybackRate(float rate) {
+ mPlayer->setPlaybackRate(rate);
+ return OK;
+}
+
status_t NuPlayerDriver::seekTo(int msec) {
ALOGD("seekTo(%p) %d ms", this, msec);
Mutex::Autolock autoLock(mLock);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
index 5cba7d9..e53abcd 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.h
@@ -47,6 +47,7 @@
virtual status_t stop();
virtual status_t pause();
virtual bool isPlaying();
+ virtual status_t setPlaybackRate(float rate);
virtual status_t seekTo(int msec);
virtual status_t getCurrentPosition(int *msec);
virtual status_t getDuration(int *msec);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 25225a8..a2ec51c 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -25,6 +25,7 @@
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/foundation/AUtils.h>
#include <media/stagefright/foundation/AWakeLock.h>
+#include <media/stagefright/MediaClock.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/Utils.h>
@@ -63,22 +64,19 @@
mDrainVideoQueuePending(false),
mAudioQueueGeneration(0),
mVideoQueueGeneration(0),
+ mAudioDrainGeneration(0),
+ mVideoDrainGeneration(0),
+ mPlaybackRate(1.0),
mAudioFirstAnchorTimeMediaUs(-1),
mAnchorTimeMediaUs(-1),
- mAnchorTimeRealUs(-1),
mAnchorNumFramesWritten(-1),
- mAnchorMaxMediaUs(-1),
mVideoLateByUs(0ll),
mHasAudio(false),
mHasVideo(false),
- mPauseStartedTimeRealUs(-1),
- mFlushingAudio(false),
- mFlushingVideo(false),
mNotifyCompleteAudio(false),
mNotifyCompleteVideo(false),
mSyncQueues(false),
mPaused(false),
- mPausePositionMediaTimeUs(-1),
mVideoSampleReceived(false),
mVideoRenderingStarted(false),
mVideoRenderingStartGeneration(0),
@@ -90,7 +88,7 @@
mTotalBuffersQueued(0),
mLastAudioBufferDrained(0),
mWakeLock(new AWakeLock()) {
-
+ mMediaClock = new MediaClock;
}
NuPlayer::Renderer::~Renderer() {
@@ -105,7 +103,8 @@
bool audio,
const sp<ABuffer> &buffer,
const sp<AMessage> ¬ifyConsumed) {
- sp<AMessage> msg = new AMessage(kWhatQueueBuffer, id());
+ sp<AMessage> msg = new AMessage(kWhatQueueBuffer, this);
+ msg->setInt32("queueGeneration", getQueueGeneration(audio));
msg->setInt32("audio", static_cast<int32_t>(audio));
msg->setBuffer("buffer", buffer);
msg->setMessage("notifyConsumed", notifyConsumed);
@@ -115,199 +114,108 @@
void NuPlayer::Renderer::queueEOS(bool audio, status_t finalResult) {
CHECK_NE(finalResult, (status_t)OK);
- sp<AMessage> msg = new AMessage(kWhatQueueEOS, id());
+ sp<AMessage> msg = new AMessage(kWhatQueueEOS, this);
+ msg->setInt32("queueGeneration", getQueueGeneration(audio));
msg->setInt32("audio", static_cast<int32_t>(audio));
msg->setInt32("finalResult", finalResult);
msg->post();
}
+void NuPlayer::Renderer::setPlaybackRate(float rate) {
+ sp<AMessage> msg = new AMessage(kWhatSetRate, this);
+ msg->setFloat("rate", rate);
+ msg->post();
+}
+
void NuPlayer::Renderer::flush(bool audio, bool notifyComplete) {
{
- Mutex::Autolock autoLock(mFlushLock);
+ Mutex::Autolock autoLock(mLock);
if (audio) {
mNotifyCompleteAudio |= notifyComplete;
- if (mFlushingAudio) {
- return;
- }
- mFlushingAudio = true;
+ ++mAudioQueueGeneration;
+ ++mAudioDrainGeneration;
} else {
mNotifyCompleteVideo |= notifyComplete;
- if (mFlushingVideo) {
- return;
- }
- mFlushingVideo = true;
+ ++mVideoQueueGeneration;
+ ++mVideoDrainGeneration;
}
+
+ clearAnchorTime_l();
+ clearAudioFirstAnchorTime_l();
+ mVideoLateByUs = 0;
+ mSyncQueues = false;
}
- sp<AMessage> msg = new AMessage(kWhatFlush, id());
+ sp<AMessage> msg = new AMessage(kWhatFlush, this);
msg->setInt32("audio", static_cast<int32_t>(audio));
msg->post();
}
void NuPlayer::Renderer::signalTimeDiscontinuity() {
- Mutex::Autolock autoLock(mLock);
- // CHECK(mAudioQueue.empty());
- // CHECK(mVideoQueue.empty());
- setAudioFirstAnchorTime(-1);
- setAnchorTime(-1, -1);
- setVideoLateByUs(0);
- mSyncQueues = false;
-}
-
-void NuPlayer::Renderer::signalAudioSinkChanged() {
- (new AMessage(kWhatAudioSinkChanged, id()))->post();
}
void NuPlayer::Renderer::signalDisableOffloadAudio() {
- (new AMessage(kWhatDisableOffloadAudio, id()))->post();
+ (new AMessage(kWhatDisableOffloadAudio, this))->post();
}
void NuPlayer::Renderer::signalEnableOffloadAudio() {
- (new AMessage(kWhatEnableOffloadAudio, id()))->post();
+ (new AMessage(kWhatEnableOffloadAudio, this))->post();
}
void NuPlayer::Renderer::pause() {
- (new AMessage(kWhatPause, id()))->post();
+ (new AMessage(kWhatPause, this))->post();
}
void NuPlayer::Renderer::resume() {
- (new AMessage(kWhatResume, id()))->post();
+ (new AMessage(kWhatResume, this))->post();
}
void NuPlayer::Renderer::setVideoFrameRate(float fps) {
- sp<AMessage> msg = new AMessage(kWhatSetVideoFrameRate, id());
+ sp<AMessage> msg = new AMessage(kWhatSetVideoFrameRate, this);
msg->setFloat("frame-rate", fps);
msg->post();
}
-// Called on any threads, except renderer's thread.
-status_t NuPlayer::Renderer::getCurrentPosition(int64_t *mediaUs) {
- {
- Mutex::Autolock autoLock(mLock);
- int64_t currentPositionUs;
- if (getCurrentPositionIfPaused_l(¤tPositionUs)) {
- *mediaUs = currentPositionUs;
- return OK;
- }
- }
- return getCurrentPositionFromAnchor(mediaUs, ALooper::GetNowUs());
-}
-
-// Called on only renderer's thread.
-status_t NuPlayer::Renderer::getCurrentPositionOnLooper(int64_t *mediaUs) {
- return getCurrentPositionOnLooper(mediaUs, ALooper::GetNowUs());
-}
-
-// Called on only renderer's thread.
-// Since mPaused and mPausePositionMediaTimeUs are changed only on renderer's
-// thread, no need to acquire mLock.
-status_t NuPlayer::Renderer::getCurrentPositionOnLooper(
- int64_t *mediaUs, int64_t nowUs, bool allowPastQueuedVideo) {
- int64_t currentPositionUs;
- if (getCurrentPositionIfPaused_l(¤tPositionUs)) {
- *mediaUs = currentPositionUs;
- return OK;
- }
- return getCurrentPositionFromAnchor(mediaUs, nowUs, allowPastQueuedVideo);
-}
-
-// Called either with mLock acquired or on renderer's thread.
-bool NuPlayer::Renderer::getCurrentPositionIfPaused_l(int64_t *mediaUs) {
- if (!mPaused || mPausePositionMediaTimeUs < 0ll) {
- return false;
- }
- *mediaUs = mPausePositionMediaTimeUs;
- return true;
-}
-
// Called on any threads.
-status_t NuPlayer::Renderer::getCurrentPositionFromAnchor(
- int64_t *mediaUs, int64_t nowUs, bool allowPastQueuedVideo) {
- Mutex::Autolock autoLock(mTimeLock);
- if (!mHasAudio && !mHasVideo) {
- return NO_INIT;
- }
-
- if (mAnchorTimeMediaUs < 0) {
- return NO_INIT;
- }
-
- int64_t positionUs = (nowUs - mAnchorTimeRealUs) + mAnchorTimeMediaUs;
-
- if (mPauseStartedTimeRealUs != -1) {
- positionUs -= (nowUs - mPauseStartedTimeRealUs);
- }
-
- // limit position to the last queued media time (for video only stream
- // position will be discrete as we don't know how long each frame lasts)
- if (mAnchorMaxMediaUs >= 0 && !allowPastQueuedVideo) {
- if (positionUs > mAnchorMaxMediaUs) {
- positionUs = mAnchorMaxMediaUs;
- }
- }
-
- if (positionUs < mAudioFirstAnchorTimeMediaUs) {
- positionUs = mAudioFirstAnchorTimeMediaUs;
- }
-
- *mediaUs = (positionUs <= 0) ? 0 : positionUs;
- return OK;
+status_t NuPlayer::Renderer::getCurrentPosition(int64_t *mediaUs) {
+ return mMediaClock->getMediaTime(ALooper::GetNowUs(), mediaUs);
}
-void NuPlayer::Renderer::setHasMedia(bool audio) {
- Mutex::Autolock autoLock(mTimeLock);
- if (audio) {
- mHasAudio = true;
- } else {
- mHasVideo = true;
- }
+void NuPlayer::Renderer::clearAudioFirstAnchorTime_l() {
+ mAudioFirstAnchorTimeMediaUs = -1;
+ mMediaClock->setStartingTimeMedia(-1);
}
-void NuPlayer::Renderer::setAudioFirstAnchorTime(int64_t mediaUs) {
- Mutex::Autolock autoLock(mTimeLock);
- mAudioFirstAnchorTimeMediaUs = mediaUs;
-}
-
-void NuPlayer::Renderer::setAudioFirstAnchorTimeIfNeeded(int64_t mediaUs) {
- Mutex::Autolock autoLock(mTimeLock);
+void NuPlayer::Renderer::setAudioFirstAnchorTimeIfNeeded_l(int64_t mediaUs) {
if (mAudioFirstAnchorTimeMediaUs == -1) {
mAudioFirstAnchorTimeMediaUs = mediaUs;
+ mMediaClock->setStartingTimeMedia(mediaUs);
}
}
-void NuPlayer::Renderer::setAnchorTime(
- int64_t mediaUs, int64_t realUs, int64_t numFramesWritten, bool resume) {
- Mutex::Autolock autoLock(mTimeLock);
- mAnchorTimeMediaUs = mediaUs;
- mAnchorTimeRealUs = realUs;
- mAnchorNumFramesWritten = numFramesWritten;
- if (resume) {
- mPauseStartedTimeRealUs = -1;
- }
+void NuPlayer::Renderer::clearAnchorTime_l() {
+ mMediaClock->clearAnchor();
+ mAnchorTimeMediaUs = -1;
+ mAnchorNumFramesWritten = -1;
}
void NuPlayer::Renderer::setVideoLateByUs(int64_t lateUs) {
- Mutex::Autolock autoLock(mTimeLock);
+ Mutex::Autolock autoLock(mLock);
mVideoLateByUs = lateUs;
}
int64_t NuPlayer::Renderer::getVideoLateByUs() {
- Mutex::Autolock autoLock(mTimeLock);
+ Mutex::Autolock autoLock(mLock);
return mVideoLateByUs;
}
-void NuPlayer::Renderer::setPauseStartedTimeRealUs(int64_t realUs) {
- Mutex::Autolock autoLock(mTimeLock);
- mPauseStartedTimeRealUs = realUs;
-}
-
status_t NuPlayer::Renderer::openAudioSink(
const sp<AMessage> &format,
bool offloadOnly,
bool hasVideo,
uint32_t flags,
bool *isOffloaded) {
- sp<AMessage> msg = new AMessage(kWhatOpenAudioSink, id());
+ sp<AMessage> msg = new AMessage(kWhatOpenAudioSink, this);
msg->setMessage("format", format);
msg->setInt32("offload-only", offloadOnly);
msg->setInt32("has-video", hasVideo);
@@ -328,7 +236,7 @@
}
void NuPlayer::Renderer::closeAudioSink() {
- sp<AMessage> msg = new AMessage(kWhatCloseAudioSink, id());
+ sp<AMessage> msg = new AMessage(kWhatCloseAudioSink, this);
sp<AMessage> response;
msg->postAndAwaitResponse(&response);
@@ -356,7 +264,7 @@
response->setInt32("err", err);
response->setInt32("offload", offloadingAudio());
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
response->postReply(replyID);
@@ -365,7 +273,7 @@
case kWhatCloseAudioSink:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
onCloseAudioSink();
@@ -384,8 +292,8 @@
case kWhatDrainAudioQueue:
{
int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- if (generation != mAudioQueueGeneration) {
+ CHECK(msg->findInt32("drainGeneration", &generation));
+ if (generation != getDrainGeneration(true /* audio */)) {
break;
}
@@ -407,9 +315,7 @@
// Let's give it more data after about half that time
// has elapsed.
- // kWhatDrainAudioQueue is used for non-offloading mode,
- // and mLock is used only for offloading mode. Therefore,
- // no need to acquire mLock here.
+ Mutex::Autolock autoLock(mLock);
postDrainAudioQueue_l(delayUs / 2);
}
break;
@@ -418,8 +324,8 @@
case kWhatDrainVideoQueue:
{
int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- if (generation != mVideoQueueGeneration) {
+ CHECK(msg->findInt32("drainGeneration", &generation));
+ if (generation != getDrainGeneration(false /* audio */)) {
break;
}
@@ -427,22 +333,20 @@
onDrainVideoQueue();
- Mutex::Autolock autoLock(mLock);
- postDrainVideoQueue_l();
+ postDrainVideoQueue();
break;
}
case kWhatPostDrainVideoQueue:
{
int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- if (generation != mVideoQueueGeneration) {
+ CHECK(msg->findInt32("drainGeneration", &generation));
+ if (generation != getDrainGeneration(false /* audio */)) {
break;
}
mDrainVideoQueuePending = false;
- Mutex::Autolock autoLock(mLock);
- postDrainVideoQueue_l();
+ postDrainVideoQueue();
break;
}
@@ -458,15 +362,19 @@
break;
}
- case kWhatFlush:
+ case kWhatSetRate:
{
- onFlush(msg);
+ CHECK(msg->findFloat("rate", &mPlaybackRate));
+ int32_t ratePermille = (int32_t)(0.5f + 1000 * mPlaybackRate);
+ mPlaybackRate = ratePermille / 1000.0f;
+ mMediaClock->setPlaybackRate(mPlaybackRate);
+ mAudioSink->setPlaybackRatePermille(ratePermille);
break;
}
- case kWhatAudioSinkChanged:
+ case kWhatFlush:
{
- onAudioSinkChanged();
+ onFlush(msg);
break;
}
@@ -511,7 +419,7 @@
case kWhatAudioOffloadPauseTimeout:
{
int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
+ CHECK(msg->findInt32("drainGeneration", &generation));
if (generation != mAudioOffloadPauseTimeoutGeneration) {
break;
}
@@ -538,19 +446,19 @@
}
mDrainAudioQueuePending = true;
- sp<AMessage> msg = new AMessage(kWhatDrainAudioQueue, id());
- msg->setInt32("generation", mAudioQueueGeneration);
+ sp<AMessage> msg = new AMessage(kWhatDrainAudioQueue, this);
+ msg->setInt32("drainGeneration", mAudioDrainGeneration);
msg->post(delayUs);
}
-void NuPlayer::Renderer::prepareForMediaRenderingStart() {
- mAudioRenderingStartGeneration = mAudioQueueGeneration;
- mVideoRenderingStartGeneration = mVideoQueueGeneration;
+void NuPlayer::Renderer::prepareForMediaRenderingStart_l() {
+ mAudioRenderingStartGeneration = mAudioDrainGeneration;
+ mVideoRenderingStartGeneration = mVideoDrainGeneration;
}
-void NuPlayer::Renderer::notifyIfMediaRenderingStarted() {
- if (mVideoRenderingStartGeneration == mVideoQueueGeneration &&
- mAudioRenderingStartGeneration == mAudioQueueGeneration) {
+void NuPlayer::Renderer::notifyIfMediaRenderingStarted_l() {
+ if (mVideoRenderingStartGeneration == mVideoDrainGeneration &&
+ mAudioRenderingStartGeneration == mAudioDrainGeneration) {
mVideoRenderingStartGeneration = -1;
mAudioRenderingStartGeneration = -1;
@@ -618,7 +526,7 @@
int64_t mediaTimeUs;
CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
ALOGV("rendering audio at media time %.2f secs", mediaTimeUs / 1E6);
- setAudioFirstAnchorTimeIfNeeded(mediaTimeUs);
+ setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs);
}
size_t copy = entry->mBuffer->size() - entry->mOffset;
@@ -638,34 +546,45 @@
entry = NULL;
}
sizeCopied += copy;
- notifyIfMediaRenderingStarted();
+
+ notifyIfMediaRenderingStarted_l();
}
if (mAudioFirstAnchorTimeMediaUs >= 0) {
int64_t nowUs = ALooper::GetNowUs();
- setAnchorTime(mAudioFirstAnchorTimeMediaUs, nowUs - getPlayedOutAudioDurationUs(nowUs));
+ int64_t nowMediaUs =
+ mAudioFirstAnchorTimeMediaUs + getPlayedOutAudioDurationUs(nowUs);
+ // we don't know how much data we are queueing for offloaded tracks.
+ mMediaClock->updateAnchor(nowMediaUs, nowUs, INT64_MAX);
}
- // we don't know how much data we are queueing for offloaded tracks
- mAnchorMaxMediaUs = -1;
-
if (hasEOS) {
- (new AMessage(kWhatStopAudioSink, id()))->post();
+ (new AMessage(kWhatStopAudioSink, this))->post();
}
return sizeCopied;
}
bool NuPlayer::Renderer::onDrainAudioQueue() {
+ // TODO: This call to getPosition checks if AudioTrack has been created
+ // in AudioSink before draining audio. If AudioTrack doesn't exist, then
+ // CHECKs on getPosition will fail.
+ // We still need to figure out why AudioTrack is not created when
+ // this function is called. One possible reason could be leftover
+ // audio. Another possible place is to check whether decoder
+ // has received INFO_FORMAT_CHANGED as the first buffer since
+ // AudioSink is opened there, and possible interactions with flush
+ // immediately after start. Investigate error message
+ // "vorbis_dsp_synthesis returned -135", along with RTSP.
uint32_t numFramesPlayed;
if (mAudioSink->getPosition(&numFramesPlayed) != OK) {
return false;
}
+#if 0
ssize_t numFramesAvailableToWrite =
mAudioSink->frameCount() - (mNumFramesWritten - numFramesPlayed);
-#if 0
if (numFramesAvailableToWrite == mAudioSink->frameCount()) {
ALOGI("audio sink underrun");
} else {
@@ -674,10 +593,7 @@
}
#endif
- size_t numBytesAvailableToWrite =
- numFramesAvailableToWrite * mAudioSink->frameSize();
-
- while (numBytesAvailableToWrite > 0 && !mAudioQueue.empty()) {
+ while (!mAudioQueue.empty()) {
QueueEntry *entry = &*mAudioQueue.begin();
mLastAudioBufferDrained = entry->mBufferOrdinal;
@@ -710,14 +626,16 @@
}
size_t copy = entry->mBuffer->size() - entry->mOffset;
- if (copy > numBytesAvailableToWrite) {
- copy = numBytesAvailableToWrite;
- }
- ssize_t written = mAudioSink->write(entry->mBuffer->data() + entry->mOffset, copy);
+ ssize_t written = mAudioSink->write(entry->mBuffer->data() + entry->mOffset,
+ copy, false /* blocking */);
if (written < 0) {
// An error in AudioSink write. Perhaps the AudioSink was not properly opened.
- ALOGE("AudioSink write error(%zd) when writing %zu bytes", written, copy);
+ if (written == WOULD_BLOCK) {
+ ALOGW("AudioSink write would block when writing %zu bytes", copy);
+ } else {
+ ALOGE("AudioSink write error(%zd) when writing %zu bytes", written, copy);
+ }
break;
}
@@ -729,73 +647,92 @@
entry = NULL;
}
- numBytesAvailableToWrite -= written;
size_t copiedFrames = written / mAudioSink->frameSize();
mNumFramesWritten += copiedFrames;
- notifyIfMediaRenderingStarted();
+ {
+ Mutex::Autolock autoLock(mLock);
+ notifyIfMediaRenderingStarted_l();
+ }
if (written != (ssize_t)copy) {
// A short count was received from AudioSink::write()
//
- // AudioSink write should block until exactly the number of bytes are delivered.
- // But it may return with a short count (without an error) when:
+ // AudioSink write is called in non-blocking mode.
+ // It may return with a short count when:
//
// 1) Size to be copied is not a multiple of the frame size. We consider this fatal.
- // 2) AudioSink is an AudioCache for data retrieval, and the AudioCache is exceeded.
+ // 2) The data to be copied exceeds the available buffer in AudioSink.
+ // 3) An error occurs and data has been partially copied to the buffer in AudioSink.
+ // 4) AudioSink is an AudioCache for data retrieval, and the AudioCache is exceeded.
// (Case 1)
// Must be a multiple of the frame size. If it is not a multiple of a frame size, it
// needs to fail, as we should not carry over fractional frames between calls.
CHECK_EQ(copy % mAudioSink->frameSize(), 0);
- // (Case 2)
+ // (Case 2, 3, 4)
// Return early to the caller.
// Beware of calling immediately again as this may busy-loop if you are not careful.
- ALOGW("AudioSink write short frame count %zd < %zu", written, copy);
+ ALOGV("AudioSink write short frame count %zd < %zu", written, copy);
break;
}
}
- mAnchorMaxMediaUs =
- mAnchorTimeMediaUs +
- (int64_t)(max((long long)mNumFramesWritten - mAnchorNumFramesWritten, 0LL)
- * 1000LL * mAudioSink->msecsPerFrame());
+ int64_t maxTimeMedia;
+ {
+ Mutex::Autolock autoLock(mLock);
+ maxTimeMedia =
+ mAnchorTimeMediaUs +
+ (int64_t)(max((long long)mNumFramesWritten - mAnchorNumFramesWritten, 0LL)
+ * 1000LL * mAudioSink->msecsPerFrame());
+ }
+ mMediaClock->updateMaxTimeMedia(maxTimeMedia);
return !mAudioQueue.empty();
}
+int64_t NuPlayer::Renderer::getDurationUsIfPlayedAtSampleRate(uint32_t numFrames) {
+ int32_t sampleRate = offloadingAudio() ?
+ mCurrentOffloadInfo.sample_rate : mCurrentPcmInfo.mSampleRate;
+ // TODO: remove the (int32_t) casting below as it may overflow at 12.4 hours.
+ return (int64_t)((int32_t)numFrames * 1000000LL / sampleRate);
+}
+
+// Calculate duration of pending samples if played at normal rate (i.e., 1.0).
int64_t NuPlayer::Renderer::getPendingAudioPlayoutDurationUs(int64_t nowUs) {
- int64_t writtenAudioDurationUs =
- mNumFramesWritten * 1000LL * mAudioSink->msecsPerFrame();
+ int64_t writtenAudioDurationUs = getDurationUsIfPlayedAtSampleRate(mNumFramesWritten);
return writtenAudioDurationUs - getPlayedOutAudioDurationUs(nowUs);
}
int64_t NuPlayer::Renderer::getRealTimeUs(int64_t mediaTimeUs, int64_t nowUs) {
- int64_t currentPositionUs;
- if (mPaused || getCurrentPositionOnLooper(
- ¤tPositionUs, nowUs, true /* allowPastQueuedVideo */) != OK) {
- // If failed to get current position, e.g. due to audio clock is not ready, then just
- // play out video immediately without delay.
+ int64_t realUs;
+ if (mMediaClock->getRealTimeFor(mediaTimeUs, &realUs) != OK) {
+ // If failed to get current position, e.g. due to audio clock is
+ // not ready, then just play out video immediately without delay.
return nowUs;
}
- return (mediaTimeUs - currentPositionUs) + nowUs;
+ return realUs;
}
void NuPlayer::Renderer::onNewAudioMediaTime(int64_t mediaTimeUs) {
+ Mutex::Autolock autoLock(mLock);
// TRICKY: vorbis decoder generates multiple frames with the same
// timestamp, so only update on the first frame with a given timestamp
if (mediaTimeUs == mAnchorTimeMediaUs) {
return;
}
- setAudioFirstAnchorTimeIfNeeded(mediaTimeUs);
+ setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs);
int64_t nowUs = ALooper::GetNowUs();
- setAnchorTime(
- mediaTimeUs, nowUs + getPendingAudioPlayoutDurationUs(nowUs), mNumFramesWritten);
+ int64_t nowMediaUs = mediaTimeUs - getPendingAudioPlayoutDurationUs(nowUs);
+ mMediaClock->updateAnchor(nowMediaUs, nowUs, mediaTimeUs);
+ mAnchorNumFramesWritten = mNumFramesWritten;
+ mAnchorTimeMediaUs = mediaTimeUs;
}
-void NuPlayer::Renderer::postDrainVideoQueue_l() {
+// Called without mLock acquired.
+void NuPlayer::Renderer::postDrainVideoQueue() {
if (mDrainVideoQueuePending
- || mSyncQueues
+ || getSyncQueues()
|| (mPaused && mVideoSampleReceived)) {
return;
}
@@ -806,8 +743,8 @@
QueueEntry &entry = *mVideoQueue.begin();
- sp<AMessage> msg = new AMessage(kWhatDrainVideoQueue, id());
- msg->setInt32("generation", mVideoQueueGeneration);
+ sp<AMessage> msg = new AMessage(kWhatDrainVideoQueue, this);
+ msg->setInt32("drainGeneration", getDrainGeneration(false /* audio */));
if (entry.mBuffer == NULL) {
// EOS doesn't carry a timestamp.
@@ -827,16 +764,19 @@
int64_t mediaTimeUs;
CHECK(entry.mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
- if (mAnchorTimeMediaUs < 0) {
- setAnchorTime(mediaTimeUs, nowUs);
- mPausePositionMediaTimeUs = mediaTimeUs;
- mAnchorMaxMediaUs = mediaTimeUs;
- realTimeUs = nowUs;
- } else {
- realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
+ {
+ Mutex::Autolock autoLock(mLock);
+ if (mAnchorTimeMediaUs < 0) {
+ mMediaClock->updateAnchor(mediaTimeUs, nowUs, mediaTimeUs);
+ mAnchorTimeMediaUs = mediaTimeUs;
+ realTimeUs = nowUs;
+ } else {
+ realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
+ }
}
if (!mHasAudio) {
- mAnchorMaxMediaUs = mediaTimeUs + 100000; // smooth out videos >= 10fps
+ // smooth out videos >= 10fps
+ mMediaClock->updateMaxTimeMedia(mediaTimeUs + 100000);
}
// Heuristics to handle situation when media time changed without a
@@ -915,16 +855,19 @@
ALOGV("video late by %lld us (%.2f secs)",
mVideoLateByUs, mVideoLateByUs / 1E6);
} else {
+ int64_t mediaUs = 0;
+ mMediaClock->getMediaTime(realTimeUs, &mediaUs);
ALOGV("rendering video at media time %.2f secs",
(mFlags & FLAG_REAL_TIME ? realTimeUs :
- (realTimeUs + mAnchorTimeMediaUs - mAnchorTimeRealUs)) / 1E6);
+ mediaUs) / 1E6);
}
} else {
setVideoLateByUs(0);
if (!mVideoSampleReceived && !mHasAudio) {
// This will ensure that the first frame after a flush won't be used as anchor
// when renderer is in paused state, because resume can happen any time after seek.
- setAnchorTime(-1, -1);
+ Mutex::Autolock autoLock(mLock);
+ clearAnchorTime_l();
}
}
@@ -941,7 +884,8 @@
mVideoRenderingStarted = true;
notifyVideoRenderingStart();
}
- notifyIfMediaRenderingStarted();
+ Mutex::Autolock autoLock(mLock);
+ notifyIfMediaRenderingStarted_l();
}
}
@@ -960,14 +904,22 @@
}
void NuPlayer::Renderer::notifyAudioOffloadTearDown() {
- (new AMessage(kWhatAudioOffloadTearDown, id()))->post();
+ (new AMessage(kWhatAudioOffloadTearDown, this))->post();
}
void NuPlayer::Renderer::onQueueBuffer(const sp<AMessage> &msg) {
int32_t audio;
CHECK(msg->findInt32("audio", &audio));
- setHasMedia(audio);
+ if (dropBufferIfStale(audio, msg)) {
+ return;
+ }
+
+ if (audio) {
+ mHasAudio = true;
+ } else {
+ mHasVideo = true;
+ }
if (mHasVideo) {
if (mVideoScheduler == NULL) {
@@ -976,10 +928,6 @@
}
}
- if (dropBufferWhileFlushing(audio, msg)) {
- return;
- }
-
sp<ABuffer> buffer;
CHECK(msg->findBuffer("buffer", &buffer));
@@ -993,15 +941,16 @@
entry.mFinalResult = OK;
entry.mBufferOrdinal = ++mTotalBuffersQueued;
- Mutex::Autolock autoLock(mLock);
if (audio) {
+ Mutex::Autolock autoLock(mLock);
mAudioQueue.push_back(entry);
postDrainAudioQueue_l();
} else {
mVideoQueue.push_back(entry);
- postDrainVideoQueue_l();
+ postDrainVideoQueue();
}
+ Mutex::Autolock autoLock(mLock);
if (!mSyncQueues || mAudioQueue.empty() || mVideoQueue.empty()) {
return;
}
@@ -1050,7 +999,9 @@
}
if (!mVideoQueue.empty()) {
- postDrainVideoQueue_l();
+ mLock.unlock();
+ postDrainVideoQueue();
+ mLock.lock();
}
}
@@ -1058,7 +1009,7 @@
int32_t audio;
CHECK(msg->findInt32("audio", &audio));
- if (dropBufferWhileFlushing(audio, msg)) {
+ if (dropBufferIfStale(audio, msg)) {
return;
}
@@ -1069,19 +1020,20 @@
entry.mOffset = 0;
entry.mFinalResult = finalResult;
- Mutex::Autolock autoLock(mLock);
if (audio) {
+ Mutex::Autolock autoLock(mLock);
if (mAudioQueue.empty() && mSyncQueues) {
syncQueuesDone_l();
}
mAudioQueue.push_back(entry);
postDrainAudioQueue_l();
} else {
- if (mVideoQueue.empty() && mSyncQueues) {
+ if (mVideoQueue.empty() && getSyncQueues()) {
+ Mutex::Autolock autoLock(mLock);
syncQueuesDone_l();
}
mVideoQueue.push_back(entry);
- postDrainVideoQueue_l();
+ postDrainVideoQueue();
}
}
@@ -1090,31 +1042,25 @@
CHECK(msg->findInt32("audio", &audio));
{
- Mutex::Autolock autoLock(mFlushLock);
+ Mutex::Autolock autoLock(mLock);
if (audio) {
- mFlushingAudio = false;
notifyComplete = mNotifyCompleteAudio;
mNotifyCompleteAudio = false;
} else {
- mFlushingVideo = false;
notifyComplete = mNotifyCompleteVideo;
mNotifyCompleteVideo = false;
}
- }
- // If we're currently syncing the queues, i.e. dropping audio while
- // aligning the first audio/video buffer times and only one of the
- // two queues has data, we may starve that queue by not requesting
- // more buffers from the decoder. If the other source then encounters
- // a discontinuity that leads to flushing, we'll never find the
- // corresponding discontinuity on the other queue.
- // Therefore we'll stop syncing the queues if at least one of them
- // is flushed.
- {
- Mutex::Autolock autoLock(mLock);
- syncQueuesDone_l();
- setPauseStartedTimeRealUs(-1);
- setAnchorTime(-1, -1);
+ // If we're currently syncing the queues, i.e. dropping audio while
+ // aligning the first audio/video buffer times and only one of the
+ // two queues has data, we may starve that queue by not requesting
+ // more buffers from the decoder. If the other source then encounters
+ // a discontinuity that leads to flushing, we'll never find the
+ // corresponding discontinuity on the other queue.
+ // Therefore we'll stop syncing the queues if at least one of them
+ // is flushed.
+ syncQueuesDone_l();
+ clearAnchorTime_l();
}
ALOGV("flushing %s", audio ? "audio" : "video");
@@ -1123,11 +1069,11 @@
Mutex::Autolock autoLock(mLock);
flushQueue(&mAudioQueue);
- ++mAudioQueueGeneration;
- prepareForMediaRenderingStart();
+ ++mAudioDrainGeneration;
+ prepareForMediaRenderingStart_l();
if (offloadingAudio()) {
- setAudioFirstAnchorTime(-1);
+ clearAudioFirstAnchorTime_l();
}
}
@@ -1142,13 +1088,14 @@
flushQueue(&mVideoQueue);
mDrainVideoQueuePending = false;
- ++mVideoQueueGeneration;
if (mVideoScheduler != NULL) {
mVideoScheduler->restart();
}
- prepareForMediaRenderingStart();
+ Mutex::Autolock autoLock(mLock);
+ ++mVideoDrainGeneration;
+ prepareForMediaRenderingStart_l();
}
mVideoSampleReceived = false;
@@ -1178,20 +1125,12 @@
notify->post();
}
-bool NuPlayer::Renderer::dropBufferWhileFlushing(
+bool NuPlayer::Renderer::dropBufferIfStale(
bool audio, const sp<AMessage> &msg) {
- bool flushing = false;
+ int32_t queueGeneration;
+ CHECK(msg->findInt32("queueGeneration", &queueGeneration));
- {
- Mutex::Autolock autoLock(mFlushLock);
- if (audio) {
- flushing = mFlushingAudio;
- } else {
- flushing = mFlushingVideo;
- }
- }
-
- if (!flushing) {
+ if (queueGeneration == getQueueGeneration(audio)) {
return false;
}
@@ -1209,7 +1148,10 @@
}
CHECK(!mDrainAudioQueuePending);
mNumFramesWritten = 0;
- mAnchorNumFramesWritten = -1;
+ {
+ Mutex::Autolock autoLock(mLock);
+ mAnchorNumFramesWritten = -1;
+ }
uint32_t written;
if (mAudioSink->getFramesWritten(&written) == OK) {
mNumFramesWritten = written;
@@ -1219,13 +1161,13 @@
void NuPlayer::Renderer::onDisableOffloadAudio() {
Mutex::Autolock autoLock(mLock);
mFlags &= ~FLAG_OFFLOAD_AUDIO;
- ++mAudioQueueGeneration;
+ ++mAudioDrainGeneration;
}
void NuPlayer::Renderer::onEnableOffloadAudio() {
Mutex::Autolock autoLock(mLock);
mFlags |= FLAG_OFFLOAD_AUDIO;
- ++mAudioQueueGeneration;
+ ++mAudioDrainGeneration;
}
void NuPlayer::Renderer::onPause() {
@@ -1234,25 +1176,13 @@
return;
}
int64_t currentPositionUs;
- int64_t pausePositionMediaTimeUs;
- if (getCurrentPositionFromAnchor(
- ¤tPositionUs, ALooper::GetNowUs()) == OK) {
- pausePositionMediaTimeUs = currentPositionUs;
- } else {
- // Set paused position to -1 (unavailabe) if we don't have anchor time
- // This could happen if client does a seekTo() immediately followed by
- // pause(). Renderer will be flushed with anchor time cleared. We don't
- // want to leave stale value in mPausePositionMediaTimeUs.
- pausePositionMediaTimeUs = -1;
- }
{
Mutex::Autolock autoLock(mLock);
- mPausePositionMediaTimeUs = pausePositionMediaTimeUs;
- ++mAudioQueueGeneration;
- ++mVideoQueueGeneration;
- prepareForMediaRenderingStart();
+ ++mAudioDrainGeneration;
+ ++mVideoDrainGeneration;
+ prepareForMediaRenderingStart_l();
mPaused = true;
- setPauseStartedTimeRealUs(ALooper::GetNowUs());
+ mMediaClock->setPlaybackRate(0.0);
}
mDrainAudioQueuePending = false;
@@ -1277,21 +1207,18 @@
mAudioSink->start();
}
- Mutex::Autolock autoLock(mLock);
- mPaused = false;
- if (mPauseStartedTimeRealUs != -1) {
- int64_t newAnchorRealUs =
- mAnchorTimeRealUs + ALooper::GetNowUs() - mPauseStartedTimeRealUs;
- setAnchorTime(
- mAnchorTimeMediaUs, newAnchorRealUs, mAnchorNumFramesWritten, true /* resume */);
- }
+ {
+ Mutex::Autolock autoLock(mLock);
+ mPaused = false;
+ mMediaClock->setPlaybackRate(mPlaybackRate);
- if (!mAudioQueue.empty()) {
- postDrainAudioQueue_l();
+ if (!mAudioQueue.empty()) {
+ postDrainAudioQueue_l();
+ }
}
if (!mVideoQueue.empty()) {
- postDrainVideoQueue_l();
+ postDrainVideoQueue();
}
}
@@ -1302,6 +1229,21 @@
mVideoScheduler->init(fps);
}
+int32_t NuPlayer::Renderer::getQueueGeneration(bool audio) {
+ Mutex::Autolock autoLock(mLock);
+ return (audio ? mAudioQueueGeneration : mVideoQueueGeneration);
+}
+
+int32_t NuPlayer::Renderer::getDrainGeneration(bool audio) {
+ Mutex::Autolock autoLock(mLock);
+ return (audio ? mAudioDrainGeneration : mVideoDrainGeneration);
+}
+
+bool NuPlayer::Renderer::getSyncQueues() {
+ Mutex::Autolock autoLock(mLock);
+ return mSyncQueues;
+}
+
// TODO: Remove unnecessary calls to getPlayedOutAudioDurationUs()
// as it acquires locks and may query the audio driver.
//
@@ -1309,6 +1251,7 @@
// accessing getTimestamp() or getPosition() every time a data buffer with
// a media time is received.
//
+// Calculate duration of played samples if played at normal rate (i.e., 1.0).
int64_t NuPlayer::Renderer::getPlayedOutAudioDurationUs(int64_t nowUs) {
uint32_t numFramesPlayed;
int64_t numFramesPlayedAt;
@@ -1346,9 +1289,8 @@
//ALOGD("getPosition: %d %lld", numFramesPlayed, numFramesPlayedAt);
}
- // TODO: remove the (int32_t) casting below as it may overflow at 12.4 hours.
//CHECK_EQ(numFramesPlayed & (1 << 31), 0); // can't be negative until 12.4 hrs, test
- int64_t durationUs = (int64_t)((int32_t)numFramesPlayed * 1000LL * mAudioSink->msecsPerFrame())
+ int64_t durationUs = getDurationUsIfPlayedAtSampleRate(numFramesPlayed)
+ nowUs - numFramesPlayedAt;
if (durationUs < 0) {
// Occurs when numFramesPlayed position is very small and the following:
@@ -1373,7 +1315,7 @@
mAudioOffloadTornDown = true;
int64_t currentPositionUs;
- if (getCurrentPositionOnLooper(¤tPositionUs) != OK) {
+ if (getCurrentPosition(¤tPositionUs) != OK) {
currentPositionUs = 0;
}
@@ -1390,8 +1332,8 @@
void NuPlayer::Renderer::startAudioOffloadPauseTimeout() {
if (offloadingAudio()) {
mWakeLock->acquire();
- sp<AMessage> msg = new AMessage(kWhatAudioOffloadPauseTimeout, id());
- msg->setInt32("generation", mAudioOffloadPauseTimeoutGeneration);
+ sp<AMessage> msg = new AMessage(kWhatAudioOffloadPauseTimeout, this);
+ msg->setInt32("drainGeneration", mAudioOffloadPauseTimeoutGeneration);
msg->post(kOffloadPauseMaxUs);
}
}
@@ -1487,6 +1429,10 @@
&offloadInfo);
if (err == OK) {
+ if (mPlaybackRate != 1.0) {
+ mAudioSink->setPlaybackRatePermille(
+ (int32_t)(mPlaybackRate * 1000 + 0.5f));
+ }
// If the playback is offloaded to h/w, we pass
// the HAL some metadata information.
// We don't want to do this for PCM because it
@@ -1542,6 +1488,10 @@
return err;
}
mCurrentPcmInfo = info;
+ if (mPlaybackRate != 1.0) {
+ mAudioSink->setPlaybackRatePermille(
+ (int32_t)(mPlaybackRate * 1000 + 0.5f));
+ }
mAudioSink->start();
}
if (audioSinkChanged) {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
index 003d1d0..38843d5 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
@@ -24,6 +24,7 @@
struct ABuffer;
class AWakeLock;
+struct MediaClock;
struct VideoFrameScheduler;
struct NuPlayer::Renderer : public AHandler {
@@ -47,6 +48,8 @@
void queueEOS(bool audio, status_t finalResult);
+ void setPlaybackRate(float rate);
+
void flush(bool audio, bool notifyComplete);
void signalTimeDiscontinuity();
@@ -61,16 +64,8 @@
void setVideoFrameRate(float fps);
- // Following setters and getters are protected by mTimeLock.
status_t getCurrentPosition(int64_t *mediaUs);
- void setHasMedia(bool audio);
- void setAudioFirstAnchorTime(int64_t mediaUs);
- void setAudioFirstAnchorTimeIfNeeded(int64_t mediaUs);
- void setAnchorTime(
- int64_t mediaUs, int64_t realUs, int64_t numFramesWritten = -1, bool resume = false);
- void setVideoLateByUs(int64_t lateUs);
int64_t getVideoLateByUs();
- void setPauseStartedTimeRealUs(int64_t realUs);
status_t openAudioSink(
const sp<AMessage> &format,
@@ -107,8 +102,8 @@
kWhatPostDrainVideoQueue = 'pDVQ',
kWhatQueueBuffer = 'queB',
kWhatQueueEOS = 'qEOS',
+ kWhatSetRate = 'setR',
kWhatFlush = 'flus',
- kWhatAudioSinkChanged = 'auSC',
kWhatPause = 'paus',
kWhatResume = 'resm',
kWhatOpenAudioSink = 'opnA',
@@ -142,26 +137,18 @@
bool mDrainVideoQueuePending;
int32_t mAudioQueueGeneration;
int32_t mVideoQueueGeneration;
+ int32_t mAudioDrainGeneration;
+ int32_t mVideoDrainGeneration;
- Mutex mTimeLock;
- // |mTimeLock| protects the following 7 member vars that are related to time.
- // Note: those members are only written on Renderer thread, so reading on Renderer thread
- // doesn't need to be protected. Otherwise accessing those members must be protected by
- // |mTimeLock|.
- // TODO: move those members to a seperated media clock class.
+ sp<MediaClock> mMediaClock;
+ float mPlaybackRate;
int64_t mAudioFirstAnchorTimeMediaUs;
int64_t mAnchorTimeMediaUs;
- int64_t mAnchorTimeRealUs;
int64_t mAnchorNumFramesWritten;
- int64_t mAnchorMaxMediaUs;
int64_t mVideoLateByUs;
bool mHasAudio;
bool mHasVideo;
- int64_t mPauseStartedTimeRealUs;
- Mutex mFlushLock; // protects the following 2 member vars.
- bool mFlushingAudio;
- bool mFlushingVideo;
bool mNotifyCompleteAudio;
bool mNotifyCompleteVideo;
@@ -169,7 +156,6 @@
// modified on only renderer's thread.
bool mPaused;
- int64_t mPausePositionMediaTimeUs;
bool mVideoSampleReceived;
bool mVideoRenderingStarted;
@@ -211,14 +197,19 @@
int64_t getPlayedOutAudioDurationUs(int64_t nowUs);
void postDrainAudioQueue_l(int64_t delayUs = 0);
+ void clearAnchorTime_l();
+ void clearAudioFirstAnchorTime_l();
+ void setAudioFirstAnchorTimeIfNeeded_l(int64_t mediaUs);
+ void setVideoLateByUs(int64_t lateUs);
+
void onNewAudioMediaTime(int64_t mediaTimeUs);
int64_t getRealTimeUs(int64_t mediaTimeUs, int64_t nowUs);
void onDrainVideoQueue();
- void postDrainVideoQueue_l();
+ void postDrainVideoQueue();
- void prepareForMediaRenderingStart();
- void notifyIfMediaRenderingStarted();
+ void prepareForMediaRenderingStart_l();
+ void notifyIfMediaRenderingStarted_l();
void onQueueBuffer(const sp<AMessage> &msg);
void onQueueEOS(const sp<AMessage> &msg);
@@ -229,6 +220,9 @@
void onPause();
void onResume();
void onSetVideoFrameRate(float fps);
+ int32_t getQueueGeneration(bool audio);
+ int32_t getDrainGeneration(bool audio);
+ bool getSyncQueues();
void onAudioOffloadTearDown(AudioOffloadTearDownReason reason);
status_t onOpenAudioSink(
const sp<AMessage> &format,
@@ -245,7 +239,7 @@
void notifyAudioOffloadTearDown();
void flushQueue(List<QueueEntry> *queue);
- bool dropBufferWhileFlushing(bool audio, const sp<AMessage> &msg);
+ bool dropBufferIfStale(bool audio, const sp<AMessage> &msg);
void syncQueuesDone_l();
bool offloadingAudio() const { return (mFlags & FLAG_OFFLOAD_AUDIO) != 0; }
@@ -253,6 +247,8 @@
void startAudioOffloadPauseTimeout();
void cancelAudioOffloadPauseTimeout();
+ int64_t getDurationUsIfPlayedAtSampleRate(uint32_t numFrames);
+
DISALLOW_EVIL_CONSTRUCTORS(Renderer);
};
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.cpp
index 885ebe4..f53afbd 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.cpp
@@ -29,9 +29,9 @@
NuPlayer::NuPlayerStreamListener::NuPlayerStreamListener(
const sp<IStreamSource> &source,
- ALooper::handler_id id)
+ const sp<AHandler> &targetHandler)
: mSource(source),
- mTargetID(id),
+ mTargetHandler(targetHandler),
mEOS(false),
mSendDataNotification(true) {
mSource->setListener(this);
@@ -65,8 +65,8 @@
if (mSendDataNotification) {
mSendDataNotification = false;
- if (mTargetID != 0) {
- (new AMessage(kWhatMoreDataQueued, mTargetID))->post();
+ if (mTargetHandler != NULL) {
+ (new AMessage(kWhatMoreDataQueued, mTargetHandler))->post();
}
}
}
@@ -86,8 +86,8 @@
if (mSendDataNotification) {
mSendDataNotification = false;
- if (mTargetID != 0) {
- (new AMessage(kWhatMoreDataQueued, mTargetID))->post();
+ if (mTargetHandler != NULL) {
+ (new AMessage(kWhatMoreDataQueued, mTargetHandler))->post();
}
}
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.h b/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.h
index 1874d80..2de829b 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerStreamListener.h
@@ -29,7 +29,7 @@
struct NuPlayer::NuPlayerStreamListener : public BnStreamListener {
NuPlayerStreamListener(
const sp<IStreamSource> &source,
- ALooper::handler_id targetID);
+ const sp<AHandler> &targetHandler);
virtual void queueBuffer(size_t index, size_t size);
@@ -59,7 +59,7 @@
Mutex mLock;
sp<IStreamSource> mSource;
- ALooper::handler_id mTargetID;
+ sp<AHandler> mTargetHandler;
sp<MemoryDealer> mMemoryDealer;
Vector<sp<IMemory> > mBuffers;
List<QueueEntry> mQueue;
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
index 0282a9f..5210fc8 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
@@ -87,7 +87,7 @@
CHECK(mHandler == NULL);
CHECK(mSDPLoader == NULL);
- sp<AMessage> notify = new AMessage(kWhatNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatNotify, this);
CHECK_EQ(mState, (int)DISCONNECTED);
mState = CONNECTING;
@@ -116,7 +116,7 @@
if (mLooper == NULL) {
return;
}
- sp<AMessage> msg = new AMessage(kWhatDisconnect, id());
+ sp<AMessage> msg = new AMessage(kWhatDisconnect, this);
sp<AMessage> dummy;
msg->postAndAwaitResponse(&dummy);
@@ -292,7 +292,7 @@
}
status_t NuPlayer::RTSPSource::seekTo(int64_t seekTimeUs) {
- sp<AMessage> msg = new AMessage(kWhatPerformSeek, id());
+ sp<AMessage> msg = new AMessage(kWhatPerformSeek, this);
msg->setInt32("generation", ++mSeekGeneration);
msg->setInt64("timeUs", seekTimeUs);
msg->post(200000ll);
@@ -311,7 +311,7 @@
void NuPlayer::RTSPSource::onMessageReceived(const sp<AMessage> &msg) {
if (msg->what() == kWhatDisconnect) {
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
mDisconnectReplyID = replyID;
@@ -600,7 +600,7 @@
ALOGE("Unable to find url in SDP");
err = UNKNOWN_ERROR;
} else {
- sp<AMessage> notify = new AMessage(kWhatNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatNotify, this);
mHandler = new MyHandler(rtspUri.c_str(), notify, mUIDValid, mUID);
mLooper->registerHandler(mHandler);
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.h b/media/libmediaplayerservice/nuplayer/RTSPSource.h
index ac3299a..5f2cf33 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.h
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.h
@@ -25,6 +25,7 @@
namespace android {
struct ALooper;
+struct AReplyToken;
struct AnotherPacketSource;
struct MyHandler;
struct SDPLoader;
@@ -96,7 +97,7 @@
bool mIsSDP;
State mState;
status_t mFinalResult;
- uint32_t mDisconnectReplyID;
+ sp<AReplyToken> mDisconnectReplyID;
Mutex mBufferingLock;
bool mBuffering;
diff --git a/media/libmediaplayerservice/nuplayer/StreamingSource.cpp b/media/libmediaplayerservice/nuplayer/StreamingSource.cpp
index b3f224d..0246b59 100644
--- a/media/libmediaplayerservice/nuplayer/StreamingSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/StreamingSource.cpp
@@ -63,7 +63,7 @@
}
void NuPlayer::StreamingSource::start() {
- mStreamListener = new NuPlayerStreamListener(mSource, 0);
+ mStreamListener = new NuPlayerStreamListener(mSource, NULL);
uint32_t sourceFlags = mSource->flags();
@@ -163,7 +163,7 @@
mBuffering = true;
}
- (new AMessage(kWhatReadBuffer, id()))->post();
+ (new AMessage(kWhatReadBuffer, this))->post();
return OK;
}
diff --git a/media/libmediaplayerservice/tests/Android.mk b/media/libmediaplayerservice/tests/Android.mk
new file mode 100644
index 0000000..7bc78ff
--- /dev/null
+++ b/media/libmediaplayerservice/tests/Android.mk
@@ -0,0 +1,24 @@
+# Build the unit tests.
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := DrmSessionManager_test
+
+LOCAL_MODULE_TAGS := tests
+
+LOCAL_SRC_FILES := \
+ DrmSessionManager_test.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+ liblog \
+ libmediaplayerservice \
+ libutils \
+
+LOCAL_C_INCLUDES := \
+ frameworks/av/include \
+ frameworks/av/media/libmediaplayerservice \
+
+LOCAL_32_BIT_ONLY := true
+
+include $(BUILD_NATIVE_TEST)
+
diff --git a/media/libmediaplayerservice/tests/DrmSessionManager_test.cpp b/media/libmediaplayerservice/tests/DrmSessionManager_test.cpp
new file mode 100644
index 0000000..d3e760b
--- /dev/null
+++ b/media/libmediaplayerservice/tests/DrmSessionManager_test.cpp
@@ -0,0 +1,249 @@
+/*
+ * Copyright (C) 2015 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 "DrmSessionManager_test"
+#include <utils/Log.h>
+
+#include <gtest/gtest.h>
+
+#include "Drm.h"
+#include "DrmSessionClientInterface.h"
+#include "DrmSessionManager.h"
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/ProcessInfoInterface.h>
+
+namespace android {
+
+struct FakeProcessInfo : public ProcessInfoInterface {
+ FakeProcessInfo() {}
+ virtual ~FakeProcessInfo() {}
+
+ virtual bool getPriority(int pid, int* priority) {
+ // For testing, use pid as priority.
+ // Lower the value higher the priority.
+ *priority = pid;
+ return true;
+ }
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(FakeProcessInfo);
+};
+
+struct FakeDrm : public DrmSessionClientInterface {
+ FakeDrm() {}
+ virtual ~FakeDrm() {}
+
+ virtual bool reclaimSession(const Vector<uint8_t>& sessionId) {
+ mReclaimedSessions.push_back(sessionId);
+ return true;
+ }
+
+ const Vector<Vector<uint8_t> >& reclaimedSessions() const {
+ return mReclaimedSessions;
+ }
+
+private:
+ Vector<Vector<uint8_t> > mReclaimedSessions;
+
+ DISALLOW_EVIL_CONSTRUCTORS(FakeDrm);
+};
+
+static const int kTestPid1 = 30;
+static const int kTestPid2 = 20;
+static const uint8_t kTestSessionId1[] = {1, 2, 3};
+static const uint8_t kTestSessionId2[] = {4, 5, 6, 7, 8};
+static const uint8_t kTestSessionId3[] = {9, 0};
+
+class DrmSessionManagerTest : public ::testing::Test {
+public:
+ DrmSessionManagerTest()
+ : mDrmSessionManager(new DrmSessionManager(new FakeProcessInfo())),
+ mTestDrm1(new FakeDrm()),
+ mTestDrm2(new FakeDrm()) {
+ GetSessionId(kTestSessionId1, ARRAY_SIZE(kTestSessionId1), &mSessionId1);
+ GetSessionId(kTestSessionId2, ARRAY_SIZE(kTestSessionId2), &mSessionId2);
+ GetSessionId(kTestSessionId3, ARRAY_SIZE(kTestSessionId3), &mSessionId3);
+ }
+
+protected:
+ static void GetSessionId(const uint8_t* ids, size_t num, Vector<uint8_t>* sessionId) {
+ for (size_t i = 0; i < num; ++i) {
+ sessionId->push_back(ids[i]);
+ }
+ }
+
+ static void ExpectEqSessionInfo(const SessionInfo& info, sp<DrmSessionClientInterface> drm,
+ const Vector<uint8_t>& sessionId, int64_t timeStamp) {
+ EXPECT_EQ(drm, info.drm);
+ EXPECT_TRUE(isEqualSessionId(sessionId, info.sessionId));
+ EXPECT_EQ(timeStamp, info.timeStamp);
+ }
+
+ void addSession() {
+ mDrmSessionManager->addSession(kTestPid1, mTestDrm1, mSessionId1);
+ mDrmSessionManager->addSession(kTestPid2, mTestDrm2, mSessionId2);
+ mDrmSessionManager->addSession(kTestPid2, mTestDrm2, mSessionId3);
+ const PidSessionInfosMap& map = sessionMap();
+ EXPECT_EQ(2, map.size());
+ ssize_t index1 = map.indexOfKey(kTestPid1);
+ ASSERT_GE(index1, 0);
+ const SessionInfos& infos1 = map[index1];
+ EXPECT_EQ(1, infos1.size());
+ ExpectEqSessionInfo(infos1[0], mTestDrm1, mSessionId1, 0);
+
+ ssize_t index2 = map.indexOfKey(kTestPid2);
+ ASSERT_GE(index2, 0);
+ const SessionInfos& infos2 = map[index2];
+ EXPECT_EQ(2, infos2.size());
+ ExpectEqSessionInfo(infos2[0], mTestDrm2, mSessionId2, 1);
+ ExpectEqSessionInfo(infos2[1], mTestDrm2, mSessionId3, 2);
+ }
+
+ const PidSessionInfosMap& sessionMap() {
+ return mDrmSessionManager->mSessionMap;
+ }
+
+ void testGetLowestPriority() {
+ int pid;
+ int priority;
+ EXPECT_FALSE(mDrmSessionManager->getLowestPriority_l(&pid, &priority));
+
+ addSession();
+ EXPECT_TRUE(mDrmSessionManager->getLowestPriority_l(&pid, &priority));
+
+ EXPECT_EQ(kTestPid1, pid);
+ FakeProcessInfo processInfo;
+ int priority1;
+ processInfo.getPriority(kTestPid1, &priority1);
+ EXPECT_EQ(priority1, priority);
+ }
+
+ void testGetLeastUsedSession() {
+ sp<DrmSessionClientInterface> drm;
+ Vector<uint8_t> sessionId;
+ EXPECT_FALSE(mDrmSessionManager->getLeastUsedSession_l(kTestPid1, &drm, &sessionId));
+
+ addSession();
+
+ EXPECT_TRUE(mDrmSessionManager->getLeastUsedSession_l(kTestPid1, &drm, &sessionId));
+ EXPECT_EQ(mTestDrm1, drm);
+ EXPECT_TRUE(isEqualSessionId(mSessionId1, sessionId));
+
+ EXPECT_TRUE(mDrmSessionManager->getLeastUsedSession_l(kTestPid2, &drm, &sessionId));
+ EXPECT_EQ(mTestDrm2, drm);
+ EXPECT_TRUE(isEqualSessionId(mSessionId2, sessionId));
+
+ // mSessionId2 is no longer the least used session.
+ mDrmSessionManager->useSession(mSessionId2);
+ EXPECT_TRUE(mDrmSessionManager->getLeastUsedSession_l(kTestPid2, &drm, &sessionId));
+ EXPECT_EQ(mTestDrm2, drm);
+ EXPECT_TRUE(isEqualSessionId(mSessionId3, sessionId));
+ }
+
+ sp<DrmSessionManager> mDrmSessionManager;
+ sp<FakeDrm> mTestDrm1;
+ sp<FakeDrm> mTestDrm2;
+ Vector<uint8_t> mSessionId1;
+ Vector<uint8_t> mSessionId2;
+ Vector<uint8_t> mSessionId3;
+};
+
+TEST_F(DrmSessionManagerTest, addSession) {
+ addSession();
+}
+
+TEST_F(DrmSessionManagerTest, useSession) {
+ addSession();
+
+ mDrmSessionManager->useSession(mSessionId1);
+ mDrmSessionManager->useSession(mSessionId3);
+
+ const PidSessionInfosMap& map = sessionMap();
+ const SessionInfos& infos1 = map.valueFor(kTestPid1);
+ const SessionInfos& infos2 = map.valueFor(kTestPid2);
+ ExpectEqSessionInfo(infos1[0], mTestDrm1, mSessionId1, 3);
+ ExpectEqSessionInfo(infos2[1], mTestDrm2, mSessionId3, 4);
+}
+
+TEST_F(DrmSessionManagerTest, removeSession) {
+ addSession();
+
+ mDrmSessionManager->removeSession(mSessionId2);
+
+ const PidSessionInfosMap& map = sessionMap();
+ EXPECT_EQ(2, map.size());
+ const SessionInfos& infos1 = map.valueFor(kTestPid1);
+ const SessionInfos& infos2 = map.valueFor(kTestPid2);
+ EXPECT_EQ(1, infos1.size());
+ EXPECT_EQ(1, infos2.size());
+ // mSessionId2 has been removed.
+ ExpectEqSessionInfo(infos2[0], mTestDrm2, mSessionId3, 2);
+}
+
+TEST_F(DrmSessionManagerTest, removeDrm) {
+ addSession();
+
+ sp<FakeDrm> drm = new FakeDrm;
+ const uint8_t ids[] = {123};
+ Vector<uint8_t> sessionId;
+ GetSessionId(ids, ARRAY_SIZE(ids), &sessionId);
+ mDrmSessionManager->addSession(kTestPid2, drm, sessionId);
+
+ mDrmSessionManager->removeDrm(mTestDrm2);
+
+ const PidSessionInfosMap& map = sessionMap();
+ const SessionInfos& infos2 = map.valueFor(kTestPid2);
+ EXPECT_EQ(1, infos2.size());
+ // mTestDrm2 has been removed.
+ ExpectEqSessionInfo(infos2[0], drm, sessionId, 3);
+}
+
+TEST_F(DrmSessionManagerTest, reclaimSession) {
+ EXPECT_FALSE(mDrmSessionManager->reclaimSession(kTestPid1));
+ addSession();
+
+ // calling pid priority is too low
+ EXPECT_FALSE(mDrmSessionManager->reclaimSession(50));
+
+ EXPECT_TRUE(mDrmSessionManager->reclaimSession(10));
+ EXPECT_EQ(1, mTestDrm1->reclaimedSessions().size());
+ EXPECT_TRUE(isEqualSessionId(mSessionId1, mTestDrm1->reclaimedSessions()[0]));
+
+ mDrmSessionManager->removeSession(mSessionId1);
+
+ // add a session from a higher priority process.
+ sp<FakeDrm> drm = new FakeDrm;
+ const uint8_t ids[] = {1, 3, 5};
+ Vector<uint8_t> sessionId;
+ GetSessionId(ids, ARRAY_SIZE(ids), &sessionId);
+ mDrmSessionManager->addSession(15, drm, sessionId);
+
+ EXPECT_TRUE(mDrmSessionManager->reclaimSession(18));
+ EXPECT_EQ(1, mTestDrm2->reclaimedSessions().size());
+ // mSessionId2 is reclaimed.
+ EXPECT_TRUE(isEqualSessionId(mSessionId2, mTestDrm2->reclaimedSessions()[0]));
+}
+
+TEST_F(DrmSessionManagerTest, getLowestPriority) {
+ testGetLowestPriority();
+}
+
+TEST_F(DrmSessionManagerTest, getLeastUsedSession_l) {
+ testGetLeastUsedSession();
+}
+
+} // namespace android
diff --git a/media/libnbaio/Android.mk b/media/libnbaio/Android.mk
index 9707c4a..1353f28 100644
--- a/media/libnbaio/Android.mk
+++ b/media/libnbaio/Android.mk
@@ -11,7 +11,6 @@
MonoPipeReader.cpp \
Pipe.cpp \
PipeReader.cpp \
- roundup.c \
SourceAudioBufferProvider.cpp
LOCAL_SRC_FILES += NBLog.cpp
@@ -27,12 +26,13 @@
LOCAL_MODULE := libnbaio
LOCAL_SHARED_LIBRARIES := \
+ libaudioutils \
libbinder \
libcommon_time_client \
libcutils \
libutils \
liblog
-LOCAL_STATIC_LIBRARIES += libinstantssq
+LOCAL_C_INCLUDES := $(call include-path-for, audio-utils)
include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp
index 0b65861..129e9ef 100644
--- a/media/libnbaio/MonoPipe.cpp
+++ b/media/libnbaio/MonoPipe.cpp
@@ -27,7 +27,7 @@
#include <utils/Trace.h>
#include <media/AudioBufferProvider.h>
#include <media/nbaio/MonoPipe.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
namespace android {
diff --git a/media/libnbaio/MonoPipeReader.cpp b/media/libnbaio/MonoPipeReader.cpp
index de82229..e4d3ed8 100644
--- a/media/libnbaio/MonoPipeReader.cpp
+++ b/media/libnbaio/MonoPipeReader.cpp
@@ -39,7 +39,7 @@
return NEGOTIATE;
}
ssize_t ret = android_atomic_acquire_load(&mPipe->mRear) - mPipe->mFront;
- ALOG_ASSERT((0 <= ret) && (ret <= mMaxFrames));
+ ALOG_ASSERT((0 <= ret) && ((size_t) ret <= mPipe->mMaxFrames));
return ret;
}
diff --git a/media/libnbaio/Pipe.cpp b/media/libnbaio/Pipe.cpp
index 6e0ec8c..13f211d 100644
--- a/media/libnbaio/Pipe.cpp
+++ b/media/libnbaio/Pipe.cpp
@@ -21,7 +21,7 @@
#include <cutils/compiler.h>
#include <utils/Log.h>
#include <media/nbaio/Pipe.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
namespace android {
diff --git a/media/libnbaio/roundup.c b/media/libnbaio/roundup.c
deleted file mode 100644
index 1d552d1..0000000
--- a/media/libnbaio/roundup.c
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (C) 2012 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/nbaio/roundup.h>
-
-unsigned roundup(unsigned v)
-{
- // __builtin_clz is undefined for zero input
- if (v == 0) {
- v = 1;
- }
- int lz = __builtin_clz((int) v);
- unsigned rounded = ((unsigned) 0x80000000) >> lz;
- // 0x800000001 and higher are actually rounded _down_ to prevent overflow
- if (v > rounded && lz > 0) {
- rounded <<= 1;
- }
- return rounded;
-}
diff --git a/media/libstagefright/AACWriter.cpp b/media/libstagefright/AACWriter.cpp
index 2e41d80..9d90dbd 100644
--- a/media/libstagefright/AACWriter.cpp
+++ b/media/libstagefright/AACWriter.cpp
@@ -36,33 +36,19 @@
namespace android {
-AACWriter::AACWriter(const char *filename)
- : mFd(-1),
- mInitCheck(NO_INIT),
- mStarted(false),
- mPaused(false),
- mResumed(false),
- mChannelCount(-1),
- mSampleRate(-1),
- mAACProfile(OMX_AUDIO_AACObjectLC) {
-
- ALOGV("AACWriter Constructor");
-
- mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
- if (mFd >= 0) {
- mInitCheck = OK;
- }
-}
-
AACWriter::AACWriter(int fd)
: mFd(dup(fd)),
mInitCheck(mFd < 0? NO_INIT: OK),
mStarted(false),
mPaused(false),
mResumed(false),
+ mThread(0),
+ mEstimatedSizeBytes(0),
+ mEstimatedDurationUs(0),
mChannelCount(-1),
mSampleRate(-1),
- mAACProfile(OMX_AUDIO_AACObjectLC) {
+ mAACProfile(OMX_AUDIO_AACObjectLC),
+ mFrameDurationUs(0) {
}
AACWriter::~AACWriter() {
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index d298cb1..97f3e20 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -419,6 +419,7 @@
mMetaDataBuffersToSubmit(0),
mRepeatFrameDelayUs(-1ll),
mMaxPtsGapUs(-1ll),
+ mMaxFps(-1),
mTimePerFrameUs(-1ll),
mTimePerCaptureUs(-1ll),
mCreateInputBuffersSuspended(false),
@@ -451,61 +452,61 @@
void ACodec::initiateSetup(const sp<AMessage> &msg) {
msg->setWhat(kWhatSetup);
- msg->setTarget(id());
+ msg->setTarget(this);
msg->post();
}
void ACodec::signalSetParameters(const sp<AMessage> ¶ms) {
- sp<AMessage> msg = new AMessage(kWhatSetParameters, id());
+ sp<AMessage> msg = new AMessage(kWhatSetParameters, this);
msg->setMessage("params", params);
msg->post();
}
void ACodec::initiateAllocateComponent(const sp<AMessage> &msg) {
msg->setWhat(kWhatAllocateComponent);
- msg->setTarget(id());
+ msg->setTarget(this);
msg->post();
}
void ACodec::initiateConfigureComponent(const sp<AMessage> &msg) {
msg->setWhat(kWhatConfigureComponent);
- msg->setTarget(id());
+ msg->setTarget(this);
msg->post();
}
void ACodec::initiateCreateInputSurface() {
- (new AMessage(kWhatCreateInputSurface, id()))->post();
+ (new AMessage(kWhatCreateInputSurface, this))->post();
}
void ACodec::signalEndOfInputStream() {
- (new AMessage(kWhatSignalEndOfInputStream, id()))->post();
+ (new AMessage(kWhatSignalEndOfInputStream, this))->post();
}
void ACodec::initiateStart() {
- (new AMessage(kWhatStart, id()))->post();
+ (new AMessage(kWhatStart, this))->post();
}
void ACodec::signalFlush() {
ALOGV("[%s] signalFlush", mComponentName.c_str());
- (new AMessage(kWhatFlush, id()))->post();
+ (new AMessage(kWhatFlush, this))->post();
}
void ACodec::signalResume() {
- (new AMessage(kWhatResume, id()))->post();
+ (new AMessage(kWhatResume, this))->post();
}
void ACodec::initiateShutdown(bool keepComponentAllocated) {
- sp<AMessage> msg = new AMessage(kWhatShutdown, id());
+ sp<AMessage> msg = new AMessage(kWhatShutdown, this);
msg->setInt32("keepComponentAllocated", keepComponentAllocated);
msg->post();
if (!keepComponentAllocated) {
// ensure shutdown completes in 3 seconds
- (new AMessage(kWhatReleaseCodecInstance, id()))->post(3000000);
+ (new AMessage(kWhatReleaseCodecInstance, this))->post(3000000);
}
}
void ACodec::signalRequestIDRFrame() {
- (new AMessage(kWhatRequestIDRFrame, id()))->post();
+ (new AMessage(kWhatRequestIDRFrame, this))->post();
}
// *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
@@ -516,7 +517,7 @@
void ACodec::signalSubmitOutputMetaDataBufferIfEOS_workaround() {
if (mPortEOS[kPortIndexInput] && !mPortEOS[kPortIndexOutput] &&
mMetaDataBuffersToSubmit > 0) {
- (new AMessage(kWhatSubmitOutputMetaDataBufferIfEOS, id()))->post();
+ (new AMessage(kWhatSubmitOutputMetaDataBufferIfEOS, this))->post();
}
}
@@ -1259,6 +1260,10 @@
mMaxPtsGapUs = -1ll;
}
+ if (!msg->findFloat("max-fps-to-encoder", &mMaxFps)) {
+ mMaxFps = -1;
+ }
+
if (!msg->findInt64("time-lapse", &mTimePerCaptureUs)) {
mTimePerCaptureUs = -1ll;
}
@@ -1675,6 +1680,11 @@
err = setMinBufferSize(kPortIndexInput, 8192); // XXX
}
+ int32_t priority;
+ if (msg->findInt32("priority", &priority)) {
+ err = setPriority(priority);
+ }
+
mBaseOutputFormat = outputFormat;
CHECK_EQ(getPortFormat(kPortIndexInput, inputFormat), (status_t)OK);
@@ -1685,6 +1695,22 @@
return err;
}
+status_t ACodec::setPriority(int32_t priority) {
+ if (priority < 0) {
+ return BAD_VALUE;
+ }
+ OMX_PARAM_U32TYPE config;
+ InitOMXParams(&config);
+ config.nU32 = (OMX_U32)priority;
+ status_t temp = mOMX->setConfig(
+ mNode, (OMX_INDEXTYPE)OMX_IndexConfigPriority,
+ &config, sizeof(config));
+ if (temp != OK) {
+ ALOGI("codec does not support config priority (err %d)", temp);
+ }
+ return OK;
+}
+
status_t ACodec::setMinBufferSize(OMX_U32 portIndex, size_t size) {
OMX_PARAM_PORTDEFINITIONTYPE def;
InitOMXParams(&def);
@@ -4297,7 +4323,7 @@
info->mData->meta()->clear();
notify->setBuffer("buffer", info->mData);
- sp<AMessage> reply = new AMessage(kWhatInputBufferFilled, mCodec->id());
+ sp<AMessage> reply = new AMessage(kWhatInputBufferFilled, mCodec);
reply->setInt32("buffer-id", info->mBufferID);
notify->setMessage("reply", reply);
@@ -4557,7 +4583,7 @@
}
sp<AMessage> reply =
- new AMessage(kWhatOutputBufferDrained, mCodec->id());
+ new AMessage(kWhatOutputBufferDrained, mCodec);
if (!mCodec->mSentFormat && rangeLength > 0) {
mCodec->sendFormatChange(reply);
@@ -4833,7 +4859,7 @@
sp<IOMX> omx = client.interface();
- sp<AMessage> notify = new AMessage(kWhatOMXDied, mCodec->id());
+ sp<AMessage> notify = new AMessage(kWhatOMXDied, mCodec);
mDeathNotifier = new DeathNotifier(notify);
if (IInterface::asBinder(omx)->linkToDeath(mDeathNotifier) != OK) {
@@ -4908,7 +4934,7 @@
return false;
}
- notify = new AMessage(kWhatOMXMessage, mCodec->id());
+ notify = new AMessage(kWhatOMXMessage, mCodec);
observer->setNotificationMessage(notify);
mCodec->mComponentName = componentName;
@@ -5114,6 +5140,21 @@
}
}
+ if (err == OK && mCodec->mMaxFps > 0) {
+ err = mCodec->mOMX->setInternalOption(
+ mCodec->mNode,
+ kPortIndexInput,
+ IOMX::INTERNAL_OPTION_MAX_FPS,
+ &mCodec->mMaxFps,
+ sizeof(mCodec->mMaxFps));
+
+ if (err != OK) {
+ ALOGE("[%s] Unable to configure max fps (err %d)",
+ mCodec->mComponentName.c_str(),
+ err);
+ }
+ }
+
if (err == OK && mCodec->mTimePerCaptureUs > 0ll
&& mCodec->mTimePerFrameUs > 0ll) {
int64_t timeLapse[2];
@@ -5984,7 +6025,7 @@
case OMX_EventPortSettingsChanged:
{
- sp<AMessage> msg = new AMessage(kWhatOMXMessage, mCodec->id());
+ sp<AMessage> msg = new AMessage(kWhatOMXMessage, mCodec);
msg->setInt32("type", omx_message::EVENT);
msg->setInt32("node", mCodec->mNode);
msg->setInt32("event", event);
diff --git a/media/libstagefright/AMRWriter.cpp b/media/libstagefright/AMRWriter.cpp
index 9aa7d95..f53d7f0 100644
--- a/media/libstagefright/AMRWriter.cpp
+++ b/media/libstagefright/AMRWriter.cpp
@@ -31,19 +31,6 @@
namespace android {
-AMRWriter::AMRWriter(const char *filename)
- : mFd(-1),
- mInitCheck(NO_INIT),
- mStarted(false),
- mPaused(false),
- mResumed(false) {
-
- mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
- if (mFd >= 0) {
- mInitCheck = OK;
- }
-}
-
AMRWriter::AMRWriter(int fd)
: mFd(dup(fd)),
mInitCheck(mFd < 0? NO_INIT: OK),
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 2629afc..a2cbdaf 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -31,11 +31,13 @@
MediaAdapter.cpp \
MediaBuffer.cpp \
MediaBufferGroup.cpp \
+ MediaClock.cpp \
MediaCodec.cpp \
MediaCodecList.cpp \
MediaCodecSource.cpp \
MediaDefs.cpp \
MediaExtractor.cpp \
+ MediaSync.cpp \
MidiExtractor.cpp \
http/MediaHTTP.cpp \
MediaMuxer.cpp \
@@ -46,6 +48,7 @@
OMXClient.cpp \
OMXCodec.cpp \
OggExtractor.cpp \
+ ProcessInfo.cpp \
SampleIterator.cpp \
SampleTable.cpp \
SkipCutBuffer.cpp \
@@ -101,6 +104,7 @@
libstagefright_color_conversion \
libstagefright_aacenc \
libstagefright_matroska \
+ libstagefright_mediafilter \
libstagefright_webm \
libstagefright_timedtext \
libvpx \
@@ -108,13 +112,14 @@
libstagefright_mpeg2ts \
libstagefright_id3 \
libFLAC \
- libmedia_helper
+ libmedia_helper \
LOCAL_SHARED_LIBRARIES += \
libstagefright_enc_common \
libstagefright_avc_common \
libstagefright_foundation \
- libdl
+ libdl \
+ libRScpp \
LOCAL_CFLAGS += -Wno-multichar
diff --git a/media/libstagefright/FileSource.cpp b/media/libstagefright/FileSource.cpp
index a7ca3da..f0db76b 100644
--- a/media/libstagefright/FileSource.cpp
+++ b/media/libstagefright/FileSource.cpp
@@ -14,6 +14,10 @@
* limitations under the License.
*/
+//#define LOG_NDEBUG 0
+#define LOG_TAG "FileSource"
+#include <utils/Log.h>
+
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/FileSource.h>
#include <sys/types.h>
diff --git a/media/libstagefright/HTTPBase.cpp b/media/libstagefright/HTTPBase.cpp
index 0c2ff15..77a652a 100644
--- a/media/libstagefright/HTTPBase.cpp
+++ b/media/libstagefright/HTTPBase.cpp
@@ -75,7 +75,11 @@
bool HTTPBase::estimateBandwidth(int32_t *bandwidth_bps) {
Mutex::Autolock autoLock(mLock);
- if (mNumBandwidthHistoryItems < 2) {
+ // Do not do bandwidth estimation if we don't have enough samples, or
+ // total bytes download are too small (<64K).
+ // Bandwidth estimation from these samples can often shoot up and cause
+ // unwanted bw adaption behaviors.
+ if (mNumBandwidthHistoryItems < 2 || mTotalTransferBytes < 65536) {
return false;
}
diff --git a/media/libstagefright/MPEG2TSWriter.cpp b/media/libstagefright/MPEG2TSWriter.cpp
index 9856f92..ef07aa0 100644
--- a/media/libstagefright/MPEG2TSWriter.cpp
+++ b/media/libstagefright/MPEG2TSWriter.cpp
@@ -135,7 +135,7 @@
mNotify = notify;
- (new AMessage(kWhatStart, id()))->post();
+ (new AMessage(kWhatStart, this))->post();
}
void MPEG2TSWriter::SourceInfo::stop() {
@@ -361,7 +361,7 @@
}
void MPEG2TSWriter::SourceInfo::readMore() {
- (new AMessage(kWhatRead, id()))->post();
+ (new AMessage(kWhatRead, this))->post();
}
void MPEG2TSWriter::SourceInfo::onMessageReceived(const sp<AMessage> &msg) {
@@ -480,19 +480,6 @@
init();
}
-MPEG2TSWriter::MPEG2TSWriter(const char *filename)
- : mFile(fopen(filename, "wb")),
- mWriteCookie(NULL),
- mWriteFunc(NULL),
- mStarted(false),
- mNumSourcesDone(0),
- mNumTSPacketsWritten(0),
- mNumTSPacketsBeforeMeta(0),
- mPATContinuityCounter(0),
- mPMTContinuityCounter(0) {
- init();
-}
-
MPEG2TSWriter::MPEG2TSWriter(
void *cookie,
ssize_t (*write)(void *cookie, const void *data, size_t size))
@@ -565,7 +552,7 @@
for (size_t i = 0; i < mSources.size(); ++i) {
sp<AMessage> notify =
- new AMessage(kWhatSourceNotify, mReflector->id());
+ new AMessage(kWhatSourceNotify, mReflector);
notify->setInt32("source-index", i);
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 8bf7f63..d0f42cc 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -354,6 +354,8 @@
MPEG4Extractor::MPEG4Extractor(const sp<DataSource> &source)
: mMoofOffset(0),
+ mMoofFound(false),
+ mMdatFound(false),
mDataSource(source),
mInitCheck(NO_INIT),
mHasVideo(false),
@@ -490,7 +492,9 @@
off64_t offset = 0;
status_t err;
- while (true) {
+ bool sawMoovOrSidx = false;
+
+ while (!(sawMoovOrSidx && (mMdatFound || mMoofFound))) {
off64_t orig_offset = offset;
err = parseChunk(&offset, 0);
@@ -502,23 +506,9 @@
ALOGE("did not advance: 0x%lld->0x%lld", orig_offset, offset);
err = ERROR_MALFORMED;
break;
- } else if (err == OK) {
- continue;
+ } else if (err == UNKNOWN_ERROR) {
+ sawMoovOrSidx = true;
}
-
- uint32_t hdr[2];
- if (mDataSource->readAt(offset, hdr, 8) < 8) {
- break;
- }
- uint32_t chunk_type = ntohl(hdr[1]);
- if (chunk_type == FOURCC('m', 'o', 'o', 'f')) {
- // store the offset of the first segment
- mMoofOffset = offset;
- } else if (chunk_type != FOURCC('m', 'd', 'a', 't')) {
- // keep parsing until we get to the data
- continue;
- }
- break;
}
if (mInitCheck == OK) {
@@ -864,6 +854,12 @@
case FOURCC('s', 'c', 'h', 'i'):
case FOURCC('e', 'd', 't', 's'):
{
+ 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);
@@ -1830,6 +1826,9 @@
case FOURCC('m', 'd', 'a', 't'):
{
ALOGV("mdat chunk, drm: %d", mIsDrm);
+
+ mMdatFound = true;
+
if (!mIsDrm) {
*offset += chunk_size;
break;
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 9f20b1d..6f6e362 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -29,6 +29,7 @@
#include <utils/Log.h>
#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/MPEG4Writer.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MetaData.h>
@@ -62,6 +63,14 @@
static const uint8_t kNalUnitTypePicParamSet = 0x08;
static const int64_t kInitialDelayTimeUs = 700000LL;
+static const char kMetaKey_Model[] = "com.android.model";
+static const char kMetaKey_Version[] = "com.android.version";
+static const char kMetaKey_Build[] = "com.android.build";
+static const char kMetaKey_CaptureFps[] = "com.android.capture.fps";
+
+/* uncomment to include model and build in meta */
+//#define SHOW_MODEL_BUILD 1
+
class MPEG4Writer::Track {
public:
Track(MPEG4Writer *owner, const sp<MediaSource> &source, size_t trackId);
@@ -345,31 +354,6 @@
Track &operator=(const Track &);
};
-MPEG4Writer::MPEG4Writer(const char *filename)
- : mFd(-1),
- mInitCheck(NO_INIT),
- mIsRealTimeRecording(true),
- mUse4ByteNalLength(true),
- mUse32BitOffset(true),
- mIsFileSizeLimitExplicitlyRequested(false),
- mPaused(false),
- mStarted(false),
- mWriterThreadStarted(false),
- mOffset(0),
- mMdatOffset(0),
- mEstimatedMoovBoxSize(0),
- mInterleaveDurationUs(1000000),
- mLatitudex10000(0),
- mLongitudex10000(0),
- mAreGeoTagsAvailable(false),
- mStartTimeOffsetMs(-1) {
-
- mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
- if (mFd >= 0) {
- mInitCheck = OK;
- }
-}
-
MPEG4Writer::MPEG4Writer(int fd)
: mFd(dup(fd)),
mInitCheck(mFd < 0? NO_INIT: OK),
@@ -383,11 +367,14 @@
mOffset(0),
mMdatOffset(0),
mEstimatedMoovBoxSize(0),
+ mMoovExtraSize(0),
mInterleaveDurationUs(1000000),
mLatitudex10000(0),
mLongitudex10000(0),
mAreGeoTagsAvailable(false),
+ mMetaKeys(new AMessage()),
mStartTimeOffsetMs(-1) {
+ addDeviceMeta();
}
MPEG4Writer::~MPEG4Writer() {
@@ -507,6 +494,34 @@
return OK;
}
+void MPEG4Writer::addDeviceMeta() {
+ // add device info and estimate space in 'moov'
+ char val[PROPERTY_VALUE_MAX];
+ size_t n;
+ // meta size is estimated by adding up the following:
+ // - meta header structures, which occur only once (total 66 bytes)
+ // - size for each key, which consists of a fixed header (32 bytes),
+ // plus key length and data length.
+ mMoovExtraSize += 66;
+ if (property_get("ro.build.version.release", val, NULL)
+ && (n = strlen(val)) > 0) {
+ mMetaKeys->setString(kMetaKey_Version, val, n + 1);
+ mMoovExtraSize += sizeof(kMetaKey_Version) + n + 32;
+ }
+#ifdef SHOW_MODEL_BUILD
+ if (property_get("ro.product.model", val, NULL)
+ && (n = strlen(val)) > 0) {
+ mMetaKeys->setString(kMetaKey_Model, val, n + 1);
+ mMoovExtraSize += sizeof(kMetaKey_Model) + n + 32;
+ }
+ if (property_get("ro.build.display.id", val, NULL)
+ && (n = strlen(val)) > 0) {
+ mMetaKeys->setString(kMetaKey_Build, val, n + 1);
+ mMoovExtraSize += sizeof(kMetaKey_Build) + n + 32;
+ }
+#endif
+}
+
int64_t MPEG4Writer::estimateMoovBoxSize(int32_t bitRate) {
// This implementation is highly experimental/heurisitic.
//
@@ -560,6 +575,9 @@
size = MAX_MOOV_BOX_SIZE;
}
+ // Account for the extra stuff (Geo, meta keys, etc.)
+ size += mMoovExtraSize;
+
ALOGI("limits: %" PRId64 "/%" PRId64 " bytes/us, bit rate: %d bps and the"
" estimated moov size %" PRId64 " bytes",
mMaxFileSizeLimitBytes, mMaxFileDurationLimitUs, bitRate, size);
@@ -971,6 +989,7 @@
if (mAreGeoTagsAvailable) {
writeUdtaBox();
}
+ writeMetaBox();
int32_t id = 1;
for (List<Track *>::iterator it = mTracks.begin();
it != mTracks.end(); ++it, ++id) {
@@ -1140,6 +1159,14 @@
return bytes;
}
+void MPEG4Writer::beginBox(uint32_t id) {
+ mBoxes.push_back(mWriteMoovBoxToMemory?
+ mMoovBoxBufferOffset: mOffset);
+
+ writeInt32(0);
+ writeInt32(id);
+}
+
void MPEG4Writer::beginBox(const char *fourcc) {
CHECK_EQ(strlen(fourcc), 4);
@@ -1264,6 +1291,18 @@
mLatitudex10000 = latitudex10000;
mLongitudex10000 = longitudex10000;
mAreGeoTagsAvailable = true;
+ mMoovExtraSize += 30;
+ return OK;
+}
+
+status_t MPEG4Writer::setCaptureRate(float captureFps) {
+ if (captureFps <= 0.0f) {
+ return BAD_VALUE;
+ }
+
+ mMetaKeys->setFloat(kMetaKey_CaptureFps, captureFps);
+ mMoovExtraSize += sizeof(kMetaKey_CaptureFps) + 4 + 32;
+
return OK;
}
@@ -3095,6 +3134,103 @@
endBox();
}
+void MPEG4Writer::writeHdlr() {
+ beginBox("hdlr");
+ writeInt32(0); // Version, Flags
+ writeInt32(0); // Predefined
+ writeFourcc("mdta");
+ writeInt32(0); // Reserved[0]
+ writeInt32(0); // Reserved[1]
+ writeInt32(0); // Reserved[2]
+ writeInt8(0); // Name (empty)
+ endBox();
+}
+
+void MPEG4Writer::writeKeys() {
+ size_t count = mMetaKeys->countEntries();
+
+ beginBox("keys");
+ writeInt32(0); // Version, Flags
+ writeInt32(count); // Entry_count
+ for (size_t i = 0; i < count; i++) {
+ AMessage::Type type;
+ const char *key = mMetaKeys->getEntryNameAt(i, &type);
+ size_t n = strlen(key);
+ writeInt32(n + 8);
+ writeFourcc("mdta");
+ write(key, n); // write without the \0
+ }
+ endBox();
+}
+
+void MPEG4Writer::writeIlst() {
+ size_t count = mMetaKeys->countEntries();
+
+ beginBox("ilst");
+ for (size_t i = 0; i < count; i++) {
+ beginBox(i + 1); // key id (1-based)
+ beginBox("data");
+ AMessage::Type type;
+ const char *key = mMetaKeys->getEntryNameAt(i, &type);
+ switch (type) {
+ case AMessage::kTypeString:
+ {
+ AString val;
+ CHECK(mMetaKeys->findString(key, &val));
+ writeInt32(1); // type = UTF8
+ writeInt32(0); // default country/language
+ write(val.c_str(), strlen(val.c_str())); // write without \0
+ break;
+ }
+
+ case AMessage::kTypeFloat:
+ {
+ float val;
+ CHECK(mMetaKeys->findFloat(key, &val));
+ writeInt32(23); // type = float32
+ writeInt32(0); // default country/language
+ writeInt32(*reinterpret_cast<int32_t *>(&val));
+ break;
+ }
+
+ case AMessage::kTypeInt32:
+ {
+ int32_t val;
+ CHECK(mMetaKeys->findInt32(key, &val));
+ writeInt32(67); // type = signed int32
+ writeInt32(0); // default country/language
+ writeInt32(val);
+ break;
+ }
+
+ default:
+ {
+ ALOGW("Unsupported key type, writing 0 instead");
+ writeInt32(77); // type = unsigned int32
+ writeInt32(0); // default country/language
+ writeInt32(0);
+ break;
+ }
+ }
+ endBox(); // data
+ endBox(); // key id
+ }
+ endBox(); // ilst
+}
+
+void MPEG4Writer::writeMetaBox() {
+ size_t count = mMetaKeys->countEntries();
+ if (count == 0) {
+ return;
+ }
+
+ beginBox("meta");
+ writeHdlr();
+ writeKeys();
+ writeIlst();
+ endBox();
+}
+
/*
* Geodata is stored according to ISO-6709 standard.
*/
diff --git a/media/libstagefright/MediaClock.cpp b/media/libstagefright/MediaClock.cpp
new file mode 100644
index 0000000..433f555
--- /dev/null
+++ b/media/libstagefright/MediaClock.cpp
@@ -0,0 +1,148 @@
+/*
+ * Copyright (C) 2015 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 "MediaClock"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaClock.h>
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+
+namespace android {
+
+MediaClock::MediaClock()
+ : mAnchorTimeMediaUs(-1),
+ mAnchorTimeRealUs(-1),
+ mMaxTimeMediaUs(INT64_MAX),
+ mStartingTimeMediaUs(-1),
+ mPlaybackRate(1.0) {
+}
+
+MediaClock::~MediaClock() {
+}
+
+void MediaClock::setStartingTimeMedia(int64_t startingTimeMediaUs) {
+ Mutex::Autolock autoLock(mLock);
+ mStartingTimeMediaUs = startingTimeMediaUs;
+}
+
+void MediaClock::clearAnchor() {
+ Mutex::Autolock autoLock(mLock);
+ mAnchorTimeMediaUs = -1;
+ mAnchorTimeRealUs = -1;
+}
+
+void MediaClock::updateAnchor(
+ int64_t anchorTimeMediaUs,
+ int64_t anchorTimeRealUs,
+ int64_t maxTimeMediaUs) {
+ if (anchorTimeMediaUs < 0 || anchorTimeRealUs < 0) {
+ ALOGW("reject anchor time since it is negative.");
+ return;
+ }
+
+ Mutex::Autolock autoLock(mLock);
+ int64_t nowUs = ALooper::GetNowUs();
+ int64_t nowMediaUs =
+ anchorTimeMediaUs + (nowUs - anchorTimeRealUs) * (double)mPlaybackRate;
+ if (nowMediaUs < 0) {
+ ALOGW("reject anchor time since it leads to negative media time.");
+ return;
+ }
+ mAnchorTimeRealUs = nowUs;
+ mAnchorTimeMediaUs = nowMediaUs;
+ mMaxTimeMediaUs = maxTimeMediaUs;
+}
+
+void MediaClock::updateMaxTimeMedia(int64_t maxTimeMediaUs) {
+ Mutex::Autolock autoLock(mLock);
+ mMaxTimeMediaUs = maxTimeMediaUs;
+}
+
+void MediaClock::setPlaybackRate(float rate) {
+ CHECK_GE(rate, 0.0);
+ Mutex::Autolock autoLock(mLock);
+ if (mAnchorTimeRealUs == -1) {
+ mPlaybackRate = rate;
+ return;
+ }
+
+ int64_t nowUs = ALooper::GetNowUs();
+ mAnchorTimeMediaUs += (nowUs - mAnchorTimeRealUs) * (double)mPlaybackRate;
+ if (mAnchorTimeMediaUs < 0) {
+ ALOGW("setRate: anchor time should not be negative, set to 0.");
+ mAnchorTimeMediaUs = 0;
+ }
+ mAnchorTimeRealUs = nowUs;
+ mPlaybackRate = rate;
+}
+
+status_t MediaClock::getMediaTime(
+ int64_t realUs, int64_t *outMediaUs, bool allowPastMaxTime) const {
+ if (outMediaUs == NULL) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock autoLock(mLock);
+ return getMediaTime_l(realUs, outMediaUs, allowPastMaxTime);
+}
+
+status_t MediaClock::getMediaTime_l(
+ int64_t realUs, int64_t *outMediaUs, bool allowPastMaxTime) const {
+ if (mAnchorTimeRealUs == -1) {
+ return NO_INIT;
+ }
+
+ int64_t mediaUs = mAnchorTimeMediaUs
+ + (realUs - mAnchorTimeRealUs) * (double)mPlaybackRate;
+ if (mediaUs > mMaxTimeMediaUs && !allowPastMaxTime) {
+ mediaUs = mMaxTimeMediaUs;
+ }
+ if (mediaUs < mStartingTimeMediaUs) {
+ mediaUs = mStartingTimeMediaUs;
+ }
+ if (mediaUs < 0) {
+ mediaUs = 0;
+ }
+ *outMediaUs = mediaUs;
+ return OK;
+}
+
+status_t MediaClock::getRealTimeFor(
+ int64_t targetMediaUs, int64_t *outRealUs) const {
+ if (outRealUs == NULL) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock autoLock(mLock);
+ if (mPlaybackRate == 0.0) {
+ return NO_INIT;
+ }
+
+ int64_t nowUs = ALooper::GetNowUs();
+ int64_t nowMediaUs;
+ status_t status =
+ getMediaTime_l(nowUs, &nowMediaUs, true /* allowPastMaxTime */);
+ if (status != OK) {
+ return status;
+ }
+ *outRealUs = (targetMediaUs - nowMediaUs) / (double)mPlaybackRate + nowUs;
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 6ca123a..0597f1d 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -36,6 +36,7 @@
#include <media/stagefright/MediaCodecList.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaFilter.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/NativeWindowWrapper.h>
#include <private/android_filesystem_config.h>
@@ -173,7 +174,7 @@
}
// static
-void MediaCodec::PostReplyWithError(int32_t replyID, int32_t err) {
+void MediaCodec::PostReplyWithError(const sp<AReplyToken> &replyID, int32_t err) {
sp<AMessage> response = new AMessage;
response->setInt32("err", err);
response->postReply(replyID);
@@ -189,7 +190,16 @@
// quickly, violating the OpenMAX specs, until that is remedied
// we need to invest in an extra looper to free the main event
// queue.
- mCodec = new ACodec;
+
+ if (nameIsType || !strncasecmp(name.c_str(), "omx.", 4)) {
+ mCodec = new ACodec;
+ } else if (!nameIsType
+ && !strncasecmp(name.c_str(), "android.filter.", 15)) {
+ mCodec = new MediaFilter;
+ } else {
+ return NAME_NOT_FOUND;
+ }
+
bool needDedicatedLooper = false;
if (nameIsType && !strncasecmp(name.c_str(), "video/", 6)) {
needDedicatedLooper = true;
@@ -227,9 +237,9 @@
mLooper->registerHandler(this);
- mCodec->setNotificationMessage(new AMessage(kWhatCodecNotify, id()));
+ mCodec->setNotificationMessage(new AMessage(kWhatCodecNotify, this));
- sp<AMessage> msg = new AMessage(kWhatInit, id());
+ sp<AMessage> msg = new AMessage(kWhatInit, this);
msg->setString("name", name);
msg->setInt32("nameIsType", nameIsType);
@@ -242,7 +252,7 @@
}
status_t MediaCodec::setCallback(const sp<AMessage> &callback) {
- sp<AMessage> msg = new AMessage(kWhatSetCallback, id());
+ sp<AMessage> msg = new AMessage(kWhatSetCallback, this);
msg->setMessage("callback", callback);
sp<AMessage> response;
@@ -254,7 +264,7 @@
const sp<Surface> &nativeWindow,
const sp<ICrypto> &crypto,
uint32_t flags) {
- sp<AMessage> msg = new AMessage(kWhatConfigure, id());
+ sp<AMessage> msg = new AMessage(kWhatConfigure, this);
msg->setMessage("format", format);
msg->setInt32("flags", flags);
@@ -288,7 +298,7 @@
status_t MediaCodec::createInputSurface(
sp<IGraphicBufferProducer>* bufferProducer) {
- sp<AMessage> msg = new AMessage(kWhatCreateInputSurface, id());
+ sp<AMessage> msg = new AMessage(kWhatCreateInputSurface, this);
sp<AMessage> response;
status_t err = PostAndAwaitResponse(msg, &response);
@@ -307,21 +317,21 @@
}
status_t MediaCodec::start() {
- sp<AMessage> msg = new AMessage(kWhatStart, id());
+ sp<AMessage> msg = new AMessage(kWhatStart, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
}
status_t MediaCodec::stop() {
- sp<AMessage> msg = new AMessage(kWhatStop, id());
+ sp<AMessage> msg = new AMessage(kWhatStop, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
}
status_t MediaCodec::release() {
- sp<AMessage> msg = new AMessage(kWhatRelease, id());
+ sp<AMessage> msg = new AMessage(kWhatRelease, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
@@ -373,7 +383,7 @@
errorDetailMsg->clear();
}
- sp<AMessage> msg = new AMessage(kWhatQueueInputBuffer, id());
+ sp<AMessage> msg = new AMessage(kWhatQueueInputBuffer, this);
msg->setSize("index", index);
msg->setSize("offset", offset);
msg->setSize("size", size);
@@ -400,7 +410,7 @@
errorDetailMsg->clear();
}
- sp<AMessage> msg = new AMessage(kWhatQueueInputBuffer, id());
+ sp<AMessage> msg = new AMessage(kWhatQueueInputBuffer, this);
msg->setSize("index", index);
msg->setSize("offset", offset);
msg->setPointer("subSamples", (void *)subSamples);
@@ -419,7 +429,7 @@
}
status_t MediaCodec::dequeueInputBuffer(size_t *index, int64_t timeoutUs) {
- sp<AMessage> msg = new AMessage(kWhatDequeueInputBuffer, id());
+ sp<AMessage> msg = new AMessage(kWhatDequeueInputBuffer, this);
msg->setInt64("timeoutUs", timeoutUs);
sp<AMessage> response;
@@ -440,7 +450,7 @@
int64_t *presentationTimeUs,
uint32_t *flags,
int64_t timeoutUs) {
- sp<AMessage> msg = new AMessage(kWhatDequeueOutputBuffer, id());
+ sp<AMessage> msg = new AMessage(kWhatDequeueOutputBuffer, this);
msg->setInt64("timeoutUs", timeoutUs);
sp<AMessage> response;
@@ -459,7 +469,7 @@
}
status_t MediaCodec::renderOutputBufferAndRelease(size_t index) {
- sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, id());
+ sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, this);
msg->setSize("index", index);
msg->setInt32("render", true);
@@ -468,7 +478,7 @@
}
status_t MediaCodec::renderOutputBufferAndRelease(size_t index, int64_t timestampNs) {
- sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, id());
+ sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, this);
msg->setSize("index", index);
msg->setInt32("render", true);
msg->setInt64("timestampNs", timestampNs);
@@ -478,7 +488,7 @@
}
status_t MediaCodec::releaseOutputBuffer(size_t index) {
- sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, id());
+ sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, this);
msg->setSize("index", index);
sp<AMessage> response;
@@ -486,14 +496,14 @@
}
status_t MediaCodec::signalEndOfInputStream() {
- sp<AMessage> msg = new AMessage(kWhatSignalEndOfInputStream, id());
+ sp<AMessage> msg = new AMessage(kWhatSignalEndOfInputStream, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
}
status_t MediaCodec::getOutputFormat(sp<AMessage> *format) const {
- sp<AMessage> msg = new AMessage(kWhatGetOutputFormat, id());
+ sp<AMessage> msg = new AMessage(kWhatGetOutputFormat, this);
sp<AMessage> response;
status_t err;
@@ -507,7 +517,7 @@
}
status_t MediaCodec::getInputFormat(sp<AMessage> *format) const {
- sp<AMessage> msg = new AMessage(kWhatGetInputFormat, id());
+ sp<AMessage> msg = new AMessage(kWhatGetInputFormat, this);
sp<AMessage> response;
status_t err;
@@ -521,7 +531,7 @@
}
status_t MediaCodec::getName(AString *name) const {
- sp<AMessage> msg = new AMessage(kWhatGetName, id());
+ sp<AMessage> msg = new AMessage(kWhatGetName, this);
sp<AMessage> response;
status_t err;
@@ -535,7 +545,7 @@
}
status_t MediaCodec::getInputBuffers(Vector<sp<ABuffer> > *buffers) const {
- sp<AMessage> msg = new AMessage(kWhatGetBuffers, id());
+ sp<AMessage> msg = new AMessage(kWhatGetBuffers, this);
msg->setInt32("portIndex", kPortIndexInput);
msg->setPointer("buffers", buffers);
@@ -544,7 +554,7 @@
}
status_t MediaCodec::getOutputBuffers(Vector<sp<ABuffer> > *buffers) const {
- sp<AMessage> msg = new AMessage(kWhatGetBuffers, id());
+ sp<AMessage> msg = new AMessage(kWhatGetBuffers, this);
msg->setInt32("portIndex", kPortIndexOutput);
msg->setPointer("buffers", buffers);
@@ -602,20 +612,20 @@
}
status_t MediaCodec::flush() {
- sp<AMessage> msg = new AMessage(kWhatFlush, id());
+ sp<AMessage> msg = new AMessage(kWhatFlush, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
}
status_t MediaCodec::requestIDRFrame() {
- (new AMessage(kWhatRequestIDRFrame, id()))->post();
+ (new AMessage(kWhatRequestIDRFrame, this))->post();
return OK;
}
void MediaCodec::requestActivityNotification(const sp<AMessage> ¬ify) {
- sp<AMessage> msg = new AMessage(kWhatRequestActivityNotification, id());
+ sp<AMessage> msg = new AMessage(kWhatRequestActivityNotification, this);
msg->setMessage("notify", notify);
msg->post();
}
@@ -640,7 +650,7 @@
}
}
-bool MediaCodec::handleDequeueInputBuffer(uint32_t replyID, bool newRequest) {
+bool MediaCodec::handleDequeueInputBuffer(const sp<AReplyToken> &replyID, bool newRequest) {
if (!isExecuting() || (mFlags & kFlagIsAsync)
|| (newRequest && (mFlags & kFlagDequeueInputPending))) {
PostReplyWithError(replyID, INVALID_OPERATION);
@@ -664,7 +674,7 @@
return true;
}
-bool MediaCodec::handleDequeueOutputBuffer(uint32_t replyID, bool newRequest) {
+bool MediaCodec::handleDequeueOutputBuffer(const sp<AReplyToken> &replyID, bool newRequest) {
sp<AMessage> response = new AMessage;
if (!isExecuting() || (mFlags & kFlagIsAsync)
@@ -1188,7 +1198,7 @@
case kWhatInit:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (mState != UNINITIALIZED) {
@@ -1224,7 +1234,7 @@
case kWhatSetCallback:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (mState == UNINITIALIZED
@@ -1256,7 +1266,7 @@
case kWhatConfigure:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (mState != INITIALIZED) {
@@ -1313,7 +1323,7 @@
case kWhatCreateInputSurface:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
// Must be configured, but can't have been started yet.
@@ -1329,7 +1339,7 @@
case kWhatStart:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (mState == FLUSHED) {
@@ -1355,7 +1365,7 @@
State targetState =
(msg->what() == kWhatStop) ? INITIALIZED : UNINITIALIZED;
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (!((mFlags & kFlagIsComponentAllocated) && targetState == UNINITIALIZED) // See 1
@@ -1403,7 +1413,7 @@
case kWhatDequeueInputBuffer:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (mFlags & kFlagIsAsync) {
@@ -1435,7 +1445,7 @@
if (timeoutUs > 0ll) {
sp<AMessage> timeoutMsg =
- new AMessage(kWhatDequeueInputTimedOut, id());
+ new AMessage(kWhatDequeueInputTimedOut, this);
timeoutMsg->setInt32(
"generation", ++mDequeueInputTimeoutGeneration);
timeoutMsg->post(timeoutUs);
@@ -1464,7 +1474,7 @@
case kWhatQueueInputBuffer:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (!isExecuting()) {
@@ -1483,7 +1493,7 @@
case kWhatDequeueOutputBuffer:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (mFlags & kFlagIsAsync) {
@@ -1509,7 +1519,7 @@
if (timeoutUs > 0ll) {
sp<AMessage> timeoutMsg =
- new AMessage(kWhatDequeueOutputTimedOut, id());
+ new AMessage(kWhatDequeueOutputTimedOut, this);
timeoutMsg->setInt32(
"generation", ++mDequeueOutputTimeoutGeneration);
timeoutMsg->post(timeoutUs);
@@ -1538,7 +1548,7 @@
case kWhatReleaseOutputBuffer:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (!isExecuting()) {
@@ -1557,7 +1567,7 @@
case kWhatSignalEndOfInputStream:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (!isExecuting()) {
@@ -1575,7 +1585,7 @@
case kWhatGetBuffers:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (!isExecuting() || (mFlags & kFlagIsAsync)) {
@@ -1609,7 +1619,7 @@
case kWhatFlush:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (!isExecuting()) {
@@ -1635,7 +1645,7 @@
sp<AMessage> format =
(msg->what() == kWhatGetOutputFormat ? mOutputFormat : mInputFormat);
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if ((mState != CONFIGURED && mState != STARTING &&
@@ -1672,7 +1682,7 @@
case kWhatGetName:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (mComponentName.empty()) {
@@ -1688,7 +1698,7 @@
case kWhatSetParameters:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
sp<AMessage> params;
@@ -1742,7 +1752,7 @@
AString errorDetailMsg;
- sp<AMessage> msg = new AMessage(kWhatQueueInputBuffer, id());
+ sp<AMessage> msg = new AMessage(kWhatQueueInputBuffer, this);
msg->setSize("index", bufferIndex);
msg->setSize("offset", 0);
msg->setSize("size", csd->size());
@@ -2197,7 +2207,7 @@
}
status_t MediaCodec::setParameters(const sp<AMessage> ¶ms) {
- sp<AMessage> msg = new AMessage(kWhatSetParameters, id());
+ sp<AMessage> msg = new AMessage(kWhatSetParameters, this);
msg->setMessage("params", params);
sp<AMessage> response;
diff --git a/media/libstagefright/MediaCodecSource.cpp b/media/libstagefright/MediaCodecSource.cpp
index c26e909..b6fa810 100644
--- a/media/libstagefright/MediaCodecSource.cpp
+++ b/media/libstagefright/MediaCodecSource.cpp
@@ -121,7 +121,7 @@
mLooper->registerHandler(this);
mNotify = notify;
- sp<AMessage> msg = new AMessage(kWhatStart, id());
+ sp<AMessage> msg = new AMessage(kWhatStart, this);
msg->setObject("meta", meta);
return postSynchronouslyAndReturnError(msg);
}
@@ -137,19 +137,19 @@
mSource->stop();
ALOGV("source (%s) stopped", mIsAudio ? "audio" : "video");
- (new AMessage(kWhatStop, id()))->post();
+ (new AMessage(kWhatStop, this))->post();
}
void MediaCodecSource::Puller::pause() {
- (new AMessage(kWhatPause, id()))->post();
+ (new AMessage(kWhatPause, this))->post();
}
void MediaCodecSource::Puller::resume() {
- (new AMessage(kWhatResume, id()))->post();
+ (new AMessage(kWhatResume, this))->post();
}
void MediaCodecSource::Puller::schedulePull() {
- sp<AMessage> msg = new AMessage(kWhatPull, id());
+ sp<AMessage> msg = new AMessage(kWhatPull, this);
msg->setInt32("generation", mPullGeneration);
msg->post();
}
@@ -182,7 +182,7 @@
sp<AMessage> response = new AMessage;
response->setInt32("err", err);
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
response->postReply(replyID);
break;
@@ -269,13 +269,13 @@
}
status_t MediaCodecSource::start(MetaData* params) {
- sp<AMessage> msg = new AMessage(kWhatStart, mReflector->id());
+ sp<AMessage> msg = new AMessage(kWhatStart, mReflector);
msg->setObject("meta", params);
return postSynchronouslyAndReturnError(msg);
}
status_t MediaCodecSource::stop() {
- sp<AMessage> msg = new AMessage(kWhatStop, mReflector->id());
+ sp<AMessage> msg = new AMessage(kWhatStop, mReflector);
status_t err = postSynchronouslyAndReturnError(msg);
// mPuller->stop() needs to be done outside MediaCodecSource's looper,
@@ -294,7 +294,7 @@
}
status_t MediaCodecSource::pause() {
- (new AMessage(kWhatPause, mReflector->id()))->post();
+ (new AMessage(kWhatPause, mReflector))->post();
return OK;
}
@@ -422,8 +422,7 @@
}
}
- mEncoderActivityNotify = new AMessage(
- kWhatEncoderActivity, mReflector->id());
+ mEncoderActivityNotify = new AMessage(kWhatEncoderActivity, mReflector);
mEncoder->setCallback(mEncoderActivityNotify);
err = mEncoder->start();
@@ -492,7 +491,7 @@
if (mStopping && mEncoderReachedEOS) {
ALOGI("encoder (%s) stopped", mIsVideo ? "video" : "audio");
// posting reply to everyone that's waiting
- List<uint32_t>::iterator it;
+ List<sp<AReplyToken>>::iterator it;
for (it = mStopReplyIDQueue.begin();
it != mStopReplyIDQueue.end(); it++) {
(new AMessage)->postReply(*it);
@@ -620,8 +619,7 @@
resume(startTimeUs);
} else {
CHECK(mPuller != NULL);
- sp<AMessage> notify = new AMessage(
- kWhatPullerNotify, mReflector->id());
+ sp<AMessage> notify = new AMessage(kWhatPullerNotify, mReflector);
err = mPuller->start(params, notify);
if (err != OK) {
return err;
@@ -768,7 +766,7 @@
}
case kWhatStart:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
sp<RefBase> obj;
@@ -784,7 +782,7 @@
{
ALOGI("encoder (%s) stopping", mIsVideo ? "video" : "audio");
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
if (mEncoderReachedEOS) {
diff --git a/media/libstagefright/MediaMuxer.cpp b/media/libstagefright/MediaMuxer.cpp
index c7c6f34..b13877d 100644
--- a/media/libstagefright/MediaMuxer.cpp
+++ b/media/libstagefright/MediaMuxer.cpp
@@ -38,21 +38,6 @@
namespace android {
-MediaMuxer::MediaMuxer(const char *path, OutputFormat format)
- : mFormat(format),
- mState(UNINITIALIZED) {
- if (format == OUTPUT_FORMAT_MPEG_4) {
- mWriter = new MPEG4Writer(path);
- } else if (format == OUTPUT_FORMAT_WEBM) {
- mWriter = new WebmWriter(path);
- }
-
- if (mWriter != NULL) {
- mFileMeta = new MetaData;
- mState = INITIALIZED;
- }
-}
-
MediaMuxer::MediaMuxer(int fd, OutputFormat format)
: mFormat(format),
mState(UNINITIALIZED) {
diff --git a/media/libstagefright/MediaSync.cpp b/media/libstagefright/MediaSync.cpp
new file mode 100644
index 0000000..7b6c7d9
--- /dev/null
+++ b/media/libstagefright/MediaSync.cpp
@@ -0,0 +1,541 @@
+/*
+ * Copyright 2015 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 "MediaSync"
+#include <inttypes.h>
+
+#include <gui/BufferQueue.h>
+#include <gui/IGraphicBufferConsumer.h>
+#include <gui/IGraphicBufferProducer.h>
+
+#include <media/AudioTrack.h>
+#include <media/stagefright/MediaClock.h>
+#include <media/stagefright/MediaSync.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include <ui/GraphicBuffer.h>
+
+// Maximum late time allowed for a video frame to be rendered. When a video
+// frame arrives later than this number, it will be discarded without rendering.
+static const int64_t kMaxAllowedVideoLateTimeUs = 40000ll;
+
+namespace android {
+
+// static
+sp<MediaSync> MediaSync::create() {
+ sp<MediaSync> sync = new MediaSync();
+ sync->mLooper->registerHandler(sync);
+ return sync;
+}
+
+MediaSync::MediaSync()
+ : mIsAbandoned(false),
+ mMutex(),
+ mReleaseCondition(),
+ mNumOutstandingBuffers(0),
+ mNativeSampleRateInHz(0),
+ mNumFramesWritten(0),
+ mHasAudio(false),
+ mNextBufferItemMediaUs(-1),
+ mPlaybackRate(0.0) {
+ mMediaClock = new MediaClock;
+
+ mLooper = new ALooper;
+ mLooper->setName("MediaSync");
+ mLooper->start(false, false, ANDROID_PRIORITY_AUDIO);
+}
+
+MediaSync::~MediaSync() {
+ if (mInput != NULL) {
+ mInput->consumerDisconnect();
+ }
+ if (mOutput != NULL) {
+ mOutput->disconnect(NATIVE_WINDOW_API_MEDIA);
+ }
+
+ if (mLooper != NULL) {
+ mLooper->unregisterHandler(id());
+ mLooper->stop();
+ }
+}
+
+status_t MediaSync::configureSurface(const sp<IGraphicBufferProducer> &output) {
+ Mutex::Autolock lock(mMutex);
+
+ // TODO: support suface change.
+ if (mOutput != NULL) {
+ ALOGE("configureSurface: output surface has already been configured.");
+ return INVALID_OPERATION;
+ }
+
+ if (output != NULL) {
+ IGraphicBufferProducer::QueueBufferOutput queueBufferOutput;
+ sp<OutputListener> listener(new OutputListener(this));
+ IInterface::asBinder(output)->linkToDeath(listener);
+ status_t status =
+ output->connect(listener,
+ NATIVE_WINDOW_API_MEDIA,
+ true /* producerControlledByApp */,
+ &queueBufferOutput);
+ if (status != NO_ERROR) {
+ ALOGE("configureSurface: failed to connect (%d)", status);
+ return status;
+ }
+
+ mOutput = output;
+ }
+
+ return NO_ERROR;
+}
+
+// |audioTrack| is used only for querying information.
+status_t MediaSync::configureAudioTrack(
+ const sp<AudioTrack> &audioTrack, uint32_t nativeSampleRateInHz) {
+ Mutex::Autolock lock(mMutex);
+
+ // TODO: support audio track change.
+ if (mAudioTrack != NULL) {
+ ALOGE("configureAudioTrack: audioTrack has already been configured.");
+ return INVALID_OPERATION;
+ }
+
+ mAudioTrack = audioTrack;
+ mNativeSampleRateInHz = nativeSampleRateInHz;
+
+ return NO_ERROR;
+}
+
+status_t MediaSync::createInputSurface(
+ sp<IGraphicBufferProducer> *outBufferProducer) {
+ if (outBufferProducer == NULL) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mMutex);
+
+ if (mOutput == NULL) {
+ return NO_INIT;
+ }
+
+ if (mInput != NULL) {
+ return INVALID_OPERATION;
+ }
+
+ sp<IGraphicBufferProducer> bufferProducer;
+ sp<IGraphicBufferConsumer> bufferConsumer;
+ BufferQueue::createBufferQueue(&bufferProducer, &bufferConsumer);
+
+ sp<InputListener> listener(new InputListener(this));
+ IInterface::asBinder(bufferConsumer)->linkToDeath(listener);
+ status_t status =
+ bufferConsumer->consumerConnect(listener, false /* controlledByApp */);
+ if (status == NO_ERROR) {
+ bufferConsumer->setConsumerName(String8("MediaSync"));
+ *outBufferProducer = bufferProducer;
+ mInput = bufferConsumer;
+ }
+ return status;
+}
+
+status_t MediaSync::setPlaybackRate(float rate) {
+ if (rate < 0.0) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mMutex);
+
+ if (rate > mPlaybackRate) {
+ mNextBufferItemMediaUs = -1;
+ }
+ mPlaybackRate = rate;
+ mMediaClock->setPlaybackRate(rate);
+ onDrainVideo_l();
+
+ return OK;
+}
+
+sp<const MediaClock> MediaSync::getMediaClock() {
+ return mMediaClock;
+}
+
+status_t MediaSync::updateQueuedAudioData(
+ size_t sizeInBytes, int64_t presentationTimeUs) {
+ if (sizeInBytes == 0) {
+ return OK;
+ }
+
+ Mutex::Autolock lock(mMutex);
+
+ if (mAudioTrack == NULL) {
+ ALOGW("updateQueuedAudioData: audioTrack has NOT been configured.");
+ return INVALID_OPERATION;
+ }
+
+ int64_t numFrames = sizeInBytes / mAudioTrack->frameSize();
+ int64_t maxMediaTimeUs = presentationTimeUs
+ + getDurationIfPlayedAtNativeSampleRate_l(numFrames);
+ mNumFramesWritten += numFrames;
+
+ int64_t nowUs = ALooper::GetNowUs();
+ int64_t nowMediaUs = maxMediaTimeUs
+ - getDurationIfPlayedAtNativeSampleRate_l(mNumFramesWritten)
+ + getPlayedOutAudioDurationMedia_l(nowUs);
+
+ int64_t oldRealTime = -1;
+ if (mNextBufferItemMediaUs != -1) {
+ oldRealTime = getRealTime(mNextBufferItemMediaUs, nowUs);
+ }
+
+ mMediaClock->updateAnchor(nowMediaUs, nowUs, maxMediaTimeUs);
+ mHasAudio = true;
+
+ if (oldRealTime != -1) {
+ int64_t newRealTime = getRealTime(mNextBufferItemMediaUs, nowUs);
+ if (newRealTime < oldRealTime) {
+ mNextBufferItemMediaUs = -1;
+ onDrainVideo_l();
+ }
+ }
+
+ return OK;
+}
+
+void MediaSync::setName(const AString &name) {
+ Mutex::Autolock lock(mMutex);
+ mInput->setConsumerName(String8(name.c_str()));
+}
+
+int64_t MediaSync::getRealTime(int64_t mediaTimeUs, int64_t nowUs) {
+ int64_t realUs;
+ if (mMediaClock->getRealTimeFor(mediaTimeUs, &realUs) != OK) {
+ // If failed to get current position, e.g. due to audio clock is
+ // not ready, then just play out video immediately without delay.
+ return nowUs;
+ }
+ return realUs;
+}
+
+int64_t MediaSync::getDurationIfPlayedAtNativeSampleRate_l(int64_t numFrames) {
+ return (numFrames * 1000000LL / mNativeSampleRateInHz);
+}
+
+int64_t MediaSync::getPlayedOutAudioDurationMedia_l(int64_t nowUs) {
+ CHECK(mAudioTrack != NULL);
+
+ uint32_t numFramesPlayed;
+ int64_t numFramesPlayedAt;
+ AudioTimestamp ts;
+ static const int64_t kStaleTimestamp100ms = 100000;
+
+ status_t res = mAudioTrack->getTimestamp(ts);
+ if (res == OK) {
+ // case 1: mixing audio tracks.
+ numFramesPlayed = ts.mPosition;
+ numFramesPlayedAt =
+ ts.mTime.tv_sec * 1000000LL + ts.mTime.tv_nsec / 1000;
+ const int64_t timestampAge = nowUs - numFramesPlayedAt;
+ if (timestampAge > kStaleTimestamp100ms) {
+ // This is an audio FIXME.
+ // getTimestamp returns a timestamp which may come from audio
+ // mixing threads. After pausing, the MixerThread may go idle,
+ // thus the mTime estimate may become stale. Assuming that the
+ // MixerThread runs 20ms, with FastMixer at 5ms, the max latency
+ // should be about 25ms with an average around 12ms (to be
+ // verified). For safety we use 100ms.
+ ALOGV("getTimestamp: returned stale timestamp nowUs(%lld) "
+ "numFramesPlayedAt(%lld)",
+ (long long)nowUs, (long long)numFramesPlayedAt);
+ numFramesPlayedAt = nowUs - kStaleTimestamp100ms;
+ }
+ //ALOGD("getTimestamp: OK %d %lld",
+ // numFramesPlayed, (long long)numFramesPlayedAt);
+ } else if (res == WOULD_BLOCK) {
+ // case 2: transitory state on start of a new track
+ numFramesPlayed = 0;
+ numFramesPlayedAt = nowUs;
+ //ALOGD("getTimestamp: WOULD_BLOCK %d %lld",
+ // numFramesPlayed, (long long)numFramesPlayedAt);
+ } else {
+ // case 3: transitory at new track or audio fast tracks.
+ res = mAudioTrack->getPosition(&numFramesPlayed);
+ CHECK_EQ(res, (status_t)OK);
+ numFramesPlayedAt = nowUs;
+ numFramesPlayedAt += 1000LL * mAudioTrack->latency() / 2; /* XXX */
+ //ALOGD("getPosition: %d %lld", numFramesPlayed, numFramesPlayedAt);
+ }
+
+ //can't be negative until 12.4 hrs, test.
+ //CHECK_EQ(numFramesPlayed & (1 << 31), 0);
+ int64_t durationUs =
+ getDurationIfPlayedAtNativeSampleRate_l(numFramesPlayed)
+ + nowUs - numFramesPlayedAt;
+ if (durationUs < 0) {
+ // Occurs when numFramesPlayed position is very small and the following:
+ // (1) In case 1, the time nowUs is computed before getTimestamp() is
+ // called and numFramesPlayedAt is greater than nowUs by time more
+ // than numFramesPlayed.
+ // (2) In case 3, using getPosition and adding mAudioTrack->latency()
+ // to numFramesPlayedAt, by a time amount greater than
+ // numFramesPlayed.
+ //
+ // Both of these are transitory conditions.
+ ALOGV("getPlayedOutAudioDurationMedia_l: negative duration %lld "
+ "set to zero", (long long)durationUs);
+ durationUs = 0;
+ }
+ ALOGV("getPlayedOutAudioDurationMedia_l(%lld) nowUs(%lld) frames(%u) "
+ "framesAt(%lld)",
+ (long long)durationUs, (long long)nowUs, numFramesPlayed,
+ (long long)numFramesPlayedAt);
+ return durationUs;
+}
+
+void MediaSync::onDrainVideo_l() {
+ if (!isPlaying()) {
+ return;
+ }
+
+ int64_t nowUs = ALooper::GetNowUs();
+
+ while (!mBufferItems.empty()) {
+ BufferItem *bufferItem = &*mBufferItems.begin();
+ int64_t itemMediaUs = bufferItem->mTimestamp / 1000;
+ int64_t itemRealUs = getRealTime(itemMediaUs, nowUs);
+ if (itemRealUs <= nowUs) {
+ if (mHasAudio) {
+ if (nowUs - itemRealUs <= kMaxAllowedVideoLateTimeUs) {
+ renderOneBufferItem_l(*bufferItem);
+ } else {
+ // too late.
+ returnBufferToInput_l(
+ bufferItem->mGraphicBuffer, bufferItem->mFence);
+ }
+ } else {
+ // always render video buffer in video-only mode.
+ renderOneBufferItem_l(*bufferItem);
+
+ // smooth out videos >= 10fps
+ mMediaClock->updateAnchor(
+ itemMediaUs, nowUs, itemMediaUs + 100000);
+ }
+
+ mBufferItems.erase(mBufferItems.begin());
+
+ if (mBufferItems.empty()) {
+ mNextBufferItemMediaUs = -1;
+ }
+ } else {
+ if (mNextBufferItemMediaUs == -1
+ || mNextBufferItemMediaUs != itemMediaUs) {
+ sp<AMessage> msg = new AMessage(kWhatDrainVideo, this);
+ msg->post(itemRealUs - nowUs);
+ }
+ break;
+ }
+ }
+}
+
+void MediaSync::onFrameAvailableFromInput() {
+ Mutex::Autolock lock(mMutex);
+
+ // If there are too many outstanding buffers, wait until a buffer is
+ // released back to the input in onBufferReleased.
+ while (mNumOutstandingBuffers >= MAX_OUTSTANDING_BUFFERS) {
+ mReleaseCondition.wait(mMutex);
+
+ // If the sync is abandoned while we are waiting, the release
+ // condition variable will be broadcast, and we should just return
+ // without attempting to do anything more (since the input queue will
+ // also be abandoned).
+ if (mIsAbandoned) {
+ return;
+ }
+ }
+ ++mNumOutstandingBuffers;
+
+ // Acquire and detach the buffer from the input.
+ BufferItem bufferItem;
+ status_t status = mInput->acquireBuffer(&bufferItem, 0 /* presentWhen */);
+ if (status != NO_ERROR) {
+ ALOGE("acquiring buffer from input failed (%d)", status);
+ return;
+ }
+
+ ALOGV("acquired buffer %#llx from input", (long long)bufferItem.mGraphicBuffer->getId());
+
+ status = mInput->detachBuffer(bufferItem.mBuf);
+ if (status != NO_ERROR) {
+ ALOGE("detaching buffer from input failed (%d)", status);
+ if (status == NO_INIT) {
+ // If the input has been abandoned, move on.
+ onAbandoned_l(true /* isInput */);
+ }
+ return;
+ }
+
+ mBufferItems.push_back(bufferItem);
+ onDrainVideo_l();
+}
+
+void MediaSync::renderOneBufferItem_l( const BufferItem &bufferItem) {
+ IGraphicBufferProducer::QueueBufferInput queueInput(
+ bufferItem.mTimestamp,
+ bufferItem.mIsAutoTimestamp,
+ bufferItem.mDataSpace,
+ bufferItem.mCrop,
+ static_cast<int32_t>(bufferItem.mScalingMode),
+ bufferItem.mTransform,
+ bufferItem.mIsDroppable,
+ bufferItem.mFence);
+
+ // Attach and queue the buffer to the output.
+ int slot;
+ status_t status = mOutput->attachBuffer(&slot, bufferItem.mGraphicBuffer);
+ ALOGE_IF(status != NO_ERROR, "attaching buffer to output failed (%d)", status);
+ if (status == NO_ERROR) {
+ IGraphicBufferProducer::QueueBufferOutput queueOutput;
+ status = mOutput->queueBuffer(slot, queueInput, &queueOutput);
+ ALOGE_IF(status != NO_ERROR, "queueing buffer to output failed (%d)", status);
+ }
+
+ if (status != NO_ERROR) {
+ returnBufferToInput_l(bufferItem.mGraphicBuffer, bufferItem.mFence);
+ if (status == NO_INIT) {
+ // If the output has been abandoned, move on.
+ onAbandoned_l(false /* isInput */);
+ }
+ return;
+ }
+
+ ALOGV("queued buffer %#llx to output", (long long)bufferItem.mGraphicBuffer->getId());
+}
+
+void MediaSync::onBufferReleasedByOutput() {
+ Mutex::Autolock lock(mMutex);
+
+ sp<GraphicBuffer> buffer;
+ sp<Fence> fence;
+ status_t status = mOutput->detachNextBuffer(&buffer, &fence);
+ ALOGE_IF(status != NO_ERROR, "detaching buffer from output failed (%d)", status);
+
+ if (status == NO_INIT) {
+ // If the output has been abandoned, we can't do anything else,
+ // since buffer is invalid.
+ onAbandoned_l(false /* isInput */);
+ return;
+ }
+
+ ALOGV("detached buffer %#llx from output", (long long)buffer->getId());
+
+ // If we've been abandoned, we can't return the buffer to the input, so just
+ // move on.
+ if (mIsAbandoned) {
+ return;
+ }
+
+ returnBufferToInput_l(buffer, fence);
+}
+
+void MediaSync::returnBufferToInput_l(
+ const sp<GraphicBuffer> &buffer, const sp<Fence> &fence) {
+ // Attach and release the buffer back to the input.
+ int consumerSlot;
+ status_t status = mInput->attachBuffer(&consumerSlot, buffer);
+ ALOGE_IF(status != NO_ERROR, "attaching buffer to input failed (%d)", status);
+ if (status == NO_ERROR) {
+ status = mInput->releaseBuffer(consumerSlot, 0 /* frameNumber */,
+ EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, fence);
+ ALOGE_IF(status != NO_ERROR, "releasing buffer to input failed (%d)", status);
+ }
+
+ if (status != NO_ERROR) {
+ // TODO: do we need to try to return this buffer later?
+ return;
+ }
+
+ ALOGV("released buffer %#llx to input", (long long)buffer->getId());
+
+ // Notify any waiting onFrameAvailable calls.
+ --mNumOutstandingBuffers;
+ mReleaseCondition.signal();
+}
+
+void MediaSync::onAbandoned_l(bool isInput) {
+ ALOGE("the %s has abandoned me", (isInput ? "input" : "output"));
+ if (!mIsAbandoned) {
+ if (isInput) {
+ mOutput->disconnect(NATIVE_WINDOW_API_MEDIA);
+ } else {
+ mInput->consumerDisconnect();
+ }
+ mIsAbandoned = true;
+ }
+ mReleaseCondition.broadcast();
+}
+
+void MediaSync::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatDrainVideo:
+ {
+ Mutex::Autolock lock(mMutex);
+ onDrainVideo_l();
+ break;
+ }
+
+ default:
+ TRESPASS();
+ break;
+ }
+}
+
+MediaSync::InputListener::InputListener(const sp<MediaSync> &sync)
+ : mSync(sync) {}
+
+MediaSync::InputListener::~InputListener() {}
+
+void MediaSync::InputListener::onFrameAvailable(const BufferItem &/* item */) {
+ mSync->onFrameAvailableFromInput();
+}
+
+// We don't care about sideband streams, since we won't relay them.
+void MediaSync::InputListener::onSidebandStreamChanged() {
+ ALOGE("onSidebandStreamChanged: got sideband stream unexpectedly.");
+}
+
+
+void MediaSync::InputListener::binderDied(const wp<IBinder> &/* who */) {
+ Mutex::Autolock lock(mSync->mMutex);
+ mSync->onAbandoned_l(true /* isInput */);
+}
+
+MediaSync::OutputListener::OutputListener(const sp<MediaSync> &sync)
+ : mSync(sync) {}
+
+MediaSync::OutputListener::~OutputListener() {}
+
+void MediaSync::OutputListener::onBufferReleased() {
+ mSync->onBufferReleasedByOutput();
+}
+
+void MediaSync::OutputListener::binderDied(const wp<IBinder> &/* who */) {
+ Mutex::Autolock lock(mSync->mMutex);
+ mSync->onAbandoned_l(false /* isInput */);
+}
+
+} // namespace android
diff --git a/media/libstagefright/NuCachedSource2.cpp b/media/libstagefright/NuCachedSource2.cpp
index 7d7d631..8d70e50 100644
--- a/media/libstagefright/NuCachedSource2.cpp
+++ b/media/libstagefright/NuCachedSource2.cpp
@@ -226,7 +226,7 @@
mLooper->start(false /* runOnCallingThread */, true /* canCallJava */);
Mutex::Autolock autoLock(mLock);
- (new AMessage(kWhatFetchMore, mReflector->id()))->post();
+ (new AMessage(kWhatFetchMore, mReflector))->post();
}
NuCachedSource2::~NuCachedSource2() {
@@ -433,7 +433,7 @@
delayUs = 100000ll;
}
- (new AMessage(kWhatFetchMore, mReflector->id()))->post(delayUs);
+ (new AMessage(kWhatFetchMore, mReflector))->post(delayUs);
}
void NuCachedSource2::onRead(const sp<AMessage> &msg) {
@@ -522,7 +522,7 @@
return size;
}
- sp<AMessage> msg = new AMessage(kWhatRead, mReflector->id());
+ sp<AMessage> msg = new AMessage(kWhatRead, mReflector);
msg->setInt64("offset", offset);
msg->setPointer("data", data);
msg->setSize("size", size);
diff --git a/media/libstagefright/ProcessInfo.cpp b/media/libstagefright/ProcessInfo.cpp
new file mode 100644
index 0000000..b4172b3
--- /dev/null
+++ b/media/libstagefright/ProcessInfo.cpp
@@ -0,0 +1,53 @@
+/*
+ * Copyright (C) 2015 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 "ProcessInfo"
+#include <utils/Log.h>
+
+#include <media/stagefright/ProcessInfo.h>
+
+#include <binder/IProcessInfoService.h>
+#include <binder/IServiceManager.h>
+
+namespace android {
+
+ProcessInfo::ProcessInfo() {}
+
+bool ProcessInfo::getPriority(int pid, int* priority) {
+ sp<IBinder> binder = defaultServiceManager()->getService(String16("processinfo"));
+ sp<IProcessInfoService> service = interface_cast<IProcessInfoService>(binder);
+
+ size_t length = 1;
+ int32_t states;
+ status_t err = service->getProcessStatesFromPids(length, &pid, &states);
+ if (err != OK) {
+ ALOGE("getProcessStatesFromPids failed");
+ return false;
+ }
+ ALOGV("pid %d states %d", pid, states);
+ if (states < 0) {
+ return false;
+ }
+
+ // Use process state as the priority. Lower the value, higher the priority.
+ *priority = states;
+ return true;
+}
+
+ProcessInfo::~ProcessInfo() {}
+
+} // namespace android
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index b3a79a0..8506e37 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -344,6 +344,28 @@
buffer->meta()->setInt32("csd", true);
buffer->meta()->setInt64("timeUs", 0);
msg->setBuffer("csd-0", buffer);
+
+ if (!meta->findData(kKeyOpusCodecDelay, &type, &data, &size)) {
+ return -EINVAL;
+ }
+
+ buffer = new ABuffer(size);
+ memcpy(buffer->data(), data, size);
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-1", buffer);
+
+ if (!meta->findData(kKeyOpusSeekPreRoll, &type, &data, &size)) {
+ return -EINVAL;
+ }
+
+ buffer = new ABuffer(size);
+ memcpy(buffer->data(), data, size);
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-2", buffer);
}
*format = msg;
@@ -800,5 +822,36 @@
return AString("<no-scheme URI suppressed>");
}
+HLSTime::HLSTime(const sp<AMessage>& meta) :
+ mSeq(-1),
+ mTimeUs(-1ll),
+ mMeta(meta) {
+ if (meta != NULL) {
+ CHECK(meta->findInt32("discontinuitySeq", &mSeq));
+ CHECK(meta->findInt64("timeUs", &mTimeUs));
+ }
+}
+
+int64_t HLSTime::getSegmentTimeUs(bool midpoint) const {
+ int64_t segmentStartTimeUs = -1ll;
+ if (mMeta != NULL) {
+ CHECK(mMeta->findInt64("segmentStartTimeUs", &segmentStartTimeUs));
+ if (midpoint) {
+ int64_t durationUs;
+ CHECK(mMeta->findInt64("segmentDurationUs", &durationUs));
+ segmentStartTimeUs += durationUs / 2;
+ }
+ }
+ return segmentStartTimeUs;
+}
+
+bool operator <(const HLSTime &t0, const HLSTime &t1) {
+ // we can only compare discontinuity sequence and timestamp.
+ // (mSegmentTimeUs is not reliable in live streaming case, it's the
+ // time starting from beginning of playlist but playlist could change.)
+ return t0.mSeq < t1.mSeq
+ || (t0.mSeq == t1.mSeq && t0.mTimeUs < t1.mTimeUs);
+}
+
} // namespace android
diff --git a/media/libstagefright/avc_utils.cpp b/media/libstagefright/avc_utils.cpp
index 5ec3438..8ef2dca 100644
--- a/media/libstagefright/avc_utils.cpp
+++ b/media/libstagefright/avc_utils.cpp
@@ -26,6 +26,7 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
+#include <utils/misc.h>
namespace android {
@@ -186,17 +187,31 @@
if (aspect_ratio_idc == 255 /* extendedSAR */) {
sar_width = br.getBits(16);
sar_height = br.getBits(16);
- } else if (aspect_ratio_idc > 0 && aspect_ratio_idc < 14) {
- static const int32_t kFixedSARWidth[] = {
- 1, 12, 10, 16, 40, 24, 20, 32, 80, 18, 15, 64, 160
+ } else {
+ static const struct { unsigned width, height; } kFixedSARs[] = {
+ { 0, 0 }, // Invalid
+ { 1, 1 },
+ { 12, 11 },
+ { 10, 11 },
+ { 16, 11 },
+ { 40, 33 },
+ { 24, 11 },
+ { 20, 11 },
+ { 32, 11 },
+ { 80, 33 },
+ { 18, 11 },
+ { 15, 11 },
+ { 64, 33 },
+ { 160, 99 },
+ { 4, 3 },
+ { 3, 2 },
+ { 2, 1 },
};
- static const int32_t kFixedSARHeight[] = {
- 1, 11, 11, 11, 33, 11, 11, 11, 33, 11, 11, 33, 99
- };
-
- sar_width = kFixedSARWidth[aspect_ratio_idc - 1];
- sar_height = kFixedSARHeight[aspect_ratio_idc - 1];
+ if (aspect_ratio_idc > 0 && aspect_ratio_idc < NELEM(kFixedSARs)) {
+ sar_width = kFixedSARs[aspect_ratio_idc].width;
+ sar_height = kFixedSARs[aspect_ratio_idc].height;
+ }
}
}
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 495bad0..10937ec 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -623,7 +623,7 @@
} else {
int64_t currentTime = mBufferTimestamps.top();
currentTime += mStreamInfo->aacSamplesPerFrame *
- 1000000ll / mStreamInfo->sampleRate;
+ 1000000ll / mStreamInfo->aacSampleRate;
mBufferTimestamps.add(currentTime);
}
} else {
@@ -874,7 +874,7 @@
// adjust/interpolate next time stamp
*currentBufLeft -= decodedSize;
*nextTimeStamp += mStreamInfo->aacSamplesPerFrame *
- 1000000ll / mStreamInfo->sampleRate;
+ 1000000ll / mStreamInfo->aacSampleRate;
ALOGV("adjusted nextTimeStamp/size to %lld/%d",
(long long) *nextTimeStamp, *currentBufLeft);
} else {
@@ -975,6 +975,7 @@
mBufferSizes.clear();
mDecodedSizes.clear();
mLastInHeader = NULL;
+ mEndOfInput = false;
} else {
int avail;
while ((avail = outputDelayRingBufferSamplesAvailable()) > 0) {
@@ -989,6 +990,7 @@
mOutputBufferCount++;
}
mOutputDelayRingBufferReadPos = mOutputDelayRingBufferWritePos;
+ mEndOfOutput = false;
}
}
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
index 8a95643..6e6a78a 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
@@ -38,7 +38,10 @@
NULL /* profileLevels */, 0 /* numProfileLevels */,
320 /* width */, 240 /* height */, callbacks, appData, component),
mMode(codingType == OMX_VIDEO_CodingVP8 ? MODE_VP8 : MODE_VP9),
+ mEOSStatus(INPUT_DATA_AVAILABLE),
mCtx(NULL),
+ mFrameParallelMode(false),
+ mTimeStampIdx(0),
mImg(NULL) {
// arbitrary from avc/hevc as vpx does not specify a min compression ratio
const size_t kMinCompressionRatio = mMode == MODE_VP8 ? 2 : 4;
@@ -51,9 +54,7 @@
}
SoftVPX::~SoftVPX() {
- vpx_codec_destroy((vpx_codec_ctx_t *)mCtx);
- delete (vpx_codec_ctx_t *)mCtx;
- mCtx = NULL;
+ destroyDecoder();
}
static int GetCPUCoreCount() {
@@ -73,12 +74,19 @@
mCtx = new vpx_codec_ctx_t;
vpx_codec_err_t vpx_err;
vpx_codec_dec_cfg_t cfg;
+ vpx_codec_flags_t flags;
memset(&cfg, 0, sizeof(vpx_codec_dec_cfg_t));
+ memset(&flags, 0, sizeof(vpx_codec_flags_t));
cfg.threads = GetCPUCoreCount();
+
+ if (mFrameParallelMode) {
+ flags |= VPX_CODEC_USE_FRAME_THREADING;
+ }
+
if ((vpx_err = vpx_codec_dec_init(
(vpx_codec_ctx_t *)mCtx,
mMode == MODE_VP8 ? &vpx_codec_vp8_dx_algo : &vpx_codec_vp9_dx_algo,
- &cfg, 0))) {
+ &cfg, flags))) {
ALOGE("on2 decoder failed to initialize. (%d)", vpx_err);
return UNKNOWN_ERROR;
}
@@ -86,86 +94,155 @@
return OK;
}
+status_t SoftVPX::destroyDecoder() {
+ vpx_codec_destroy((vpx_codec_ctx_t *)mCtx);
+ delete (vpx_codec_ctx_t *)mCtx;
+ mCtx = NULL;
+ return OK;
+}
+
+bool SoftVPX::outputBuffers(bool flushDecoder, bool display, bool eos, bool *portWillReset) {
+ List<BufferInfo *> &inQueue = getPortQueue(0);
+ List<BufferInfo *> &outQueue = getPortQueue(1);
+ BufferInfo *outInfo = NULL;
+ OMX_BUFFERHEADERTYPE *outHeader = NULL;
+ vpx_codec_iter_t iter = NULL;
+
+ if (flushDecoder && mFrameParallelMode) {
+ // Flush decoder by passing NULL data ptr and 0 size.
+ // Ideally, this should never fail.
+ if (vpx_codec_decode((vpx_codec_ctx_t *)mCtx, NULL, 0, NULL, 0)) {
+ ALOGE("Failed to flush on2 decoder.");
+ return false;
+ }
+ }
+
+ if (!display) {
+ if (!flushDecoder) {
+ ALOGE("Invalid operation.");
+ return false;
+ }
+ // Drop all the decoded frames in decoder.
+ while ((mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter))) {
+ }
+ return true;
+ }
+
+ while (!outQueue.empty()) {
+ if (mImg == NULL) {
+ mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter);
+ if (mImg == NULL) {
+ break;
+ }
+ }
+ uint32_t width = mImg->d_w;
+ uint32_t height = mImg->d_h;
+ outInfo = *outQueue.begin();
+ outHeader = outInfo->mHeader;
+ CHECK_EQ(mImg->fmt, IMG_FMT_I420);
+ handlePortSettingsChange(portWillReset, width, height);
+ if (*portWillReset) {
+ return true;
+ }
+
+ outHeader->nOffset = 0;
+ outHeader->nFilledLen = (width * height * 3) / 2;
+ outHeader->nFlags = 0;
+ outHeader->nTimeStamp = *(OMX_TICKS *)mImg->user_priv;
+
+ uint8_t *dst = outHeader->pBuffer;
+ const uint8_t *srcY = (const uint8_t *)mImg->planes[PLANE_Y];
+ const uint8_t *srcU = (const uint8_t *)mImg->planes[PLANE_U];
+ const uint8_t *srcV = (const uint8_t *)mImg->planes[PLANE_V];
+ size_t srcYStride = mImg->stride[PLANE_Y];
+ size_t srcUStride = mImg->stride[PLANE_U];
+ size_t srcVStride = mImg->stride[PLANE_V];
+ copyYV12FrameToOutputBuffer(dst, srcY, srcU, srcV, srcYStride, srcUStride, srcVStride);
+
+ mImg = NULL;
+ outInfo->mOwnedByUs = false;
+ outQueue.erase(outQueue.begin());
+ outInfo = NULL;
+ notifyFillBufferDone(outHeader);
+ outHeader = NULL;
+ }
+
+ if (!eos) {
+ return true;
+ }
+
+ if (!outQueue.empty()) {
+ outInfo = *outQueue.begin();
+ outQueue.erase(outQueue.begin());
+ outHeader = outInfo->mHeader;
+ outHeader->nTimeStamp = 0;
+ outHeader->nFilledLen = 0;
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+ mEOSStatus = OUTPUT_FRAMES_FLUSHED;
+ }
+ return true;
+}
+
void SoftVPX::onQueueFilled(OMX_U32 /* portIndex */) {
- if (mOutputPortSettingsChange != NONE) {
+ if (mOutputPortSettingsChange != NONE || mEOSStatus == OUTPUT_FRAMES_FLUSHED) {
return;
}
List<BufferInfo *> &inQueue = getPortQueue(0);
List<BufferInfo *> &outQueue = getPortQueue(1);
bool EOSseen = false;
+ vpx_codec_err_t err;
+ bool portWillReset = false;
- while (!inQueue.empty() && !outQueue.empty()) {
- BufferInfo *inInfo = *inQueue.begin();
- OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
-
- BufferInfo *outInfo = *outQueue.begin();
- OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
-
- if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- EOSseen = true;
- if (inHeader->nFilledLen == 0) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
-
- outHeader->nFilledLen = 0;
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
-
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
- }
- }
-
- if (mImg == NULL) {
- if (vpx_codec_decode(
- (vpx_codec_ctx_t *)mCtx,
- inHeader->pBuffer + inHeader->nOffset,
- inHeader->nFilledLen,
- NULL,
- 0)) {
- ALOGE("on2 decoder failed to decode frame.");
-
+ while ((mEOSStatus == INPUT_EOS_SEEN || !inQueue.empty())
+ && !outQueue.empty()) {
+ // Output the pending frames that left from last port reset or decoder flush.
+ if (mEOSStatus == INPUT_EOS_SEEN || mImg != NULL) {
+ if (!outputBuffers(
+ mEOSStatus == INPUT_EOS_SEEN, true /* display */,
+ mEOSStatus == INPUT_EOS_SEEN, &portWillReset)) {
+ ALOGE("on2 decoder failed to output frame.");
notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
return;
}
- vpx_codec_iter_t iter = NULL;
- mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter);
- }
-
- if (mImg != NULL) {
- CHECK_EQ(mImg->fmt, IMG_FMT_I420);
-
- uint32_t width = mImg->d_w;
- uint32_t height = mImg->d_h;
- bool portWillReset = false;
- handlePortSettingsChange(&portWillReset, width, height);
- if (portWillReset) {
+ if (portWillReset || mEOSStatus == OUTPUT_FRAMES_FLUSHED ||
+ mEOSStatus == INPUT_EOS_SEEN) {
return;
}
+ }
- outHeader->nOffset = 0;
- outHeader->nFilledLen = (width * height * 3) / 2;
- outHeader->nFlags = EOSseen ? OMX_BUFFERFLAG_EOS : 0;
- outHeader->nTimeStamp = inHeader->nTimeStamp;
+ BufferInfo *inInfo = *inQueue.begin();
+ OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+ mTimeStamps[mTimeStampIdx] = inHeader->nTimeStamp;
- uint8_t *dst = outHeader->pBuffer;
- const uint8_t *srcY = (const uint8_t *)mImg->planes[PLANE_Y];
- const uint8_t *srcU = (const uint8_t *)mImg->planes[PLANE_U];
- const uint8_t *srcV = (const uint8_t *)mImg->planes[PLANE_V];
- size_t srcYStride = mImg->stride[PLANE_Y];
- size_t srcUStride = mImg->stride[PLANE_U];
- size_t srcVStride = mImg->stride[PLANE_V];
- copyYV12FrameToOutputBuffer(dst, srcY, srcU, srcV, srcYStride, srcUStride, srcVStride);
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ mEOSStatus = INPUT_EOS_SEEN;
+ EOSseen = true;
+ }
- mImg = NULL;
- outInfo->mOwnedByUs = false;
- outQueue.erase(outQueue.begin());
- outInfo = NULL;
- notifyFillBufferDone(outHeader);
- outHeader = NULL;
+ if (inHeader->nFilledLen > 0 &&
+ vpx_codec_decode((vpx_codec_ctx_t *)mCtx,
+ inHeader->pBuffer + inHeader->nOffset,
+ inHeader->nFilledLen,
+ &mTimeStamps[mTimeStampIdx], 0)) {
+ ALOGE("on2 decoder failed to decode frame.");
+ notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+ return;
+ }
+ mTimeStampIdx = (mTimeStampIdx + 1) % kNumBuffers;
+
+ if (!outputBuffers(
+ EOSseen /* flushDecoder */, true /* display */, EOSseen, &portWillReset)) {
+ ALOGE("on2 decoder failed to output frame.");
+ notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+ return;
+ }
+ if (portWillReset) {
+ return;
}
inInfo->mOwnedByUs = false;
@@ -176,6 +253,30 @@
}
}
+void SoftVPX::onPortFlushCompleted(OMX_U32 portIndex) {
+ if (portIndex == kInputPortIndex) {
+ bool portWillReset = false;
+ if (!outputBuffers(
+ true /* flushDecoder */, false /* display */, false /* eos */, &portWillReset)) {
+ ALOGE("Failed to flush decoder.");
+ notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+ return;
+ }
+ mEOSStatus = INPUT_DATA_AVAILABLE;
+ }
+}
+
+void SoftVPX::onReset() {
+ bool portWillReset = false;
+ if (!outputBuffers(
+ true /* flushDecoder */, false /* display */, false /* eos */, &portWillReset)) {
+ ALOGW("Failed to flush decoder. Try to hard reset decoder");
+ destroyDecoder();
+ initDecoder();
+ }
+ mEOSStatus = INPUT_DATA_AVAILABLE;
+}
+
} // namespace android
android::SoftOMXComponent *createSoftOMXComponent(
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.h b/media/libstagefright/codecs/on2/dec/SoftVPX.h
index 8f68693..8ccbae2 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.h
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.h
@@ -38,6 +38,8 @@
virtual ~SoftVPX();
virtual void onQueueFilled(OMX_U32 portIndex);
+ virtual void onPortFlushCompleted(OMX_U32 portIndex);
+ virtual void onReset();
private:
enum {
@@ -49,11 +51,21 @@
MODE_VP9
} mMode;
- void *mCtx;
+ enum {
+ INPUT_DATA_AVAILABLE, // VPX component is ready to decode data.
+ INPUT_EOS_SEEN, // VPX component saw EOS and is flushing On2 decoder.
+ OUTPUT_FRAMES_FLUSHED // VPX component finished flushing On2 decoder.
+ } mEOSStatus;
+ void *mCtx;
+ bool mFrameParallelMode; // Frame parallel is only supported by VP9 decoder.
+ OMX_TICKS mTimeStamps[kNumBuffers];
+ uint8_t mTimeStampIdx;
vpx_image_t *mImg;
status_t initDecoder();
+ status_t destroyDecoder();
+ bool outputBuffers(bool flushDecoder, bool display, bool eos, bool *portWillReset);
DISALLOW_EVIL_CONSTRUCTORS(SoftVPX);
};
diff --git a/media/libstagefright/codecs/opus/dec/SoftOpus.cpp b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
index b8084ae..6322dc2 100644
--- a/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
+++ b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
@@ -345,9 +345,15 @@
}
uint8_t channel_mapping[kMaxChannels] = {0};
- memcpy(&channel_mapping,
- kDefaultOpusChannelLayout,
- kMaxChannelsWithDefaultLayout);
+ if (mHeader->channels <= kMaxChannelsWithDefaultLayout) {
+ memcpy(&channel_mapping,
+ kDefaultOpusChannelLayout,
+ kMaxChannelsWithDefaultLayout);
+ } else {
+ memcpy(&channel_mapping,
+ mHeader->stream_map,
+ mHeader->channels);
+ }
int status = OPUS_INVALID_STATE;
mDecoder = opus_multistream_decoder_create(kRate,
diff --git a/media/libstagefright/colorconversion/SoftwareRenderer.cpp b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
index 4e75250..21da707 100644
--- a/media/libstagefright/colorconversion/SoftwareRenderer.cpp
+++ b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
@@ -98,33 +98,49 @@
mCropWidth = mCropRight - mCropLeft + 1;
mCropHeight = mCropBottom - mCropTop + 1;
- int halFormat;
- size_t bufWidth, bufHeight;
+ // by default convert everything to RGB565
+ int halFormat = HAL_PIXEL_FORMAT_RGB_565;
+ size_t bufWidth = mCropWidth;
+ size_t bufHeight = mCropHeight;
- switch (mColorFormat) {
- case OMX_COLOR_FormatYUV420Planar:
- case OMX_TI_COLOR_FormatYUV420PackedSemiPlanar:
- case OMX_COLOR_FormatYUV420SemiPlanar:
- {
- if (!runningInEmulator()) {
+ // hardware has YUV12 and RGBA8888 support, so convert known formats
+ if (!runningInEmulator()) {
+ switch (mColorFormat) {
+ case OMX_COLOR_FormatYUV420Planar:
+ case OMX_COLOR_FormatYUV420SemiPlanar:
+ case OMX_TI_COLOR_FormatYUV420PackedSemiPlanar:
+ {
halFormat = HAL_PIXEL_FORMAT_YV12;
bufWidth = (mCropWidth + 1) & ~1;
bufHeight = (mCropHeight + 1) & ~1;
break;
}
-
- // fall through.
+ case OMX_COLOR_Format24bitRGB888:
+ {
+ halFormat = HAL_PIXEL_FORMAT_RGB_888;
+ bufWidth = (mCropWidth + 1) & ~1;
+ bufHeight = (mCropHeight + 1) & ~1;
+ break;
+ }
+ case OMX_COLOR_Format32bitARGB8888:
+ case OMX_COLOR_Format32BitRGBA8888:
+ {
+ halFormat = HAL_PIXEL_FORMAT_RGBA_8888;
+ bufWidth = (mCropWidth + 1) & ~1;
+ bufHeight = (mCropHeight + 1) & ~1;
+ break;
+ }
+ default:
+ {
+ break;
+ }
}
+ }
- default:
- halFormat = HAL_PIXEL_FORMAT_RGB_565;
- bufWidth = mCropWidth;
- bufHeight = mCropHeight;
-
- mConverter = new ColorConverter(
- mColorFormat, OMX_COLOR_Format16bitRGB565);
- CHECK(mConverter->isValid());
- break;
+ if (halFormat == HAL_PIXEL_FORMAT_RGB_565) {
+ mConverter = new ColorConverter(
+ mColorFormat, OMX_COLOR_Format16bitRGB565);
+ CHECK(mConverter->isValid());
}
CHECK(mNativeWindow != NULL);
@@ -201,6 +217,8 @@
CHECK_EQ(0, mapper.lock(
buf->handle, GRALLOC_USAGE_SW_WRITE_OFTEN, bounds, &dst));
+ // TODO move the other conversions also into ColorConverter, and
+ // fix cropping issues (when mCropLeft/Top != 0 or mWidth != mCropWidth)
if (mConverter) {
mConverter->convert(
data,
@@ -211,7 +229,8 @@
0, 0, mCropWidth - 1, mCropHeight - 1);
} else if (mColorFormat == OMX_COLOR_FormatYUV420Planar) {
const uint8_t *src_y = (const uint8_t *)data;
- const uint8_t *src_u = (const uint8_t *)data + mWidth * mHeight;
+ const uint8_t *src_u =
+ (const uint8_t *)data + mWidth * mHeight;
const uint8_t *src_v = src_u + (mWidth / 2 * mHeight / 2);
uint8_t *dst_y = (uint8_t *)dst;
@@ -239,11 +258,9 @@
}
} else if (mColorFormat == OMX_TI_COLOR_FormatYUV420PackedSemiPlanar
|| mColorFormat == OMX_COLOR_FormatYUV420SemiPlanar) {
- const uint8_t *src_y =
- (const uint8_t *)data;
-
- const uint8_t *src_uv =
- (const uint8_t *)data + mWidth * (mHeight - mCropTop / 2);
+ const uint8_t *src_y = (const uint8_t *)data;
+ const uint8_t *src_uv = (const uint8_t *)data
+ + mWidth * (mHeight - mCropTop / 2);
uint8_t *dst_y = (uint8_t *)dst;
@@ -271,6 +288,38 @@
dst_u += dst_c_stride;
dst_v += dst_c_stride;
}
+ } else if (mColorFormat == OMX_COLOR_Format24bitRGB888) {
+ uint8_t* srcPtr = (uint8_t*)data;
+ uint8_t* dstPtr = (uint8_t*)dst;
+
+ for (size_t y = 0; y < (size_t)mCropHeight; ++y) {
+ memcpy(dstPtr, srcPtr, mCropWidth * 3);
+ srcPtr += mWidth * 3;
+ dstPtr += buf->stride * 3;
+ }
+ } else if (mColorFormat == OMX_COLOR_Format32bitARGB8888) {
+ uint8_t *srcPtr, *dstPtr;
+
+ for (size_t y = 0; y < (size_t)mCropHeight; ++y) {
+ srcPtr = (uint8_t*)data + mWidth * 4 * y;
+ dstPtr = (uint8_t*)dst + buf->stride * 4 * y;
+ for (size_t x = 0; x < (size_t)mCropWidth; ++x) {
+ uint8_t a = *srcPtr++;
+ for (size_t i = 0; i < 3; ++i) { // copy RGB
+ *dstPtr++ = *srcPtr++;
+ }
+ *dstPtr++ = a; // alpha last (ARGB to RGBA)
+ }
+ }
+ } else if (mColorFormat == OMX_COLOR_Format32BitRGBA8888) {
+ uint8_t* srcPtr = (uint8_t*)data;
+ uint8_t* dstPtr = (uint8_t*)dst;
+
+ for (size_t y = 0; y < (size_t)mCropHeight; ++y) {
+ memcpy(dstPtr, srcPtr, mCropWidth * 4);
+ srcPtr += mWidth * 4;
+ dstPtr += buf->stride * 4;
+ }
} else {
LOG_ALWAYS_FATAL("bad color format %#x", mColorFormat);
}
diff --git a/media/libstagefright/filters/Android.mk b/media/libstagefright/filters/Android.mk
new file mode 100644
index 0000000..36ab444
--- /dev/null
+++ b/media/libstagefright/filters/Android.mk
@@ -0,0 +1,27 @@
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ ColorConvert.cpp \
+ GraphicBufferListener.cpp \
+ IntrinsicBlurFilter.cpp \
+ MediaFilter.cpp \
+ RSFilter.cpp \
+ SaturationFilter.cpp \
+ saturationARGB.rs \
+ SimpleFilter.cpp \
+ ZeroFilter.cpp
+
+LOCAL_C_INCLUDES := \
+ $(TOP)/frameworks/native/include/media/openmax \
+ $(TOP)/frameworks/rs/cpp \
+ $(TOP)/frameworks/rs \
+
+intermediates := $(call intermediates-dir-for,STATIC_LIBRARIES,libRS,TARGET,)
+LOCAL_C_INCLUDES += $(intermediates)
+
+LOCAL_CFLAGS += -Wno-multichar
+
+LOCAL_MODULE:= libstagefright_mediafilter
+
+include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libstagefright/filters/ColorConvert.cpp b/media/libstagefright/filters/ColorConvert.cpp
new file mode 100644
index 0000000..a5039f9
--- /dev/null
+++ b/media/libstagefright/filters/ColorConvert.cpp
@@ -0,0 +1,111 @@
+/*
+ * 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.
+ */
+
+#include "ColorConvert.h"
+
+#ifndef max
+#define max(a,b) ((a) > (b) ? (a) : (b))
+#endif
+#ifndef min
+#define min(a,b) ((a) < (b) ? (a) : (b))
+#endif
+
+namespace android {
+
+void YUVToRGB(
+ int32_t y, int32_t u, int32_t v,
+ int32_t* r, int32_t* g, int32_t* b) {
+ y -= 16;
+ u -= 128;
+ v -= 128;
+
+ *b = 1192 * y + 2066 * u;
+ *g = 1192 * y - 833 * v - 400 * u;
+ *r = 1192 * y + 1634 * v;
+
+ *r = min(262143, max(0, *r));
+ *g = min(262143, max(0, *g));
+ *b = min(262143, max(0, *b));
+
+ *r >>= 10;
+ *g >>= 10;
+ *b >>= 10;
+}
+
+void convertYUV420spToARGB(
+ uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+ uint8_t *dest) {
+ const int32_t bytes_per_pixel = 2;
+
+ for (int32_t i = 0; i < height; i++) {
+ for (int32_t j = 0; j < width; j++) {
+ int32_t y = *(pY + i * width + j);
+ int32_t u = *(pUV + (i/2) * width + bytes_per_pixel * (j/2));
+ int32_t v = *(pUV + (i/2) * width + bytes_per_pixel * (j/2) + 1);
+
+ int32_t r, g, b;
+ YUVToRGB(y, u, v, &r, &g, &b);
+
+ *dest++ = 0xFF;
+ *dest++ = r;
+ *dest++ = g;
+ *dest++ = b;
+ }
+ }
+}
+
+void convertYUV420spToRGB888(
+ uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+ uint8_t *dest) {
+ const int32_t bytes_per_pixel = 2;
+
+ for (int32_t i = 0; i < height; i++) {
+ for (int32_t j = 0; j < width; j++) {
+ int32_t y = *(pY + i * width + j);
+ int32_t u = *(pUV + (i/2) * width + bytes_per_pixel * (j/2));
+ int32_t v = *(pUV + (i/2) * width + bytes_per_pixel * (j/2) + 1);
+
+ int32_t r, g, b;
+ YUVToRGB(y, u, v, &r, &g, &b);
+
+ *dest++ = r;
+ *dest++ = g;
+ *dest++ = b;
+ }
+ }
+}
+
+// HACK - not even slightly optimized
+// TODO: remove when RGBA support is added to SoftwareRenderer
+void convertRGBAToARGB(
+ uint8_t *src, int32_t width, int32_t height, uint32_t stride,
+ uint8_t *dest) {
+ for (size_t i = 0; i < height; ++i) {
+ for (size_t j = 0; j < width; ++j) {
+ uint8_t r = *src++;
+ uint8_t g = *src++;
+ uint8_t b = *src++;
+ uint8_t a = *src++;
+ *dest++ = a;
+ *dest++ = r;
+ *dest++ = g;
+ *dest++ = b;
+ }
+ src += (stride - width) * 4;
+ }
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/ColorConvert.h b/media/libstagefright/filters/ColorConvert.h
new file mode 100644
index 0000000..13faa02
--- /dev/null
+++ b/media/libstagefright/filters/ColorConvert.h
@@ -0,0 +1,43 @@
+/*
+ * 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 COLOR_CONVERT_H_
+#define COLOR_CONVERT_H_
+
+#include <inttypes.h>
+
+namespace android {
+
+void YUVToRGB(
+ int32_t y, int32_t u, int32_t v,
+ int32_t* r, int32_t* g, int32_t* b);
+
+void convertYUV420spToARGB(
+ uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+ uint8_t *dest);
+
+void convertYUV420spToRGB888(
+ uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+ uint8_t *dest);
+
+// TODO: remove when RGBA support is added to SoftwareRenderer
+void convertRGBAToARGB(
+ uint8_t *src, int32_t width, int32_t height, uint32_t stride,
+ uint8_t *dest);
+
+} // namespace android
+
+#endif // COLOR_CONVERT_H_
diff --git a/media/libstagefright/filters/GraphicBufferListener.cpp b/media/libstagefright/filters/GraphicBufferListener.cpp
new file mode 100644
index 0000000..66374ba
--- /dev/null
+++ b/media/libstagefright/filters/GraphicBufferListener.cpp
@@ -0,0 +1,154 @@
+/*
+ * 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 "GraphicBufferListener"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaErrors.h>
+
+#include <gui/BufferItem.h>
+
+#include "GraphicBufferListener.h"
+
+namespace android {
+
+status_t GraphicBufferListener::init(
+ const sp<AMessage> ¬ify,
+ size_t bufferWidth, size_t bufferHeight, size_t bufferCount) {
+ mNotify = notify;
+
+ String8 name("GraphicBufferListener");
+ BufferQueue::createBufferQueue(&mProducer, &mConsumer);
+ mConsumer->setConsumerName(name);
+ mConsumer->setDefaultBufferSize(bufferWidth, bufferHeight);
+ mConsumer->setConsumerUsageBits(GRALLOC_USAGE_SW_READ_OFTEN);
+
+ status_t err = mConsumer->setMaxAcquiredBufferCount(bufferCount);
+ if (err != NO_ERROR) {
+ ALOGE("Unable to set BQ max acquired buffer count to %u: %d",
+ bufferCount, err);
+ return err;
+ }
+
+ wp<BufferQueue::ConsumerListener> listener =
+ static_cast<BufferQueue::ConsumerListener*>(this);
+ sp<BufferQueue::ProxyConsumerListener> proxy =
+ new BufferQueue::ProxyConsumerListener(listener);
+
+ err = mConsumer->consumerConnect(proxy, false);
+ if (err != NO_ERROR) {
+ ALOGE("Error connecting to BufferQueue: %s (%d)",
+ strerror(-err), err);
+ return err;
+ }
+
+ ALOGV("init() successful.");
+
+ return OK;
+}
+
+void GraphicBufferListener::onFrameAvailable(const BufferItem& /* item */) {
+ ALOGV("onFrameAvailable() called");
+
+ {
+ Mutex::Autolock autoLock(mMutex);
+ mNumFramesAvailable++;
+ }
+
+ sp<AMessage> notify = mNotify->dup();
+ mNotify->setWhat(kWhatFrameAvailable);
+ mNotify->post();
+}
+
+void GraphicBufferListener::onBuffersReleased() {
+ ALOGV("onBuffersReleased() called");
+ // nothing to do
+}
+
+void GraphicBufferListener::onSidebandStreamChanged() {
+ ALOGW("GraphicBufferListener cannot consume sideband streams.");
+ // nothing to do
+}
+
+BufferItem GraphicBufferListener::getBufferItem() {
+ BufferItem item;
+
+ {
+ Mutex::Autolock autoLock(mMutex);
+ if (mNumFramesAvailable <= 0) {
+ ALOGE("getBuffer() called with no frames available");
+ return item;
+ }
+ mNumFramesAvailable--;
+ }
+
+ status_t err = mConsumer->acquireBuffer(&item, 0);
+ if (err == BufferQueue::NO_BUFFER_AVAILABLE) {
+ // shouldn't happen, since we track num frames available
+ ALOGE("frame was not available");
+ item.mBuf = -1;
+ return item;
+ } else if (err != OK) {
+ ALOGE("acquireBuffer returned err=%d", err);
+ item.mBuf = -1;
+ return item;
+ }
+
+ // Wait for it to become available.
+ err = item.mFence->waitForever("GraphicBufferListener::getBufferItem");
+ if (err != OK) {
+ ALOGW("failed to wait for buffer fence: %d", err);
+ // keep going
+ }
+
+ // If this is the first time we're seeing this buffer, add it to our
+ // slot table.
+ if (item.mGraphicBuffer != NULL) {
+ ALOGV("setting mBufferSlot %d", item.mBuf);
+ mBufferSlot[item.mBuf] = item.mGraphicBuffer;
+ }
+
+ return item;
+}
+
+sp<GraphicBuffer> GraphicBufferListener::getBuffer(BufferItem item) {
+ sp<GraphicBuffer> buf;
+ if (item.mBuf < 0 || item.mBuf >= BufferQueue::NUM_BUFFER_SLOTS) {
+ ALOGE("getBuffer() received invalid BufferItem: mBuf==%d", item.mBuf);
+ return buf;
+ }
+
+ buf = mBufferSlot[item.mBuf];
+ CHECK(buf.get() != NULL);
+
+ return buf;
+}
+
+status_t GraphicBufferListener::releaseBuffer(BufferItem item) {
+ if (item.mBuf < 0 || item.mBuf >= BufferQueue::NUM_BUFFER_SLOTS) {
+ ALOGE("getBuffer() received invalid BufferItem: mBuf==%d", item.mBuf);
+ return ERROR_OUT_OF_RANGE;
+ }
+
+ mConsumer->releaseBuffer(item.mBuf, item.mFrameNumber,
+ EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE);
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/GraphicBufferListener.h b/media/libstagefright/filters/GraphicBufferListener.h
new file mode 100644
index 0000000..586bf65
--- /dev/null
+++ b/media/libstagefright/filters/GraphicBufferListener.h
@@ -0,0 +1,70 @@
+/*
+ * 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 GRAPHIC_BUFFER_LISTENER_H_
+#define GRAPHIC_BUFFER_LISTENER_H_
+
+#include <gui/BufferQueue.h>
+
+namespace android {
+
+struct AMessage;
+
+struct GraphicBufferListener : public BufferQueue::ConsumerListener {
+public:
+ GraphicBufferListener() {};
+
+ status_t init(
+ const sp<AMessage> ¬ify,
+ size_t bufferWidth, size_t bufferHeight, size_t bufferCount);
+
+ virtual void onFrameAvailable(const BufferItem& item);
+ virtual void onBuffersReleased();
+ virtual void onSidebandStreamChanged();
+
+ // Returns the handle to the producer side of the BufferQueue. Buffers
+ // queued on this will be received by GraphicBufferListener.
+ sp<IGraphicBufferProducer> getIGraphicBufferProducer() const {
+ return mProducer;
+ }
+
+ BufferItem getBufferItem();
+ sp<GraphicBuffer> getBuffer(BufferItem item);
+ status_t releaseBuffer(BufferItem item);
+
+ enum {
+ kWhatFrameAvailable = 'frav',
+ };
+
+private:
+ sp<AMessage> mNotify;
+ size_t mNumFramesAvailable;
+
+ mutable Mutex mMutex;
+
+ // Our BufferQueue interfaces. mProducer is passed to the producer through
+ // getIGraphicBufferProducer, and mConsumer is used internally to retrieve
+ // the buffers queued by the producer.
+ sp<IGraphicBufferProducer> mProducer;
+ sp<IGraphicBufferConsumer> mConsumer;
+
+ // Cache of GraphicBuffers from the buffer queue.
+ sp<GraphicBuffer> mBufferSlot[BufferQueue::NUM_BUFFER_SLOTS];
+};
+
+} // namespace android
+
+#endif // GRAPHIC_BUFFER_LISTENER_H
diff --git a/media/libstagefright/filters/IntrinsicBlurFilter.cpp b/media/libstagefright/filters/IntrinsicBlurFilter.cpp
new file mode 100644
index 0000000..cbcf699
--- /dev/null
+++ b/media/libstagefright/filters/IntrinsicBlurFilter.cpp
@@ -0,0 +1,99 @@
+/*
+ * 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 "IntrinsicBlurFilter"
+
+#include <utils/Log.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "IntrinsicBlurFilter.h"
+
+namespace android {
+
+status_t IntrinsicBlurFilter::configure(const sp<AMessage> &msg) {
+ status_t err = SimpleFilter::configure(msg);
+ if (err != OK) {
+ return err;
+ }
+
+ if (!msg->findString("cacheDir", &mCacheDir)) {
+ ALOGE("Failed to find cache directory in config message.");
+ return NAME_NOT_FOUND;
+ }
+
+ return OK;
+}
+
+status_t IntrinsicBlurFilter::start() {
+ // TODO: use a single RS context object for entire application
+ mRS = new RSC::RS();
+
+ if (!mRS->init(mCacheDir.c_str())) {
+ ALOGE("Failed to initialize RenderScript context.");
+ return NO_INIT;
+ }
+
+ // 32-bit elements for ARGB8888
+ RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS);
+
+ RSC::Type::Builder tb(mRS, e);
+ tb.setX(mWidth);
+ tb.setY(mHeight);
+ RSC::sp<const RSC::Type> t = tb.create();
+
+ mAllocIn = RSC::Allocation::createTyped(mRS, t);
+ mAllocOut = RSC::Allocation::createTyped(mRS, t);
+
+ mBlur = RSC::ScriptIntrinsicBlur::create(mRS, e);
+ mBlur->setRadius(mBlurRadius);
+ mBlur->setInput(mAllocIn);
+
+ return OK;
+}
+
+void IntrinsicBlurFilter::reset() {
+ mBlur.clear();
+ mAllocOut.clear();
+ mAllocIn.clear();
+ mRS.clear();
+}
+
+status_t IntrinsicBlurFilter::setParameters(const sp<AMessage> &msg) {
+ sp<AMessage> params;
+ CHECK(msg->findMessage("params", ¶ms));
+
+ float blurRadius;
+ if (params->findFloat("blur-radius", &blurRadius)) {
+ mBlurRadius = blurRadius;
+ }
+
+ return OK;
+}
+
+status_t IntrinsicBlurFilter::processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+ mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data());
+ mBlur->forEach(mAllocOut);
+ mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data());
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/IntrinsicBlurFilter.h b/media/libstagefright/filters/IntrinsicBlurFilter.h
new file mode 100644
index 0000000..4707ab7
--- /dev/null
+++ b/media/libstagefright/filters/IntrinsicBlurFilter.h
@@ -0,0 +1,50 @@
+/*
+ * 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 INTRINSIC_BLUR_FILTER_H_
+#define INTRINSIC_BLUR_FILTER_H_
+
+#include "RenderScript.h"
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct IntrinsicBlurFilter : public SimpleFilter {
+public:
+ IntrinsicBlurFilter() : mBlurRadius(1.f) {};
+
+ virtual status_t configure(const sp<AMessage> &msg);
+ virtual status_t start();
+ virtual void reset();
+ virtual status_t setParameters(const sp<AMessage> &msg);
+ virtual status_t processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+ virtual ~IntrinsicBlurFilter() {};
+
+private:
+ AString mCacheDir;
+ RSC::sp<RSC::RS> mRS;
+ RSC::sp<RSC::Allocation> mAllocIn;
+ RSC::sp<RSC::Allocation> mAllocOut;
+ RSC::sp<RSC::ScriptIntrinsicBlur> mBlur;
+ float mBlurRadius;
+};
+
+} // namespace android
+
+#endif // INTRINSIC_BLUR_FILTER_H_
diff --git a/media/libstagefright/filters/MediaFilter.cpp b/media/libstagefright/filters/MediaFilter.cpp
new file mode 100644
index 0000000..0a09575
--- /dev/null
+++ b/media/libstagefright/filters/MediaFilter.cpp
@@ -0,0 +1,818 @@
+/*
+ * 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 "MediaFilter"
+
+#include <inttypes.h>
+#include <utils/Trace.h>
+
+#include <binder/MemoryDealer.h>
+
+#include <media/stagefright/BufferProducerWrapper.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaFilter.h>
+
+#include <gui/BufferItem.h>
+
+#include "ColorConvert.h"
+#include "GraphicBufferListener.h"
+#include "IntrinsicBlurFilter.h"
+#include "RSFilter.h"
+#include "SaturationFilter.h"
+#include "ZeroFilter.h"
+
+namespace android {
+
+// parameter: number of input and output buffers
+static const size_t kBufferCountActual = 4;
+
+MediaFilter::MediaFilter()
+ : mState(UNINITIALIZED),
+ mGeneration(0),
+ mGraphicBufferListener(NULL) {
+}
+
+MediaFilter::~MediaFilter() {
+}
+
+//////////////////// PUBLIC FUNCTIONS //////////////////////////////////////////
+
+void MediaFilter::setNotificationMessage(const sp<AMessage> &msg) {
+ mNotify = msg;
+}
+
+void MediaFilter::initiateAllocateComponent(const sp<AMessage> &msg) {
+ msg->setWhat(kWhatAllocateComponent);
+ msg->setTarget(this);
+ msg->post();
+}
+
+void MediaFilter::initiateConfigureComponent(const sp<AMessage> &msg) {
+ msg->setWhat(kWhatConfigureComponent);
+ msg->setTarget(this);
+ msg->post();
+}
+
+void MediaFilter::initiateCreateInputSurface() {
+ (new AMessage(kWhatCreateInputSurface, this))->post();
+}
+
+void MediaFilter::initiateStart() {
+ (new AMessage(kWhatStart, this))->post();
+}
+
+void MediaFilter::initiateShutdown(bool keepComponentAllocated) {
+ sp<AMessage> msg = new AMessage(kWhatShutdown, this);
+ msg->setInt32("keepComponentAllocated", keepComponentAllocated);
+ msg->post();
+}
+
+void MediaFilter::signalFlush() {
+ (new AMessage(kWhatFlush, this))->post();
+}
+
+void MediaFilter::signalResume() {
+ (new AMessage(kWhatResume, this))->post();
+}
+
+// nothing to do
+void MediaFilter::signalRequestIDRFrame() {
+ return;
+}
+
+void MediaFilter::signalSetParameters(const sp<AMessage> ¶ms) {
+ sp<AMessage> msg = new AMessage(kWhatSetParameters, this);
+ msg->setMessage("params", params);
+ msg->post();
+}
+
+void MediaFilter::signalEndOfInputStream() {
+ (new AMessage(kWhatSignalEndOfInputStream, this))->post();
+}
+
+void MediaFilter::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatAllocateComponent:
+ {
+ onAllocateComponent(msg);
+ break;
+ }
+ case kWhatConfigureComponent:
+ {
+ onConfigureComponent(msg);
+ break;
+ }
+ case kWhatStart:
+ {
+ onStart();
+ break;
+ }
+ case kWhatProcessBuffers:
+ {
+ processBuffers();
+ break;
+ }
+ case kWhatInputBufferFilled:
+ {
+ onInputBufferFilled(msg);
+ break;
+ }
+ case kWhatOutputBufferDrained:
+ {
+ onOutputBufferDrained(msg);
+ break;
+ }
+ case kWhatShutdown:
+ {
+ onShutdown(msg);
+ break;
+ }
+ case kWhatFlush:
+ {
+ onFlush();
+ break;
+ }
+ case kWhatResume:
+ {
+ // nothing to do
+ break;
+ }
+ case kWhatSetParameters:
+ {
+ onSetParameters(msg);
+ break;
+ }
+ case kWhatCreateInputSurface:
+ {
+ onCreateInputSurface();
+ break;
+ }
+ case GraphicBufferListener::kWhatFrameAvailable:
+ {
+ onInputFrameAvailable();
+ break;
+ }
+ case kWhatSignalEndOfInputStream:
+ {
+ onSignalEndOfInputStream();
+ break;
+ }
+ default:
+ {
+ ALOGE("Message not handled:\n%s", msg->debugString().c_str());
+ break;
+ }
+ }
+}
+
+//////////////////// PORT DESCRIPTION //////////////////////////////////////////
+
+MediaFilter::PortDescription::PortDescription() {
+}
+
+void MediaFilter::PortDescription::addBuffer(
+ IOMX::buffer_id id, const sp<ABuffer> &buffer) {
+ mBufferIDs.push_back(id);
+ mBuffers.push_back(buffer);
+}
+
+size_t MediaFilter::PortDescription::countBuffers() {
+ return mBufferIDs.size();
+}
+
+IOMX::buffer_id MediaFilter::PortDescription::bufferIDAt(size_t index) const {
+ return mBufferIDs.itemAt(index);
+}
+
+sp<ABuffer> MediaFilter::PortDescription::bufferAt(size_t index) const {
+ return mBuffers.itemAt(index);
+}
+
+//////////////////// HELPER FUNCTIONS //////////////////////////////////////////
+
+void MediaFilter::signalProcessBuffers() {
+ (new AMessage(kWhatProcessBuffers, this))->post();
+}
+
+void MediaFilter::signalError(status_t error) {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatError);
+ notify->setInt32("err", error);
+ notify->post();
+}
+
+status_t MediaFilter::allocateBuffersOnPort(OMX_U32 portIndex) {
+ CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
+ const bool isInput = portIndex == kPortIndexInput;
+ const size_t bufferSize = isInput ? mMaxInputSize : mMaxOutputSize;
+
+ CHECK(mDealer[portIndex] == NULL);
+ CHECK(mBuffers[portIndex].isEmpty());
+
+ ALOGV("Allocating %zu buffers of size %zu on %s port",
+ kBufferCountActual, bufferSize,
+ isInput ? "input" : "output");
+
+ size_t totalSize = kBufferCountActual * bufferSize;
+
+ mDealer[portIndex] = new MemoryDealer(totalSize, "MediaFilter");
+
+ for (size_t i = 0; i < kBufferCountActual; ++i) {
+ sp<IMemory> mem = mDealer[portIndex]->allocate(bufferSize);
+ CHECK(mem.get() != NULL);
+
+ BufferInfo info;
+ info.mStatus = BufferInfo::OWNED_BY_US;
+ info.mBufferID = i;
+ info.mGeneration = mGeneration;
+ info.mOutputFlags = 0;
+ info.mData = new ABuffer(mem->pointer(), bufferSize);
+ info.mData->meta()->setInt64("timeUs", 0);
+
+ mBuffers[portIndex].push_back(info);
+
+ if (!isInput) {
+ mAvailableOutputBuffers.push(
+ &mBuffers[portIndex].editItemAt(i));
+ }
+ }
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatBuffersAllocated);
+
+ notify->setInt32("portIndex", portIndex);
+
+ sp<PortDescription> desc = new PortDescription;
+
+ for (size_t i = 0; i < mBuffers[portIndex].size(); ++i) {
+ const BufferInfo &info = mBuffers[portIndex][i];
+
+ desc->addBuffer(info.mBufferID, info.mData);
+ }
+
+ notify->setObject("portDesc", desc);
+ notify->post();
+
+ return OK;
+}
+
+MediaFilter::BufferInfo* MediaFilter::findBufferByID(
+ uint32_t portIndex, IOMX::buffer_id bufferID,
+ ssize_t *index) {
+ for (size_t i = 0; i < mBuffers[portIndex].size(); ++i) {
+ BufferInfo *info = &mBuffers[portIndex].editItemAt(i);
+
+ if (info->mBufferID == bufferID) {
+ if (index != NULL) {
+ *index = i;
+ }
+ return info;
+ }
+ }
+
+ TRESPASS();
+
+ return NULL;
+}
+
+void MediaFilter::postFillThisBuffer(BufferInfo *info) {
+ ALOGV("postFillThisBuffer on buffer %d", info->mBufferID);
+ if (mPortEOS[kPortIndexInput]) {
+ return;
+ }
+
+ CHECK_EQ((int)info->mStatus, (int)BufferInfo::OWNED_BY_US);
+
+ info->mGeneration = mGeneration;
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatFillThisBuffer);
+ notify->setInt32("buffer-id", info->mBufferID);
+
+ info->mData->meta()->clear();
+ notify->setBuffer("buffer", info->mData);
+
+ sp<AMessage> reply = new AMessage(kWhatInputBufferFilled, this);
+ reply->setInt32("buffer-id", info->mBufferID);
+
+ notify->setMessage("reply", reply);
+
+ info->mStatus = BufferInfo::OWNED_BY_UPSTREAM;
+ notify->post();
+}
+
+void MediaFilter::postDrainThisBuffer(BufferInfo *info) {
+ CHECK_EQ((int)info->mStatus, (int)BufferInfo::OWNED_BY_US);
+
+ info->mGeneration = mGeneration;
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatDrainThisBuffer);
+ notify->setInt32("buffer-id", info->mBufferID);
+ notify->setInt32("flags", info->mOutputFlags);
+ notify->setBuffer("buffer", info->mData);
+
+ sp<AMessage> reply = new AMessage(kWhatOutputBufferDrained, this);
+ reply->setInt32("buffer-id", info->mBufferID);
+
+ notify->setMessage("reply", reply);
+
+ notify->post();
+
+ info->mStatus = BufferInfo::OWNED_BY_UPSTREAM;
+}
+
+void MediaFilter::postEOS() {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatEOS);
+ notify->setInt32("err", ERROR_END_OF_STREAM);
+ notify->post();
+
+ ALOGV("Sent kWhatEOS.");
+}
+
+void MediaFilter::sendFormatChange() {
+ sp<AMessage> notify = mNotify->dup();
+
+ notify->setInt32("what", kWhatOutputFormatChanged);
+
+ AString mime;
+ CHECK(mOutputFormat->findString("mime", &mime));
+ notify->setString("mime", mime.c_str());
+
+ notify->setInt32("stride", mStride);
+ notify->setInt32("slice-height", mSliceHeight);
+ notify->setInt32("color-format", mColorFormatOut);
+ notify->setRect("crop", 0, 0, mStride - 1, mSliceHeight - 1);
+ notify->setInt32("width", mWidth);
+ notify->setInt32("height", mHeight);
+
+ notify->post();
+}
+
+void MediaFilter::requestFillEmptyInput() {
+ if (mPortEOS[kPortIndexInput]) {
+ return;
+ }
+
+ for (size_t i = 0; i < mBuffers[kPortIndexInput].size(); ++i) {
+ BufferInfo *info = &mBuffers[kPortIndexInput].editItemAt(i);
+
+ if (info->mStatus == BufferInfo::OWNED_BY_US) {
+ postFillThisBuffer(info);
+ }
+ }
+}
+
+void MediaFilter::processBuffers() {
+ if (mAvailableInputBuffers.empty() || mAvailableOutputBuffers.empty()) {
+ ALOGV("Skipping process (buffers unavailable)");
+ return;
+ }
+
+ if (mPortEOS[kPortIndexOutput]) {
+ // TODO notify caller of queueInput error when it is supported
+ // in MediaCodec
+ ALOGW("Tried to process a buffer after EOS.");
+ return;
+ }
+
+ BufferInfo *inputInfo = mAvailableInputBuffers[0];
+ mAvailableInputBuffers.removeAt(0);
+ BufferInfo *outputInfo = mAvailableOutputBuffers[0];
+ mAvailableOutputBuffers.removeAt(0);
+
+ status_t err;
+ err = mFilter->processBuffers(inputInfo->mData, outputInfo->mData);
+ if (err != (status_t)OK) {
+ outputInfo->mData->meta()->setInt32("err", err);
+ }
+
+ int64_t timeUs;
+ CHECK(inputInfo->mData->meta()->findInt64("timeUs", &timeUs));
+ outputInfo->mData->meta()->setInt64("timeUs", timeUs);
+ outputInfo->mOutputFlags = 0;
+ int32_t eos = 0;
+ if (inputInfo->mData->meta()->findInt32("eos", &eos) && eos != 0) {
+ outputInfo->mOutputFlags |= OMX_BUFFERFLAG_EOS;
+ mPortEOS[kPortIndexOutput] = true;
+ outputInfo->mData->meta()->setInt32("eos", eos);
+ postEOS();
+ ALOGV("Output stream saw EOS.");
+ }
+
+ ALOGV("Processed input buffer %u [%zu], output buffer %u [%zu]",
+ inputInfo->mBufferID, inputInfo->mData->size(),
+ outputInfo->mBufferID, outputInfo->mData->size());
+
+ if (mGraphicBufferListener != NULL) {
+ delete inputInfo;
+ } else {
+ postFillThisBuffer(inputInfo);
+ }
+ postDrainThisBuffer(outputInfo);
+
+ // prevent any corner case where buffers could get stuck in queue
+ signalProcessBuffers();
+}
+
+void MediaFilter::onAllocateComponent(const sp<AMessage> &msg) {
+ CHECK_EQ(mState, UNINITIALIZED);
+
+ CHECK(msg->findString("componentName", &mComponentName));
+ const char* name = mComponentName.c_str();
+ if (!strcasecmp(name, "android.filter.zerofilter")) {
+ mFilter = new ZeroFilter;
+ } else if (!strcasecmp(name, "android.filter.saturation")) {
+ mFilter = new SaturationFilter;
+ } else if (!strcasecmp(name, "android.filter.intrinsicblur")) {
+ mFilter = new IntrinsicBlurFilter;
+ } else if (!strcasecmp(name, "android.filter.RenderScript")) {
+ mFilter = new RSFilter;
+ } else {
+ ALOGE("Unrecognized filter name: %s", name);
+ signalError(NAME_NOT_FOUND);
+ return;
+ }
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatComponentAllocated);
+ // HACK - need "OMX.google" to use MediaCodec's software renderer
+ notify->setString("componentName", "OMX.google.MediaFilter");
+ notify->post();
+ mState = INITIALIZED;
+ ALOGV("Handled kWhatAllocateComponent.");
+}
+
+void MediaFilter::onConfigureComponent(const sp<AMessage> &msg) {
+ // TODO: generalize to allow audio filters as well as video
+
+ CHECK_EQ(mState, INITIALIZED);
+
+ // get params - at least mime, width & height
+ AString mime;
+ CHECK(msg->findString("mime", &mime));
+ if (strcasecmp(mime.c_str(), MEDIA_MIMETYPE_VIDEO_RAW)) {
+ ALOGE("Bad mime: %s", mime.c_str());
+ signalError(BAD_VALUE);
+ return;
+ }
+
+ CHECK(msg->findInt32("width", &mWidth));
+ CHECK(msg->findInt32("height", &mHeight));
+ if (!msg->findInt32("stride", &mStride)) {
+ mStride = mWidth;
+ }
+ if (!msg->findInt32("slice-height", &mSliceHeight)) {
+ mSliceHeight = mHeight;
+ }
+
+ mMaxInputSize = mWidth * mHeight * 4; // room for ARGB8888
+ int32_t maxInputSize;
+ if (msg->findInt32("max-input-size", &maxInputSize)
+ && (size_t)maxInputSize > mMaxInputSize) {
+ mMaxInputSize = maxInputSize;
+ }
+
+ if (!msg->findInt32("color-format", &mColorFormatIn)) {
+ // default to OMX_COLOR_Format32bitARGB8888
+ mColorFormatIn = OMX_COLOR_Format32bitARGB8888;
+ msg->setInt32("color-format", mColorFormatIn);
+ }
+ mColorFormatOut = mColorFormatIn;
+
+ mMaxOutputSize = mWidth * mHeight * 4; // room for ARGB8888
+
+ AString cacheDir;
+ if (!msg->findString("cacheDir", &cacheDir)) {
+ ALOGE("Failed to find cache directory in config message.");
+ signalError(NAME_NOT_FOUND);
+ return;
+ }
+
+ status_t err;
+ err = mFilter->configure(msg);
+ if (err != (status_t)OK) {
+ ALOGE("Failed to configure filter component, err %d", err);
+ signalError(err);
+ return;
+ }
+
+ mInputFormat = new AMessage();
+ mInputFormat->setString("mime", mime.c_str());
+ mInputFormat->setInt32("stride", mStride);
+ mInputFormat->setInt32("slice-height", mSliceHeight);
+ mInputFormat->setInt32("color-format", mColorFormatIn);
+ mInputFormat->setRect("crop", 0, 0, mStride, mSliceHeight);
+ mInputFormat->setInt32("width", mWidth);
+ mInputFormat->setInt32("height", mHeight);
+
+ mOutputFormat = new AMessage();
+ mOutputFormat->setString("mime", mime.c_str());
+ mOutputFormat->setInt32("stride", mStride);
+ mOutputFormat->setInt32("slice-height", mSliceHeight);
+ mOutputFormat->setInt32("color-format", mColorFormatOut);
+ mOutputFormat->setRect("crop", 0, 0, mStride, mSliceHeight);
+ mOutputFormat->setInt32("width", mWidth);
+ mOutputFormat->setInt32("height", mHeight);
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatComponentConfigured);
+ notify->setString("componentName", "MediaFilter");
+ notify->setMessage("input-format", mInputFormat);
+ notify->setMessage("output-format", mOutputFormat);
+ notify->post();
+ mState = CONFIGURED;
+ ALOGV("Handled kWhatConfigureComponent.");
+
+ sendFormatChange();
+}
+
+void MediaFilter::onStart() {
+ CHECK_EQ(mState, CONFIGURED);
+
+ allocateBuffersOnPort(kPortIndexInput);
+
+ allocateBuffersOnPort(kPortIndexOutput);
+
+ status_t err = mFilter->start();
+ if (err != (status_t)OK) {
+ ALOGE("Failed to start filter component, err %d", err);
+ signalError(err);
+ return;
+ }
+
+ mPortEOS[kPortIndexInput] = false;
+ mPortEOS[kPortIndexOutput] = false;
+ mInputEOSResult = OK;
+ mState = STARTED;
+
+ requestFillEmptyInput();
+ ALOGV("Handled kWhatStart.");
+}
+
+void MediaFilter::onInputBufferFilled(const sp<AMessage> &msg) {
+ IOMX::buffer_id bufferID;
+ CHECK(msg->findInt32("buffer-id", (int32_t*)&bufferID));
+ BufferInfo *info = findBufferByID(kPortIndexInput, bufferID);
+
+ if (mState != STARTED) {
+ // we're not running, so we'll just keep that buffer...
+ info->mStatus = BufferInfo::OWNED_BY_US;
+ return;
+ }
+
+ if (info->mGeneration != mGeneration) {
+ ALOGV("Caught a stale input buffer [ID %d]", bufferID);
+ // buffer is stale (taken before a flush/shutdown) - repost it
+ CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_US);
+ postFillThisBuffer(info);
+ return;
+ }
+
+ CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_UPSTREAM);
+ info->mStatus = BufferInfo::OWNED_BY_US;
+
+ sp<ABuffer> buffer;
+ int32_t err = OK;
+ bool eos = false;
+
+ if (!msg->findBuffer("buffer", &buffer)) {
+ // these are unfilled buffers returned by client
+ CHECK(msg->findInt32("err", &err));
+
+ if (err == OK) {
+ // buffers with no errors are returned on MediaCodec.flush
+ ALOGV("saw unfilled buffer (MediaCodec.flush)");
+ postFillThisBuffer(info);
+ return;
+ } else {
+ ALOGV("saw error %d instead of an input buffer", err);
+ eos = true;
+ }
+
+ buffer.clear();
+ }
+
+ int32_t isCSD;
+ if (buffer != NULL && buffer->meta()->findInt32("csd", &isCSD)
+ && isCSD != 0) {
+ // ignore codec-specific data buffers
+ ALOGW("MediaFilter received a codec-specific data buffer");
+ postFillThisBuffer(info);
+ return;
+ }
+
+ int32_t tmp;
+ if (buffer != NULL && buffer->meta()->findInt32("eos", &tmp) && tmp) {
+ eos = true;
+ err = ERROR_END_OF_STREAM;
+ }
+
+ mAvailableInputBuffers.push_back(info);
+ processBuffers();
+
+ if (eos) {
+ mPortEOS[kPortIndexInput] = true;
+ mInputEOSResult = err;
+ }
+
+ ALOGV("Handled kWhatInputBufferFilled. [ID %u]", bufferID);
+}
+
+void MediaFilter::onOutputBufferDrained(const sp<AMessage> &msg) {
+ IOMX::buffer_id bufferID;
+ CHECK(msg->findInt32("buffer-id", (int32_t*)&bufferID));
+ BufferInfo *info = findBufferByID(kPortIndexOutput, bufferID);
+
+ if (mState != STARTED) {
+ // we're not running, so we'll just keep that buffer...
+ info->mStatus = BufferInfo::OWNED_BY_US;
+ return;
+ }
+
+ if (info->mGeneration != mGeneration) {
+ ALOGV("Caught a stale output buffer [ID %d]", bufferID);
+ // buffer is stale (taken before a flush/shutdown) - keep it
+ CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_US);
+ return;
+ }
+
+ CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_UPSTREAM);
+ info->mStatus = BufferInfo::OWNED_BY_US;
+
+ mAvailableOutputBuffers.push_back(info);
+
+ processBuffers();
+
+ ALOGV("Handled kWhatOutputBufferDrained. [ID %u]",
+ bufferID);
+}
+
+void MediaFilter::onShutdown(const sp<AMessage> &msg) {
+ mGeneration++;
+
+ if (mState != UNINITIALIZED) {
+ mFilter->reset();
+ }
+
+ int32_t keepComponentAllocated;
+ CHECK(msg->findInt32("keepComponentAllocated", &keepComponentAllocated));
+ if (!keepComponentAllocated || mState == UNINITIALIZED) {
+ mState = UNINITIALIZED;
+ } else {
+ mState = INITIALIZED;
+ }
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatShutdownCompleted);
+ notify->post();
+}
+
+void MediaFilter::onFlush() {
+ mGeneration++;
+
+ mAvailableInputBuffers.clear();
+ for (size_t i = 0; i < mBuffers[kPortIndexInput].size(); ++i) {
+ BufferInfo *info = &mBuffers[kPortIndexInput].editItemAt(i);
+ info->mStatus = BufferInfo::OWNED_BY_US;
+ }
+ mAvailableOutputBuffers.clear();
+ for (size_t i = 0; i < mBuffers[kPortIndexOutput].size(); ++i) {
+ BufferInfo *info = &mBuffers[kPortIndexOutput].editItemAt(i);
+ info->mStatus = BufferInfo::OWNED_BY_US;
+ mAvailableOutputBuffers.push_back(info);
+ }
+
+ mPortEOS[kPortIndexInput] = false;
+ mPortEOS[kPortIndexOutput] = false;
+ mInputEOSResult = OK;
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatFlushCompleted);
+ notify->post();
+ ALOGV("Posted kWhatFlushCompleted");
+
+ // MediaCodec returns all input buffers after flush, so in
+ // onInputBufferFilled we call postFillThisBuffer on them
+}
+
+void MediaFilter::onSetParameters(const sp<AMessage> &msg) {
+ CHECK(mState != STARTED);
+
+ status_t err = mFilter->setParameters(msg);
+ if (err != (status_t)OK) {
+ ALOGE("setParameters returned err %d", err);
+ }
+}
+
+void MediaFilter::onCreateInputSurface() {
+ CHECK(mState == CONFIGURED);
+
+ mGraphicBufferListener = new GraphicBufferListener;
+
+ sp<AMessage> notify = new AMessage();
+ notify->setTarget(this);
+ status_t err = mGraphicBufferListener->init(
+ notify, mStride, mSliceHeight, kBufferCountActual);
+
+ if (err != OK) {
+ ALOGE("Failed to init mGraphicBufferListener: %d", err);
+ signalError(err);
+ return;
+ }
+
+ sp<AMessage> reply = mNotify->dup();
+ reply->setInt32("what", CodecBase::kWhatInputSurfaceCreated);
+ reply->setObject(
+ "input-surface",
+ new BufferProducerWrapper(
+ mGraphicBufferListener->getIGraphicBufferProducer()));
+ reply->post();
+}
+
+void MediaFilter::onInputFrameAvailable() {
+ BufferItem item = mGraphicBufferListener->getBufferItem();
+ sp<GraphicBuffer> buf = mGraphicBufferListener->getBuffer(item);
+
+ // get pointer to graphic buffer
+ void* bufPtr;
+ buf->lock(GraphicBuffer::USAGE_SW_READ_OFTEN, &bufPtr);
+
+ // HACK - there is no OMX_COLOR_FORMATTYPE value for RGBA, so the format
+ // conversion is hardcoded until we add this.
+ // TODO: check input format and convert only if necessary
+ // copy RGBA graphic buffer into temporary ARGB input buffer
+ BufferInfo *inputInfo = new BufferInfo;
+ inputInfo->mData = new ABuffer(buf->getWidth() * buf->getHeight() * 4);
+ ALOGV("Copying surface data into temp buffer.");
+ convertRGBAToARGB(
+ (uint8_t*)bufPtr, buf->getWidth(), buf->getHeight(),
+ buf->getStride(), inputInfo->mData->data());
+ inputInfo->mBufferID = item.mBuf;
+ inputInfo->mGeneration = mGeneration;
+ inputInfo->mOutputFlags = 0;
+ inputInfo->mStatus = BufferInfo::OWNED_BY_US;
+ inputInfo->mData->meta()->setInt64("timeUs", item.mTimestamp / 1000);
+
+ mAvailableInputBuffers.push_back(inputInfo);
+
+ mGraphicBufferListener->releaseBuffer(item);
+
+ signalProcessBuffers();
+}
+
+void MediaFilter::onSignalEndOfInputStream() {
+ // if using input surface, need to send an EOS output buffer
+ if (mGraphicBufferListener != NULL) {
+ Vector<BufferInfo> *outputBufs = &mBuffers[kPortIndexOutput];
+ BufferInfo* eosBuf;
+ bool foundBuf = false;
+ for (size_t i = 0; i < kBufferCountActual; i++) {
+ eosBuf = &outputBufs->editItemAt(i);
+ if (eosBuf->mStatus == BufferInfo::OWNED_BY_US) {
+ foundBuf = true;
+ break;
+ }
+ }
+
+ if (!foundBuf) {
+ ALOGE("onSignalEndOfInputStream failed to find an output buffer");
+ return;
+ }
+
+ eosBuf->mOutputFlags = OMX_BUFFERFLAG_EOS;
+ eosBuf->mGeneration = mGeneration;
+ eosBuf->mData->setRange(0, 0);
+ postDrainThisBuffer(eosBuf);
+ ALOGV("Posted EOS on output buffer %zu", eosBuf->mBufferID);
+ }
+
+ mPortEOS[kPortIndexOutput] = true;
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatSignaledInputEOS);
+ notify->post();
+
+ ALOGV("Output stream saw EOS.");
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/RSFilter.cpp b/media/libstagefright/filters/RSFilter.cpp
new file mode 100644
index 0000000..b569945
--- /dev/null
+++ b/media/libstagefright/filters/RSFilter.cpp
@@ -0,0 +1,96 @@
+/*
+ * 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 "RSFilter"
+
+#include <utils/Log.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "RSFilter.h"
+
+namespace android {
+
+RSFilter::RSFilter() {
+
+}
+
+RSFilter::~RSFilter() {
+
+}
+
+status_t RSFilter::configure(const sp<AMessage> &msg) {
+ status_t err = SimpleFilter::configure(msg);
+ if (err != OK) {
+ return err;
+ }
+
+ if (!msg->findString("cacheDir", &mCacheDir)) {
+ ALOGE("Failed to find cache directory in config message.");
+ return NAME_NOT_FOUND;
+ }
+
+ sp<RenderScriptWrapper> wrapper;
+ if (!msg->findObject("rs-wrapper", (sp<RefBase>*)&wrapper)) {
+ ALOGE("Failed to find RenderScriptWrapper in config message.");
+ return NAME_NOT_FOUND;
+ }
+
+ mRS = wrapper->mContext;
+ mCallback = wrapper->mCallback;
+
+ return OK;
+}
+
+status_t RSFilter::start() {
+ // 32-bit elements for ARGB8888
+ RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS);
+
+ RSC::Type::Builder tb(mRS, e);
+ tb.setX(mWidth);
+ tb.setY(mHeight);
+ RSC::sp<const RSC::Type> t = tb.create();
+
+ mAllocIn = RSC::Allocation::createTyped(mRS, t);
+ mAllocOut = RSC::Allocation::createTyped(mRS, t);
+
+ return OK;
+}
+
+void RSFilter::reset() {
+ mCallback.clear();
+ mAllocOut.clear();
+ mAllocIn.clear();
+ mRS.clear();
+}
+
+status_t RSFilter::setParameters(const sp<AMessage> &msg) {
+ return mCallback->handleSetParameters(msg);
+}
+
+status_t RSFilter::processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+ mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data());
+ mCallback->processBuffers(mAllocIn.get(), mAllocOut.get());
+ mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data());
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/RSFilter.h b/media/libstagefright/filters/RSFilter.h
new file mode 100644
index 0000000..c5b5074
--- /dev/null
+++ b/media/libstagefright/filters/RSFilter.h
@@ -0,0 +1,53 @@
+/*
+ * 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 RS_FILTER_H_
+#define RS_FILTER_H_
+
+#include <media/stagefright/RenderScriptWrapper.h>
+#include <RenderScript.h>
+
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct AString;
+
+struct RSFilter : public SimpleFilter {
+public:
+ RSFilter();
+
+ virtual status_t configure(const sp<AMessage> &msg);
+ virtual status_t start();
+ virtual void reset();
+ virtual status_t setParameters(const sp<AMessage> &msg);
+ virtual status_t processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+ virtual ~RSFilter();
+
+private:
+ AString mCacheDir;
+ sp<RenderScriptWrapper::RSFilterCallback> mCallback;
+ RSC::sp<RSC::RS> mRS;
+ RSC::sp<RSC::Allocation> mAllocIn;
+ RSC::sp<RSC::Allocation> mAllocOut;
+};
+
+} // namespace android
+
+#endif // RS_FILTER_H_
diff --git a/media/libstagefright/filters/SaturationFilter.cpp b/media/libstagefright/filters/SaturationFilter.cpp
new file mode 100644
index 0000000..ba5f75a
--- /dev/null
+++ b/media/libstagefright/filters/SaturationFilter.cpp
@@ -0,0 +1,99 @@
+/*
+ * 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 "SaturationFilter"
+
+#include <utils/Log.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "SaturationFilter.h"
+
+namespace android {
+
+status_t SaturationFilter::configure(const sp<AMessage> &msg) {
+ status_t err = SimpleFilter::configure(msg);
+ if (err != OK) {
+ return err;
+ }
+
+ if (!msg->findString("cacheDir", &mCacheDir)) {
+ ALOGE("Failed to find cache directory in config message.");
+ return NAME_NOT_FOUND;
+ }
+
+ return OK;
+}
+
+status_t SaturationFilter::start() {
+ // TODO: use a single RS context object for entire application
+ mRS = new RSC::RS();
+
+ if (!mRS->init(mCacheDir.c_str())) {
+ ALOGE("Failed to initialize RenderScript context.");
+ return NO_INIT;
+ }
+
+ // 32-bit elements for ARGB8888
+ RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS);
+
+ RSC::Type::Builder tb(mRS, e);
+ tb.setX(mWidth);
+ tb.setY(mHeight);
+ RSC::sp<const RSC::Type> t = tb.create();
+
+ mAllocIn = RSC::Allocation::createTyped(mRS, t);
+ mAllocOut = RSC::Allocation::createTyped(mRS, t);
+
+ mScript = new ScriptC_saturationARGB(mRS);
+
+ mScript->set_gSaturation(mSaturation);
+
+ return OK;
+}
+
+void SaturationFilter::reset() {
+ mScript.clear();
+ mAllocOut.clear();
+ mAllocIn.clear();
+ mRS.clear();
+}
+
+status_t SaturationFilter::setParameters(const sp<AMessage> &msg) {
+ sp<AMessage> params;
+ CHECK(msg->findMessage("params", ¶ms));
+
+ float saturation;
+ if (params->findFloat("saturation", &saturation)) {
+ mSaturation = saturation;
+ }
+
+ return OK;
+}
+
+status_t SaturationFilter::processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+ mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data());
+ mScript->forEach_root(mAllocIn, mAllocOut);
+ mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data());
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/SaturationFilter.h b/media/libstagefright/filters/SaturationFilter.h
new file mode 100644
index 0000000..0545021
--- /dev/null
+++ b/media/libstagefright/filters/SaturationFilter.h
@@ -0,0 +1,52 @@
+/*
+ * 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 SATURATION_FILTER_H_
+#define SATURATION_FILTER_H_
+
+#include <RenderScript.h>
+
+#include "ScriptC_saturationARGB.h"
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct SaturationFilter : public SimpleFilter {
+public:
+ SaturationFilter() : mSaturation(1.f) {};
+
+ virtual status_t configure(const sp<AMessage> &msg);
+ virtual status_t start();
+ virtual void reset();
+ virtual status_t setParameters(const sp<AMessage> &msg);
+ virtual status_t processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+ virtual ~SaturationFilter() {};
+
+private:
+ AString mCacheDir;
+ RSC::sp<RSC::RS> mRS;
+ RSC::sp<RSC::Allocation> mAllocIn;
+ RSC::sp<RSC::Allocation> mAllocOut;
+ RSC::sp<ScriptC_saturationARGB> mScript;
+ float mSaturation;
+};
+
+} // namespace android
+
+#endif // SATURATION_FILTER_H_
diff --git a/media/libstagefright/filters/SimpleFilter.cpp b/media/libstagefright/filters/SimpleFilter.cpp
new file mode 100644
index 0000000..6c1ca2c
--- /dev/null
+++ b/media/libstagefright/filters/SimpleFilter.cpp
@@ -0,0 +1,39 @@
+/*
+ * 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.
+ */
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "SimpleFilter.h"
+
+namespace android {
+
+status_t SimpleFilter::configure(const sp<AMessage> &msg) {
+ CHECK(msg->findInt32("width", &mWidth));
+ CHECK(msg->findInt32("height", &mHeight));
+ if (!msg->findInt32("stride", &mStride)) {
+ mStride = mWidth;
+ }
+ if (!msg->findInt32("slice-height", &mSliceHeight)) {
+ mSliceHeight = mHeight;
+ }
+ CHECK(msg->findInt32("color-format", &mColorFormatIn));
+ mColorFormatOut = mColorFormatIn;
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/SimpleFilter.h b/media/libstagefright/filters/SimpleFilter.h
new file mode 100644
index 0000000..4cd37ef
--- /dev/null
+++ b/media/libstagefright/filters/SimpleFilter.h
@@ -0,0 +1,52 @@
+/*
+ * 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 SIMPLE_FILTER_H_
+#define SIMPLE_FILTER_H_
+
+#include <stdint.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+struct ABuffer;
+struct AMessage;
+
+namespace android {
+
+struct SimpleFilter : public RefBase {
+public:
+ SimpleFilter() : mWidth(0), mHeight(0), mStride(0), mSliceHeight(0),
+ mColorFormatIn(0), mColorFormatOut(0) {};
+
+ virtual status_t configure(const sp<AMessage> &msg);
+
+ virtual status_t start() = 0;
+ virtual void reset() = 0;
+ virtual status_t setParameters(const sp<AMessage> &msg) = 0;
+ virtual status_t processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) = 0;
+
+protected:
+ int32_t mWidth, mHeight;
+ int32_t mStride, mSliceHeight;
+ int32_t mColorFormatIn, mColorFormatOut;
+
+ virtual ~SimpleFilter() {};
+};
+
+} // namespace android
+
+#endif // SIMPLE_FILTER_H_
diff --git a/media/libstagefright/filters/ZeroFilter.cpp b/media/libstagefright/filters/ZeroFilter.cpp
new file mode 100644
index 0000000..3f1243c
--- /dev/null
+++ b/media/libstagefright/filters/ZeroFilter.cpp
@@ -0,0 +1,57 @@
+/*
+ * 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 "ZeroFilter"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "ZeroFilter.h"
+
+namespace android {
+
+status_t ZeroFilter::setParameters(const sp<AMessage> &msg) {
+ sp<AMessage> params;
+ CHECK(msg->findMessage("params", ¶ms));
+
+ int32_t invert;
+ if (params->findInt32("invert", &invert)) {
+ mInvertData = (invert != 0);
+ }
+
+ return OK;
+}
+
+status_t ZeroFilter::processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+ // assuming identical input & output buffers, since we're a copy filter
+ if (mInvertData) {
+ uint32_t* src = (uint32_t*)srcBuffer->data();
+ uint32_t* dest = (uint32_t*)outBuffer->data();
+ for (size_t i = 0; i < srcBuffer->size() / 4; ++i) {
+ *(dest++) = *(src++) ^ 0xFFFFFFFF;
+ }
+ } else {
+ memcpy(outBuffer->data(), srcBuffer->data(), srcBuffer->size());
+ }
+ outBuffer->setRange(0, srcBuffer->size());
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/ZeroFilter.h b/media/libstagefright/filters/ZeroFilter.h
new file mode 100644
index 0000000..bd34dfb
--- /dev/null
+++ b/media/libstagefright/filters/ZeroFilter.h
@@ -0,0 +1,43 @@
+/*
+ * 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 ZERO_FILTER_H_
+#define ZERO_FILTER_H_
+
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct ZeroFilter : public SimpleFilter {
+public:
+ ZeroFilter() : mInvertData(false) {};
+
+ virtual status_t start() { return OK; };
+ virtual void reset() {};
+ virtual status_t setParameters(const sp<AMessage> &msg);
+ virtual status_t processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+ virtual ~ZeroFilter() {};
+
+private:
+ bool mInvertData;
+};
+
+} // namespace android
+
+#endif // ZERO_FILTER_H_
diff --git a/media/libstagefright/filters/saturation.rs b/media/libstagefright/filters/saturation.rs
new file mode 100644
index 0000000..2c867ac
--- /dev/null
+++ b/media/libstagefright/filters/saturation.rs
@@ -0,0 +1,40 @@
+// Sample script for RGB888 support (compare to saturationARGB.rs)
+/*
+ * 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.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+
+// global variables (parameters accessible to application code)
+float gSaturation = 1.0f;
+
+void root(const uchar3 *v_in, uchar3 *v_out) {
+ // scale 0-255 uchar to 0-1.0 float
+ float3 in = {v_in->r * 0.003921569f, v_in->g * 0.003921569f,
+ v_in->b * 0.003921569f};
+
+ // apply saturation filter
+ float3 result = dot(in, gMonoMult);
+ result = mix(result, in, gSaturation);
+
+ // convert to uchar, copied from rsPackColorTo8888
+ v_out->x = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+ v_out->y = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+ v_out->z = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/media/libstagefright/filters/saturationARGB.rs b/media/libstagefright/filters/saturationARGB.rs
new file mode 100644
index 0000000..1de9dd8
--- /dev/null
+++ b/media/libstagefright/filters/saturationARGB.rs
@@ -0,0 +1,40 @@
+/*
+ * 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.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+
+// global variables (parameters accessible to application code)
+float gSaturation = 1.0f;
+
+void root(const uchar4 *v_in, uchar4 *v_out) {
+ v_out->x = v_in->x; // don't modify A
+
+ // get RGB, scale 0-255 uchar to 0-1.0 float
+ float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f,
+ v_in->w * 0.003921569f};
+
+ // apply saturation filter
+ float3 result = dot(rgb, gMonoMult);
+ result = mix(result, rgb, gSaturation);
+
+ v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+ v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+ v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/media/libstagefright/foundation/AHandler.cpp b/media/libstagefright/foundation/AHandler.cpp
index bd5f7e9..7dbbe54 100644
--- a/media/libstagefright/foundation/AHandler.cpp
+++ b/media/libstagefright/foundation/AHandler.cpp
@@ -19,15 +19,23 @@
#include <utils/Log.h>
#include <media/stagefright/foundation/AHandler.h>
-
-#include <media/stagefright/foundation/ALooperRoster.h>
+#include <media/stagefright/foundation/AMessage.h>
namespace android {
-sp<ALooper> AHandler::looper() {
- extern ALooperRoster gLooperRoster;
+void AHandler::deliverMessage(const sp<AMessage> &msg) {
+ onMessageReceived(msg);
+ mMessageCounter++;
- return gLooperRoster.findLooper(id());
+ if (mVerboseStats) {
+ uint32_t what = msg->what();
+ ssize_t idx = mMessages.indexOfKey(what);
+ if (idx < 0) {
+ mMessages.add(what, 1);
+ } else {
+ mMessages.editValueAt(idx)++;
+ }
+ }
}
} // namespace android
diff --git a/media/libstagefright/foundation/ALooper.cpp b/media/libstagefright/foundation/ALooper.cpp
index 88b1c92..90b5f68 100644
--- a/media/libstagefright/foundation/ALooper.cpp
+++ b/media/libstagefright/foundation/ALooper.cpp
@@ -16,6 +16,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "ALooper"
+
+#include <media/stagefright/foundation/ADebug.h>
+
#include <utils/Log.h>
#include <sys/time.h>
@@ -210,7 +213,7 @@
mEventQueue.erase(mEventQueue.begin());
}
- gLooperRoster.deliverMessage(event.mMessage);
+ event.mMessage->deliver();
// NOTE: It's important to note that at this point our "ALooper" object
// may no longer exist (its final reference may have gone away while
@@ -220,4 +223,29 @@
return true;
}
+// to be called by AMessage::postAndAwaitResponse only
+sp<AReplyToken> ALooper::createReplyToken() {
+ return new AReplyToken(this);
+}
+
+// to be called by AMessage::postAndAwaitResponse only
+status_t ALooper::awaitResponse(const sp<AReplyToken> &replyToken, sp<AMessage> *response) {
+ // return status in case we want to handle an interrupted wait
+ Mutex::Autolock autoLock(mRepliesLock);
+ CHECK(replyToken != NULL);
+ while (!replyToken->retrieveReply(response)) {
+ mRepliesCondition.wait(mRepliesLock);
+ }
+ return OK;
+}
+
+status_t ALooper::postReply(const sp<AReplyToken> &replyToken, const sp<AMessage> &reply) {
+ Mutex::Autolock autoLock(mRepliesLock);
+ status_t err = replyToken->setReply(reply);
+ if (err == OK) {
+ mRepliesCondition.broadcast();
+ }
+ return err;
+}
+
} // namespace android
diff --git a/media/libstagefright/foundation/ALooperRoster.cpp b/media/libstagefright/foundation/ALooperRoster.cpp
index 2d57aee..473ce1b 100644
--- a/media/libstagefright/foundation/ALooperRoster.cpp
+++ b/media/libstagefright/foundation/ALooperRoster.cpp
@@ -30,8 +30,7 @@
static bool verboseStats = false;
ALooperRoster::ALooperRoster()
- : mNextHandlerID(1),
- mNextReplyID(1) {
+ : mNextHandlerID(1) {
}
ALooper::handler_id ALooperRoster::registerHandler(
@@ -49,7 +48,7 @@
ALooper::handler_id handlerID = mNextHandlerID++;
mHandlers.add(handlerID, info);
- handler->setID(handlerID);
+ handler->setID(handlerID, looper);
return handlerID;
}
@@ -68,7 +67,7 @@
sp<AHandler> handler = info.mHandler.promote();
if (handler != NULL) {
- handler->setID(0);
+ handler->setID(0, NULL);
}
mHandlers.removeItemsAt(index);
@@ -100,116 +99,6 @@
}
}
-status_t ALooperRoster::postMessage(
- const sp<AMessage> &msg, int64_t delayUs) {
-
- sp<ALooper> looper = findLooper(msg->target());
-
- if (looper == NULL) {
- return -ENOENT;
- }
- looper->post(msg, delayUs);
- return OK;
-}
-
-void ALooperRoster::deliverMessage(const sp<AMessage> &msg) {
- sp<AHandler> handler;
-
- {
- Mutex::Autolock autoLock(mLock);
-
- ssize_t index = mHandlers.indexOfKey(msg->target());
-
- if (index < 0) {
- ALOGW("failed to deliver message. Target handler not registered.");
- return;
- }
-
- const HandlerInfo &info = mHandlers.valueAt(index);
- handler = info.mHandler.promote();
-
- if (handler == NULL) {
- ALOGW("failed to deliver message. "
- "Target handler %d registered, but object gone.",
- msg->target());
-
- mHandlers.removeItemsAt(index);
- return;
- }
- }
-
- handler->onMessageReceived(msg);
- handler->mMessageCounter++;
-
- if (verboseStats) {
- uint32_t what = msg->what();
- ssize_t idx = handler->mMessages.indexOfKey(what);
- if (idx < 0) {
- handler->mMessages.add(what, 1);
- } else {
- handler->mMessages.editValueAt(idx)++;
- }
- }
-}
-
-sp<ALooper> ALooperRoster::findLooper(ALooper::handler_id handlerID) {
- Mutex::Autolock autoLock(mLock);
-
- ssize_t index = mHandlers.indexOfKey(handlerID);
-
- if (index < 0) {
- return NULL;
- }
-
- sp<ALooper> looper = mHandlers.valueAt(index).mLooper.promote();
-
- if (looper == NULL) {
- mHandlers.removeItemsAt(index);
- return NULL;
- }
-
- return looper;
-}
-
-status_t ALooperRoster::postAndAwaitResponse(
- const sp<AMessage> &msg, sp<AMessage> *response) {
- sp<ALooper> looper = findLooper(msg->target());
-
- if (looper == NULL) {
- ALOGW("failed to post message. "
- "Target handler %d still registered, but object gone.",
- msg->target());
- response->clear();
- return -ENOENT;
- }
-
- Mutex::Autolock autoLock(mLock);
-
- uint32_t replyID = mNextReplyID++;
-
- msg->setInt32("replyID", replyID);
-
- looper->post(msg, 0 /* delayUs */);
-
- ssize_t index;
- while ((index = mReplies.indexOfKey(replyID)) < 0) {
- mRepliesCondition.wait(mLock);
- }
-
- *response = mReplies.valueAt(index);
- mReplies.removeItemsAt(index);
-
- return OK;
-}
-
-void ALooperRoster::postReply(uint32_t replyID, const sp<AMessage> &reply) {
- Mutex::Autolock autoLock(mLock);
-
- CHECK(mReplies.indexOfKey(replyID) < 0);
- mReplies.add(replyID, reply);
- mRepliesCondition.broadcast();
-}
-
static void makeFourCC(uint32_t fourcc, char *s) {
s[0] = (fourcc >> 24) & 0xff;
if (s[0]) {
@@ -225,7 +114,7 @@
void ALooperRoster::dump(int fd, const Vector<String16>& args) {
bool clear = false;
bool oldVerbose = verboseStats;
- for (size_t i = 0;i < args.size(); i++) {
+ for (size_t i = 0; i < args.size(); i++) {
if (args[i] == String16("-c")) {
clear = true;
} else if (args[i] == String16("-von")) {
@@ -241,22 +130,23 @@
Mutex::Autolock autoLock(mLock);
size_t n = mHandlers.size();
- s.appendFormat(" %zd registered handlers:\n", n);
+ s.appendFormat(" %zu registered handlers:\n", n);
for (size_t i = 0; i < n; i++) {
- s.appendFormat(" %zd: ", i);
+ s.appendFormat(" %d: ", mHandlers.keyAt(i));
HandlerInfo &info = mHandlers.editValueAt(i);
sp<ALooper> looper = info.mLooper.promote();
if (looper != NULL) {
- s.append(looper->mName.c_str());
+ s.append(looper->getName());
sp<AHandler> handler = info.mHandler.promote();
if (handler != NULL) {
+ handler->mVerboseStats = verboseStats;
s.appendFormat(": %u messages processed", handler->mMessageCounter);
if (verboseStats) {
for (size_t j = 0; j < handler->mMessages.size(); j++) {
char fourcc[15];
makeFourCC(handler->mMessages.keyAt(j), fourcc);
- s.appendFormat("\n %s: %d",
+ s.appendFormat("\n %s: %u",
fourcc,
handler->mMessages.valueAt(j));
}
diff --git a/media/libstagefright/foundation/AMessage.cpp b/media/libstagefright/foundation/AMessage.cpp
index 1f46bc9..e549ff6 100644
--- a/media/libstagefright/foundation/AMessage.cpp
+++ b/media/libstagefright/foundation/AMessage.cpp
@@ -27,6 +27,7 @@
#include "ABuffer.h"
#include "ADebug.h"
#include "ALooperRoster.h"
+#include "AHandler.h"
#include "AString.h"
#include <binder/Parcel.h>
@@ -36,12 +37,29 @@
extern ALooperRoster gLooperRoster;
-AMessage::AMessage(uint32_t what, ALooper::handler_id target)
- : mWhat(what),
- mTarget(target),
+status_t AReplyToken::setReply(const sp<AMessage> &reply) {
+ if (mReplied) {
+ ALOGE("trying to post a duplicate reply");
+ return -EBUSY;
+ }
+ CHECK(mReply == NULL);
+ mReply = reply;
+ mReplied = true;
+ return OK;
+}
+
+AMessage::AMessage(void)
+ : mWhat(0),
+ mTarget(0),
mNumItems(0) {
}
+AMessage::AMessage(uint32_t what, const sp<const AHandler> &handler)
+ : mWhat(what),
+ mNumItems(0) {
+ setTarget(handler);
+}
+
AMessage::~AMessage() {
clear();
}
@@ -54,12 +72,16 @@
return mWhat;
}
-void AMessage::setTarget(ALooper::handler_id handlerID) {
- mTarget = handlerID;
-}
-
-ALooper::handler_id AMessage::target() const {
- return mTarget;
+void AMessage::setTarget(const sp<const AHandler> &handler) {
+ if (handler == NULL) {
+ mTarget = 0;
+ mHandler.clear();
+ mLooper.clear();
+ } else {
+ mTarget = handler->id();
+ mHandler = handler->getHandler();
+ mLooper = handler->getLooper();
+ }
}
void AMessage::clear() {
@@ -322,33 +344,76 @@
return true;
}
-void AMessage::post(int64_t delayUs) {
- gLooperRoster.postMessage(this, delayUs);
+void AMessage::deliver() {
+ sp<AHandler> handler = mHandler.promote();
+ if (handler == NULL) {
+ ALOGW("failed to deliver message as target handler %d is gone.", mTarget);
+ return;
+ }
+
+ handler->deliverMessage(this);
+}
+
+status_t AMessage::post(int64_t delayUs) {
+ sp<ALooper> looper = mLooper.promote();
+ if (looper == NULL) {
+ ALOGW("failed to post message as target looper for handler %d is gone.", mTarget);
+ return -ENOENT;
+ }
+
+ looper->post(this, delayUs);
+ return OK;
}
status_t AMessage::postAndAwaitResponse(sp<AMessage> *response) {
- return gLooperRoster.postAndAwaitResponse(this, response);
+ sp<ALooper> looper = mLooper.promote();
+ if (looper == NULL) {
+ ALOGW("failed to post message as target looper for handler %d is gone.", mTarget);
+ return -ENOENT;
+ }
+
+ sp<AReplyToken> token = looper->createReplyToken();
+ if (token == NULL) {
+ ALOGE("failed to create reply token");
+ return -ENOMEM;
+ }
+ setObject("replyID", token);
+
+ looper->post(this, 0 /* delayUs */);
+ return looper->awaitResponse(token, response);
}
-void AMessage::postReply(uint32_t replyID) {
- gLooperRoster.postReply(replyID, this);
+status_t AMessage::postReply(const sp<AReplyToken> &replyToken) {
+ if (replyToken == NULL) {
+ ALOGW("failed to post reply to a NULL token");
+ return -ENOENT;
+ }
+ sp<ALooper> looper = replyToken->getLooper();
+ if (looper == NULL) {
+ ALOGW("failed to post reply as target looper is gone.");
+ return -ENOENT;
+ }
+ return looper->postReply(replyToken, this);
}
-bool AMessage::senderAwaitsResponse(uint32_t *replyID) const {
- int32_t tmp;
- bool found = findInt32("replyID", &tmp);
+bool AMessage::senderAwaitsResponse(sp<AReplyToken> *replyToken) {
+ sp<RefBase> tmp;
+ bool found = findObject("replyID", &tmp);
if (!found) {
return false;
}
- *replyID = static_cast<uint32_t>(tmp);
+ *replyToken = static_cast<AReplyToken *>(tmp.get());
+ tmp.clear();
+ setObject("replyID", tmp);
+ // TODO: delete Object instead of setting it to NULL
- return true;
+ return *replyToken != NULL;
}
sp<AMessage> AMessage::dup() const {
- sp<AMessage> msg = new AMessage(mWhat, mTarget);
+ sp<AMessage> msg = new AMessage(mWhat, mHandler.promote());
msg->mNumItems = mNumItems;
#ifdef DUMP_STATS
@@ -532,7 +597,8 @@
// static
sp<AMessage> AMessage::FromParcel(const Parcel &parcel) {
int32_t what = parcel.readInt32();
- sp<AMessage> msg = new AMessage(what);
+ sp<AMessage> msg = new AMessage();
+ msg->setWhat(what);
msg->mNumItems = static_cast<size_t>(parcel.readInt32());
for (size_t i = 0; i < msg->mNumItems; ++i) {
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index d0f3bc2..2d93152 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -33,6 +33,7 @@
#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/DataSource.h>
#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaErrors.h>
@@ -49,8 +50,96 @@
namespace android {
-// Number of recently-read bytes to use for bandwidth estimation
-const size_t LiveSession::kBandwidthHistoryBytes = 200 * 1024;
+// static
+// Bandwidth Switch Mark Defaults
+const int64_t LiveSession::kUpSwitchMarkUs = 15000000ll;
+const int64_t LiveSession::kDownSwitchMarkUs = 20000000ll;
+const int64_t LiveSession::kUpSwitchMarginUs = 5000000ll;
+const int64_t LiveSession::kResumeThresholdUs = 100000ll;
+
+// Buffer Prepare/Ready/Underflow Marks
+const int64_t LiveSession::kReadyMarkUs = 5000000ll;
+const int64_t LiveSession::kPrepareMarkUs = 1500000ll;
+const int64_t LiveSession::kUnderflowMarkUs = 1000000ll;
+
+struct LiveSession::BandwidthEstimator : public RefBase {
+ BandwidthEstimator();
+
+ void addBandwidthMeasurement(size_t numBytes, int64_t delayUs);
+ bool estimateBandwidth(int32_t *bandwidth);
+
+private:
+ // Bandwidth estimation parameters
+ static const int32_t kMaxBandwidthHistoryItems = 20;
+ static const int64_t kMaxBandwidthHistoryWindowUs = 5000000ll; // 5 sec
+
+ struct BandwidthEntry {
+ int64_t mDelayUs;
+ size_t mNumBytes;
+ };
+
+ Mutex mLock;
+ List<BandwidthEntry> mBandwidthHistory;
+ int64_t mTotalTransferTimeUs;
+ size_t mTotalTransferBytes;
+
+ DISALLOW_EVIL_CONSTRUCTORS(BandwidthEstimator);
+};
+
+LiveSession::BandwidthEstimator::BandwidthEstimator() :
+ mTotalTransferTimeUs(0),
+ mTotalTransferBytes(0) {
+}
+
+void LiveSession::BandwidthEstimator::addBandwidthMeasurement(
+ size_t numBytes, int64_t delayUs) {
+ AutoMutex autoLock(mLock);
+
+ BandwidthEntry entry;
+ entry.mDelayUs = delayUs;
+ entry.mNumBytes = numBytes;
+ mTotalTransferTimeUs += delayUs;
+ mTotalTransferBytes += numBytes;
+ mBandwidthHistory.push_back(entry);
+
+ // trim old samples, keeping at least kMaxBandwidthHistoryItems samples,
+ // and total transfer time at least kMaxBandwidthHistoryWindowUs.
+ while (mBandwidthHistory.size() > kMaxBandwidthHistoryItems) {
+ List<BandwidthEntry>::iterator it = mBandwidthHistory.begin();
+ if (mTotalTransferTimeUs - it->mDelayUs < kMaxBandwidthHistoryWindowUs) {
+ break;
+ }
+ mTotalTransferTimeUs -= it->mDelayUs;
+ mTotalTransferBytes -= it->mNumBytes;
+ mBandwidthHistory.erase(mBandwidthHistory.begin());
+ }
+}
+
+bool LiveSession::BandwidthEstimator::estimateBandwidth(int32_t *bandwidthBps) {
+ AutoMutex autoLock(mLock);
+
+ if (mBandwidthHistory.size() < 2) {
+ return false;
+ }
+
+ *bandwidthBps = ((double)mTotalTransferBytes * 8E6 / mTotalTransferTimeUs);
+ return true;
+}
+
+//static
+const char *LiveSession::getKeyForStream(StreamType type) {
+ switch (type) {
+ case STREAMTYPE_VIDEO:
+ return "timeUsVideo";
+ case STREAMTYPE_AUDIO:
+ return "timeUsAudio";
+ case STREAMTYPE_SUBTITLES:
+ return "timeUsSubtitle";
+ default:
+ TRESPASS();
+ }
+ return NULL;
+}
LiveSession::LiveSession(
const sp<AMessage> ¬ify, uint32_t flags,
@@ -58,146 +147,68 @@
: mNotify(notify),
mFlags(flags),
mHTTPService(httpService),
+ mBuffering(false),
mInPreparationPhase(true),
+ mPollBufferingGeneration(0),
+ mPrevBufferPercentage(-1),
mHTTPDataSource(new MediaHTTP(mHTTPService->makeHTTPConnection())),
mCurBandwidthIndex(-1),
+ mOrigBandwidthIndex(-1),
+ mLastBandwidthBps(-1ll),
+ mBandwidthEstimator(new BandwidthEstimator()),
mStreamMask(0),
mNewStreamMask(0),
mSwapMask(0),
- mCheckBandwidthGeneration(0),
mSwitchGeneration(0),
mSubtitleGeneration(0),
mLastDequeuedTimeUs(0ll),
mRealTimeBaseUs(0ll),
mReconfigurationInProgress(false),
mSwitchInProgress(false),
- mDisconnectReplyID(0),
- mSeekReplyID(0),
+ mUpSwitchMark(kUpSwitchMarkUs),
+ mDownSwitchMark(kDownSwitchMarkUs),
+ mUpSwitchMargin(kUpSwitchMarginUs),
mFirstTimeUsValid(false),
mFirstTimeUs(0),
mLastSeekTimeUs(0) {
-
mStreams[kAudioIndex] = StreamItem("audio");
mStreams[kVideoIndex] = StreamItem("video");
mStreams[kSubtitleIndex] = StreamItem("subtitles");
for (size_t i = 0; i < kMaxStreams; ++i) {
- mDiscontinuities.add(indexToType(i), new AnotherPacketSource(NULL /* meta */));
mPacketSources.add(indexToType(i), new AnotherPacketSource(NULL /* meta */));
mPacketSources2.add(indexToType(i), new AnotherPacketSource(NULL /* meta */));
- mBuffering[i] = false;
}
-
- size_t numHistoryItems = kBandwidthHistoryBytes /
- PlaylistFetcher::kDownloadBlockSize + 1;
- if (numHistoryItems < 5) {
- numHistoryItems = 5;
- }
- mHTTPDataSource->setBandwidthHistorySize(numHistoryItems);
}
LiveSession::~LiveSession() {
-}
-
-sp<ABuffer> LiveSession::createFormatChangeBuffer(bool swap) {
- ABuffer *discontinuity = new ABuffer(0);
- discontinuity->meta()->setInt32("discontinuity", ATSParser::DISCONTINUITY_FORMATCHANGE);
- discontinuity->meta()->setInt32("swapPacketSource", swap);
- discontinuity->meta()->setInt32("switchGeneration", mSwitchGeneration);
- discontinuity->meta()->setInt64("timeUs", -1);
- return discontinuity;
-}
-
-void LiveSession::swapPacketSource(StreamType stream) {
- sp<AnotherPacketSource> &aps = mPacketSources.editValueFor(stream);
- sp<AnotherPacketSource> &aps2 = mPacketSources2.editValueFor(stream);
- sp<AnotherPacketSource> tmp = aps;
- aps = aps2;
- aps2 = tmp;
- aps2->clear();
+ if (mFetcherLooper != NULL) {
+ mFetcherLooper->stop();
+ }
}
status_t LiveSession::dequeueAccessUnit(
StreamType stream, sp<ABuffer> *accessUnit) {
- if (!(mStreamMask & stream)) {
- // return -EWOULDBLOCK to avoid halting the decoder
- // when switching between audio/video and audio only.
- return -EWOULDBLOCK;
- }
-
- status_t finalResult;
- sp<AnotherPacketSource> discontinuityQueue = mDiscontinuities.valueFor(stream);
- if (discontinuityQueue->hasBufferAvailable(&finalResult)) {
- discontinuityQueue->dequeueAccessUnit(accessUnit);
- // seeking, track switching
- sp<AMessage> extra;
- int64_t timeUs;
- if ((*accessUnit)->meta()->findMessage("extra", &extra)
- && extra != NULL
- && extra->findInt64("timeUs", &timeUs)) {
- // seeking only
- mLastSeekTimeUs = timeUs;
- mDiscontinuityOffsetTimesUs.clear();
- mDiscontinuityAbsStartTimesUs.clear();
- }
- return INFO_DISCONTINUITY;
- }
-
+ status_t finalResult = OK;
sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(stream);
ssize_t idx = typeToIndex(stream);
- if (!packetSource->hasBufferAvailable(&finalResult)) {
+ // Do not let client pull data if we don't have data packets yet.
+ // We might only have a format discontinuity queued without data.
+ // When NuPlayerDecoder dequeues the format discontinuity, it will
+ // immediately try to getFormat. If we return NULL, NuPlayerDecoder
+ // thinks it can do seamless change, so will not shutdown decoder.
+ // When the actual format arrives, it can't handle it and get stuck.
+ if (!packetSource->hasDataBufferAvailable(&finalResult)) {
if (finalResult == OK) {
- mBuffering[idx] = true;
return -EAGAIN;
} else {
return finalResult;
}
}
- int32_t targetDuration = 0;
- sp<AMessage> meta = packetSource->getLatestEnqueuedMeta();
- if (meta != NULL) {
- meta->findInt32("targetDuration", &targetDuration);
- }
-
- int64_t targetDurationUs = targetDuration * 1000000ll;
- if (targetDurationUs == 0 ||
- targetDurationUs > PlaylistFetcher::kMinBufferedDurationUs) {
- // Fetchers limit buffering to
- // min(3 * targetDuration, kMinBufferedDurationUs)
- targetDurationUs = PlaylistFetcher::kMinBufferedDurationUs;
- }
-
- if (mBuffering[idx]) {
- if (mSwitchInProgress
- || packetSource->isFinished(0)
- || packetSource->getEstimatedDurationUs() > targetDurationUs) {
- mBuffering[idx] = false;
- }
- }
-
- if (mBuffering[idx]) {
- return -EAGAIN;
- }
-
- // wait for counterpart
- sp<AnotherPacketSource> otherSource;
- uint32_t mask = mNewStreamMask & mStreamMask;
- uint32_t fetchersMask = 0;
- for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
- uint32_t fetcherMask = mFetcherInfos.valueAt(i).mFetcher->getStreamTypeMask();
- fetchersMask |= fetcherMask;
- }
- mask &= fetchersMask;
- if (stream == STREAMTYPE_AUDIO && (mask & STREAMTYPE_VIDEO)) {
- otherSource = mPacketSources.valueFor(STREAMTYPE_VIDEO);
- } else if (stream == STREAMTYPE_VIDEO && (mask & STREAMTYPE_AUDIO)) {
- otherSource = mPacketSources.valueFor(STREAMTYPE_AUDIO);
- }
- if (otherSource != NULL && !otherSource->hasBufferAvailable(&finalResult)) {
- return finalResult == OK ? -EAGAIN : finalResult;
- }
+ // Let the client dequeue as long as we have buffers available
+ // Do not make pause/resume decisions here.
status_t err = packetSource->dequeueAccessUnit(accessUnit);
@@ -235,42 +246,6 @@
streamStr,
type,
extra == NULL ? "NULL" : extra->debugString().c_str());
-
- int32_t swap;
- if ((*accessUnit)->meta()->findInt32("swapPacketSource", &swap) && swap) {
- int32_t switchGeneration;
- CHECK((*accessUnit)->meta()->findInt32("switchGeneration", &switchGeneration));
- {
- Mutex::Autolock lock(mSwapMutex);
- if (switchGeneration == mSwitchGeneration) {
- swapPacketSource(stream);
- sp<AMessage> msg = new AMessage(kWhatSwapped, id());
- msg->setInt32("stream", stream);
- msg->setInt32("switchGeneration", switchGeneration);
- msg->post();
- }
- }
- } else {
- size_t seq = strm.mCurDiscontinuitySeq;
- int64_t offsetTimeUs;
- if (mDiscontinuityOffsetTimesUs.indexOfKey(seq) >= 0) {
- offsetTimeUs = mDiscontinuityOffsetTimesUs.valueFor(seq);
- } else {
- offsetTimeUs = 0;
- }
-
- seq += 1;
- if (mDiscontinuityAbsStartTimesUs.indexOfKey(strm.mCurDiscontinuitySeq) >= 0) {
- int64_t firstTimeUs;
- firstTimeUs = mDiscontinuityAbsStartTimesUs.valueFor(strm.mCurDiscontinuitySeq);
- offsetTimeUs += strm.mLastDequeuedTimeUs - firstTimeUs;
- offsetTimeUs += strm.mLastSampleDurationUs;
- } else {
- offsetTimeUs += strm.mLastSampleDurationUs;
- }
-
- mDiscontinuityOffsetTimesUs.add(seq, offsetTimeUs);
- }
} else if (err == OK) {
if (stream == STREAMTYPE_AUDIO || stream == STREAMTYPE_VIDEO) {
@@ -278,7 +253,26 @@
int32_t discontinuitySeq = 0;
CHECK((*accessUnit)->meta()->findInt64("timeUs", &timeUs));
(*accessUnit)->meta()->findInt32("discontinuitySeq", &discontinuitySeq);
- strm.mCurDiscontinuitySeq = discontinuitySeq;
+ if (discontinuitySeq > (int32_t) strm.mCurDiscontinuitySeq) {
+ int64_t offsetTimeUs;
+ if (mDiscontinuityOffsetTimesUs.indexOfKey(strm.mCurDiscontinuitySeq) >= 0) {
+ offsetTimeUs = mDiscontinuityOffsetTimesUs.valueFor(strm.mCurDiscontinuitySeq);
+ } else {
+ offsetTimeUs = 0;
+ }
+
+ if (mDiscontinuityAbsStartTimesUs.indexOfKey(strm.mCurDiscontinuitySeq) >= 0) {
+ int64_t firstTimeUs;
+ firstTimeUs = mDiscontinuityAbsStartTimesUs.valueFor(strm.mCurDiscontinuitySeq);
+ offsetTimeUs += strm.mLastDequeuedTimeUs - firstTimeUs;
+ offsetTimeUs += strm.mLastSampleDurationUs;
+ } else {
+ offsetTimeUs += strm.mLastSampleDurationUs;
+ }
+
+ mDiscontinuityOffsetTimesUs.add(discontinuitySeq, offsetTimeUs);
+ strm.mCurDiscontinuitySeq = discontinuitySeq;
+ }
int32_t discard = 0;
int64_t firstTimeUs;
@@ -331,7 +325,6 @@
}
status_t LiveSession::getStreamFormat(StreamType stream, sp<AMessage> *format) {
- // No swapPacketSource race condition; called from the same thread as dequeueAccessUnit.
if (!(mStreamMask & stream)) {
return UNKNOWN_ERROR;
}
@@ -344,12 +337,21 @@
return -EAGAIN;
}
+ if (stream == STREAMTYPE_AUDIO) {
+ // set AAC input buffer size to 32K bytes (256kbps x 1sec)
+ meta->setInt32(kKeyMaxInputSize, 32 * 1024);
+ }
+
return convertMetaDataToMessage(meta, format);
}
+sp<HTTPBase> LiveSession::getHTTPDataSource() {
+ return new MediaHTTP(mHTTPService->makeHTTPConnection());
+}
+
void LiveSession::connectAsync(
const char *url, const KeyedVector<String8, String8> *headers) {
- sp<AMessage> msg = new AMessage(kWhatConnect, id());
+ sp<AMessage> msg = new AMessage(kWhatConnect, this);
msg->setString("url", url);
if (headers != NULL) {
@@ -362,7 +364,7 @@
}
status_t LiveSession::disconnect() {
- sp<AMessage> msg = new AMessage(kWhatDisconnect, id());
+ sp<AMessage> msg = new AMessage(kWhatDisconnect, this);
sp<AMessage> response;
status_t err = msg->postAndAwaitResponse(&response);
@@ -371,7 +373,7 @@
}
status_t LiveSession::seekTo(int64_t timeUs) {
- sp<AMessage> msg = new AMessage(kWhatSeek, id());
+ sp<AMessage> msg = new AMessage(kWhatSeek, this);
msg->setInt64("timeUs", timeUs);
sp<AMessage> response;
@@ -380,6 +382,95 @@
return err;
}
+bool LiveSession::checkSwitchProgress(
+ sp<AMessage> &stopParams, int64_t delayUs, bool *needResumeUntil) {
+ AString newUri;
+ CHECK(stopParams->findString("uri", &newUri));
+
+ *needResumeUntil = false;
+ sp<AMessage> firstNewMeta[kMaxStreams];
+ for (size_t i = 0; i < kMaxStreams; ++i) {
+ StreamType stream = indexToType(i);
+ if (!(mSwapMask & mNewStreamMask & stream)
+ || (mStreams[i].mNewUri != newUri)) {
+ continue;
+ }
+ if (stream == STREAMTYPE_SUBTITLES) {
+ continue;
+ }
+ sp<AnotherPacketSource> &source = mPacketSources.editValueAt(i);
+
+ // First, get latest dequeued meta, which is where the decoder is at.
+ // (when upswitching, we take the meta after a certain delay, so that
+ // the decoder is left with some cushion)
+ sp<AMessage> lastDequeueMeta, lastEnqueueMeta;
+ if (delayUs > 0) {
+ lastDequeueMeta = source->getMetaAfterLastDequeued(delayUs);
+ if (lastDequeueMeta == NULL) {
+ // this means we don't have enough cushion, try again later
+ ALOGV("[%s] up switching failed due to insufficient buffer",
+ stream == STREAMTYPE_AUDIO ? "audio" : "video");
+ return false;
+ }
+ } else {
+ // It's okay for lastDequeueMeta to be NULL here, it means the
+ // decoder hasn't even started dequeueing
+ lastDequeueMeta = source->getLatestDequeuedMeta();
+ }
+ // Then, trim off packets at beginning of mPacketSources2 that's before
+ // the latest dequeued time. These samples are definitely too late.
+ firstNewMeta[i] = mPacketSources2.editValueAt(i)
+ ->trimBuffersBeforeMeta(lastDequeueMeta);
+
+ // Now firstNewMeta[i] is the first sample after the trim.
+ // If it's NULL, we failed because dequeue already past all samples
+ // in mPacketSource2, we have to try again.
+ if (firstNewMeta[i] == NULL) {
+ HLSTime dequeueTime(lastDequeueMeta);
+ ALOGV("[%s] dequeue time (%d, %lld) past start time",
+ stream == STREAMTYPE_AUDIO ? "audio" : "video",
+ dequeueTime.mSeq, (long long) dequeueTime.mTimeUs);
+ return false;
+ }
+
+ // Otherwise, we check if mPacketSources2 overlaps with what old fetcher
+ // already fetched, and see if we need to resumeUntil
+ lastEnqueueMeta = source->getLatestEnqueuedMeta();
+ // lastEnqueueMeta == NULL means old fetcher stopped at a discontinuity
+ // boundary, no need to resume as the content will look different anyways
+ if (lastEnqueueMeta != NULL) {
+ HLSTime lastTime(lastEnqueueMeta), startTime(firstNewMeta[i]);
+
+ // no need to resume old fetcher if new fetcher started in different
+ // discontinuity sequence, as the content will look different.
+ *needResumeUntil |= (startTime.mSeq == lastTime.mSeq
+ && startTime.mTimeUs - lastTime.mTimeUs > kResumeThresholdUs);
+
+ // update the stopTime for resumeUntil
+ stopParams->setInt32("discontinuitySeq", startTime.mSeq);
+ stopParams->setInt64(getKeyForStream(stream), startTime.mTimeUs);
+ }
+ }
+
+ // if we're here, it means dequeue progress hasn't passed some samples in
+ // mPacketSource2, we can trim off the excess in mPacketSource.
+ // (old fetcher might still need to resumeUntil the start time of new fetcher)
+ for (size_t i = 0; i < kMaxStreams; ++i) {
+ StreamType stream = indexToType(i);
+ if (!(mSwapMask & mNewStreamMask & stream)
+ || (newUri != mStreams[i].mNewUri)
+ || stream == STREAMTYPE_SUBTITLES) {
+ continue;
+ }
+ mPacketSources.valueFor(stream)->trimBuffersAfterMeta(firstNewMeta[i]);
+ }
+
+ // no resumeUntil if already underflow
+ *needResumeUntil &= !mBuffering;
+
+ return true;
+}
+
void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
switch (msg->what()) {
case kWhatConnect:
@@ -402,7 +493,7 @@
case kWhatSeek:
{
- uint32_t seekReplyID;
+ sp<AReplyToken> seekReplyID;
CHECK(msg->senderAwaitsResponse(&seekReplyID));
mSeekReplyID = seekReplyID;
mSeekReply = new AMessage;
@@ -426,16 +517,25 @@
case PlaylistFetcher::kWhatPaused:
case PlaylistFetcher::kWhatStopped:
{
- if (what == PlaylistFetcher::kWhatStopped) {
- AString uri;
- CHECK(msg->findString("uri", &uri));
- if (mFetcherInfos.removeItem(uri) < 0) {
- // ignore duplicated kWhatStopped messages.
- break;
- }
+ AString uri;
+ CHECK(msg->findString("uri", &uri));
+ ssize_t index = mFetcherInfos.indexOfKey(uri);
+ if (index < 0) {
+ // ignore msgs from fetchers that's already gone
+ break;
+ }
- if (mSwitchInProgress) {
- tryToFinishBandwidthSwitch();
+ if (what == PlaylistFetcher::kWhatStopped) {
+ mFetcherLooper->unregisterHandler(
+ mFetcherInfos[index].mFetcher->id());
+ mFetcherInfos.removeItemsAt(index);
+ } else if (what == PlaylistFetcher::kWhatPaused) {
+ int32_t seekMode;
+ CHECK(msg->findInt32("seekMode", &seekMode));
+ for (size_t i = 0; i < kMaxStreams; ++i) {
+ if (mStreams[i].mUri == uri) {
+ mStreams[i].mSeekMode = (SeekMode) seekMode;
+ }
}
}
@@ -443,14 +543,6 @@
CHECK_GT(mContinuationCounter, 0);
if (--mContinuationCounter == 0) {
mContinuation->post();
-
- if (mSeekReplyID != 0) {
- CHECK(mSeekReply != NULL);
- mSeekReply->setInt32("err", OK);
- mSeekReply->postReply(mSeekReplyID);
- mSeekReplyID = 0;
- mSeekReply.clear();
- }
}
}
break;
@@ -464,8 +556,21 @@
int64_t durationUs;
CHECK(msg->findInt64("durationUs", &durationUs));
- FetcherInfo *info = &mFetcherInfos.editValueFor(uri);
- info->mDurationUs = durationUs;
+ ssize_t index = mFetcherInfos.indexOfKey(uri);
+ if (index >= 0) {
+ FetcherInfo *info = &mFetcherInfos.editValueFor(uri);
+ info->mDurationUs = durationUs;
+ }
+ break;
+ }
+
+ case PlaylistFetcher::kWhatTargetDurationUpdate:
+ {
+ int64_t targetDurationUs;
+ CHECK(msg->findInt64("targetDurationUs", &targetDurationUs));
+ mUpSwitchMark = min(kUpSwitchMarkUs, targetDurationUs * 7 / 4);
+ mDownSwitchMark = min(kDownSwitchMarkUs, targetDurationUs * 9 / 4);
+ mUpSwitchMargin = min(kUpSwitchMarginUs, targetDurationUs);
break;
}
@@ -506,38 +611,23 @@
mPacketSources.valueFor(
STREAMTYPE_SUBTITLES)->signalEOS(err);
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatError);
- notify->setInt32("err", err);
- notify->post();
+ postError(err);
break;
}
- case PlaylistFetcher::kWhatTemporarilyDoneFetching:
+ case PlaylistFetcher::kWhatStopReached:
{
- AString uri;
- CHECK(msg->findString("uri", &uri));
+ ALOGV("kWhatStopReached");
- if (mFetcherInfos.indexOfKey(uri) < 0) {
- ALOGE("couldn't find uri");
+ AString oldUri;
+ CHECK(msg->findString("uri", &oldUri));
+
+ ssize_t index = mFetcherInfos.indexOfKey(oldUri);
+ if (index < 0) {
break;
}
- FetcherInfo *info = &mFetcherInfos.editValueFor(uri);
- info->mIsPrepared = true;
- if (mInPreparationPhase) {
- bool allFetchersPrepared = true;
- for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
- if (!mFetcherInfos.valueAt(i).mIsPrepared) {
- allFetchersPrepared = false;
- break;
- }
- }
-
- if (allFetchersPrepared) {
- postPrepared(OK);
- }
- }
+ tryToFinishBandwidthSwitch(oldUri);
break;
}
@@ -550,15 +640,69 @@
break;
}
- // Resume fetcher for the original variant; the resumed fetcher should
- // continue until the timestamps found in msg, which is stored by the
- // new fetcher to indicate where the new variant has started buffering.
- for (size_t i = 0; i < mFetcherInfos.size(); i++) {
- const FetcherInfo info = mFetcherInfos.valueAt(i);
- if (info.mToBeRemoved) {
- info.mFetcher->resumeUntilAsync(msg);
+ AString uri;
+ CHECK(msg->findString("uri", &uri));
+
+ // mark new fetcher mToBeResumed
+ ssize_t index = mFetcherInfos.indexOfKey(uri);
+ if (index >= 0) {
+ mFetcherInfos.editValueAt(index).mToBeResumed = true;
+ }
+
+ // temporarily disable packet sources to be swapped to prevent
+ // NuPlayerDecoder from dequeuing while we check progress
+ for (size_t i = 0; i < mPacketSources.size(); ++i) {
+ if ((mSwapMask & mPacketSources.keyAt(i))
+ && uri == mStreams[i].mNewUri) {
+ mPacketSources.editValueAt(i)->enable(false);
}
}
+ bool switchUp = (mCurBandwidthIndex > mOrigBandwidthIndex);
+ // If switching up, require a cushion bigger than kUnderflowMark
+ // to avoid buffering immediately after the switch.
+ // (If we don't have that cushion we'd rather cancel and try again.)
+ int64_t delayUs = switchUp ? (kUnderflowMarkUs + 1000000ll) : 0;
+ bool needResumeUntil = false;
+ sp<AMessage> stopParams = msg;
+ if (checkSwitchProgress(stopParams, delayUs, &needResumeUntil)) {
+ // playback time hasn't passed startAt time
+ if (!needResumeUntil) {
+ for (size_t i = 0; i < kMaxStreams; ++i) {
+ if ((mSwapMask & indexToType(i))
+ && uri == mStreams[i].mNewUri) {
+ // have to make a copy of mStreams[i].mUri because
+ // tryToFinishBandwidthSwitch is modifying mStreams[]
+ AString oldURI = mStreams[i].mUri;
+ tryToFinishBandwidthSwitch(oldURI);
+ break;
+ }
+ }
+ } else {
+ // startAt time is after last enqueue time
+ // Resume fetcher for the original variant; the resumed fetcher should
+ // continue until the timestamps found in msg, which is stored by the
+ // new fetcher to indicate where the new variant has started buffering.
+ for (size_t i = 0; i < mFetcherInfos.size(); i++) {
+ const FetcherInfo &info = mFetcherInfos.valueAt(i);
+ if (info.mToBeRemoved) {
+ info.mFetcher->resumeUntilAsync(stopParams);
+ }
+ }
+ }
+ } else {
+ // playback time passed startAt time
+ if (switchUp) {
+ // if switching up, cancel and retry if condition satisfies again
+ cancelBandwidthSwitch(true /* resume */);
+ } else {
+ resumeFetcher(uri, mSwapMask, -1, true /* newUri */);
+ }
+ }
+ // re-enable all packet sources
+ for (size_t i = 0; i < mPacketSources.size(); ++i) {
+ mPacketSources.editValueAt(i)->enable(true);
+ }
+
break;
}
@@ -569,19 +713,6 @@
break;
}
- case kWhatCheckBandwidth:
- {
- int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
-
- if (generation != mCheckBandwidthGeneration) {
- break;
- }
-
- onCheckBandwidth(msg);
- break;
- }
-
case kWhatChangeConfiguration:
{
onChangeConfiguration(msg);
@@ -606,21 +737,13 @@
break;
}
- case kWhatSwapped:
+ case kWhatPollBuffering:
{
- onSwapped(msg);
- break;
- }
-
- case kWhatCheckSwitchDown:
- {
- onCheckSwitchDown();
- break;
- }
-
- case kWhatSwitchDown:
- {
- onSwitchDown();
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+ if (generation == mPollBufferingGeneration) {
+ onPollBuffering();
+ }
break;
}
@@ -691,6 +814,14 @@
return;
}
+ // create looper for fetchers
+ if (mFetcherLooper == NULL) {
+ mFetcherLooper = new ALooper();
+
+ mFetcherLooper->setName("Fetcher");
+ mFetcherLooper->start(false, false);
+ }
+
// We trust the content provider to make a reasonable choice of preferred
// initial bandwidth by listing it first in the variant playlist.
// At startup we really don't have a good estimate on the available
@@ -700,6 +831,7 @@
size_t initialBandwidthIndex = 0;
if (mPlaylist->isVariantPlaylist()) {
+ Vector<BandwidthItem> itemsWithVideo;
for (size_t i = 0; i < mPlaylist->size(); ++i) {
BandwidthItem item;
@@ -711,14 +843,22 @@
CHECK(meta->findInt32("bandwidth", (int32_t *)&item.mBandwidth));
- if (initialBandwidth == 0) {
- initialBandwidth = item.mBandwidth;
- }
-
mBandwidthItems.push(item);
+ if (mPlaylist->hasType(i, "video")) {
+ itemsWithVideo.push(item);
+ }
+ }
+ // remove the audio-only variants if we have at least one with video
+ if (!itemsWithVideo.empty()
+ && itemsWithVideo.size() < mBandwidthItems.size()) {
+ mBandwidthItems.clear();
+ for (size_t i = 0; i < itemsWithVideo.size(); ++i) {
+ mBandwidthItems.push(itemsWithVideo[i]);
+ }
}
CHECK_GT(mBandwidthItems.size(), 0u);
+ initialBandwidth = mBandwidthItems[0].mBandwidth;
mBandwidthItems.sort(SortByBandwidth);
@@ -742,22 +882,20 @@
}
void LiveSession::finishDisconnect() {
+ ALOGV("finishDisconnect");
+
// No reconfiguration is currently pending, make sure none will trigger
// during disconnection either.
- cancelCheckBandwidthEvent();
-
- // Protect mPacketSources from a swapPacketSource race condition through disconnect.
- // (finishDisconnect, onFinishDisconnect2)
cancelBandwidthSwitch();
- // cancel switch down monitor
- mSwitchDownMonitor.clear();
+ // cancel buffer polling
+ cancelPollBuffering();
for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
mFetcherInfos.valueAt(i).mFetcher->stopAsync();
}
- sp<AMessage> msg = new AMessage(kWhatFinishDisconnect2, id());
+ sp<AMessage> msg = new AMessage(kWhatFinishDisconnect2, this);
mContinuationCounter = mFetcherInfos.size();
mContinuation = msg;
@@ -780,7 +918,7 @@
response->setInt32("err", OK);
response->postReply(mDisconnectReplyID);
- mDisconnectReplyID = 0;
+ mDisconnectReplyID.clear();
}
sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) {
@@ -790,16 +928,16 @@
return NULL;
}
- sp<AMessage> notify = new AMessage(kWhatFetcherNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatFetcherNotify, this);
notify->setString("uri", uri);
notify->setInt32("switchGeneration", mSwitchGeneration);
FetcherInfo info;
info.mFetcher = new PlaylistFetcher(notify, this, uri, mSubtitleGeneration);
info.mDurationUs = -1ll;
- info.mIsPrepared = false;
info.mToBeRemoved = false;
- looper()->registerHandler(info.mFetcher);
+ info.mToBeResumed = false;
+ mFetcherLooper->registerHandler(info.mFetcher);
mFetcherInfos.add(uri, info);
@@ -827,14 +965,15 @@
int64_t range_offset, int64_t range_length,
uint32_t block_size, /* download block size */
sp<DataSource> *source, /* to return and reuse source */
- String8 *actualUrl) {
+ String8 *actualUrl,
+ bool forceConnectHTTP /* force connect HTTP when resuing source */) {
off64_t size;
sp<DataSource> temp_source;
if (source == NULL) {
source = &temp_source;
}
- if (*source == NULL) {
+ if (*source == NULL || forceConnectHTTP) {
if (!strncasecmp(url, "file://", 7)) {
*source = new FileSource(url + 7);
} else if (strncasecmp(url, "http://", 7)
@@ -853,13 +992,18 @@
? "" : AStringPrintf("%lld",
range_offset + range_length - 1).c_str()).c_str()));
}
- status_t err = mHTTPDataSource->connect(url, &headers);
+
+ HTTPBase* httpDataSource =
+ (*source == NULL) ? mHTTPDataSource.get() : (HTTPBase*)source->get();
+ status_t err = httpDataSource->connect(url, &headers);
if (err != OK) {
return err;
}
- *source = mHTTPDataSource;
+ if (*source == NULL) {
+ *source = mHTTPDataSource;
+ }
}
}
@@ -949,6 +1093,9 @@
String8 actualUrl;
ssize_t err = fetchFile(url, &buffer, 0, -1, 0, NULL, &actualUrl);
+ // close off the connection after use
+ mHTTPDataSource->disconnect();
+
if (err <= 0) {
return NULL;
}
@@ -995,8 +1142,108 @@
}
#endif
-size_t LiveSession::getBandwidthIndex() {
- if (mBandwidthItems.size() == 0) {
+bool LiveSession::resumeFetcher(
+ const AString &uri, uint32_t streamMask, int64_t timeUs, bool newUri) {
+ ssize_t index = mFetcherInfos.indexOfKey(uri);
+ if (index < 0) {
+ ALOGE("did not find fetcher for uri: %s", uri.c_str());
+ return false;
+ }
+
+ bool resume = false;
+ sp<AnotherPacketSource> sources[kMaxStreams];
+ for (size_t i = 0; i < kMaxStreams; ++i) {
+ if ((streamMask & indexToType(i))
+ && ((!newUri && uri == mStreams[i].mUri)
+ || (newUri && uri == mStreams[i].mNewUri))) {
+ resume = true;
+ if (newUri) {
+ sources[i] = mPacketSources2.valueFor(indexToType(i));
+ sources[i]->clear();
+ } else {
+ sources[i] = mPacketSources.valueFor(indexToType(i));
+ }
+ }
+ }
+
+ if (resume) {
+ ALOGV("resuming fetcher %s, timeUs %lld", uri.c_str(), (long long)timeUs);
+ SeekMode seekMode = newUri ? kSeekModeNextSample : kSeekModeExactPosition;
+ mFetcherInfos.editValueAt(index).mFetcher->startAsync(
+ sources[kAudioIndex],
+ sources[kVideoIndex],
+ sources[kSubtitleIndex],
+ timeUs, -1, -1, seekMode);
+ }
+
+ return resume;
+}
+
+float LiveSession::getAbortThreshold(
+ ssize_t currentBWIndex, ssize_t targetBWIndex) const {
+ float abortThreshold = -1.0f;
+ if (currentBWIndex > 0 && targetBWIndex < currentBWIndex) {
+ /*
+ If we're switching down, we need to decide whether to
+
+ 1) finish last segment of high-bandwidth variant, or
+ 2) abort last segment of high-bandwidth variant, and fetch an
+ overlapping portion from low-bandwidth variant.
+
+ Here we try to maximize the amount of buffer left when the
+ switch point is met. Given the following parameters:
+
+ B: our current buffering level in seconds
+ T: target duration in seconds
+ X: sample duration in seconds remain to fetch in last segment
+ bw0: bandwidth of old variant (as specified in playlist)
+ bw1: bandwidth of new variant (as specified in playlist)
+ bw: measured bandwidth available
+
+ If we choose 1), when switch happens at the end of current
+ segment, our buffering will be
+ B + X - X * bw0 / bw
+
+ If we choose 2), when switch happens where we aborted current
+ segment, our buffering will be
+ B - (T - X) * bw1 / bw
+
+ We should only choose 1) if
+ X/T < bw1 / (bw1 + bw0 - bw)
+ */
+
+ // Taking the measured current bandwidth at 50% face value only,
+ // as our bandwidth estimation is a lagging indicator. Being
+ // conservative on this, we prefer switching to lower bandwidth
+ // unless we're really confident finishing up the last segment
+ // of higher bandwidth will be fast.
+ CHECK(mLastBandwidthBps >= 0);
+ abortThreshold =
+ (float)mBandwidthItems.itemAt(targetBWIndex).mBandwidth
+ / ((float)mBandwidthItems.itemAt(targetBWIndex).mBandwidth
+ + (float)mBandwidthItems.itemAt(currentBWIndex).mBandwidth
+ - (float)mLastBandwidthBps * 0.5f);
+ if (abortThreshold < 0.0f) {
+ abortThreshold = -1.0f; // do not abort
+ }
+ ALOGV("Switching Down: bps %ld => %ld, measured %d, abort ratio %.2f",
+ mBandwidthItems.itemAt(currentBWIndex).mBandwidth,
+ mBandwidthItems.itemAt(targetBWIndex).mBandwidth,
+ mLastBandwidthBps,
+ abortThreshold);
+ }
+ return abortThreshold;
+}
+
+void LiveSession::addBandwidthMeasurement(size_t numBytes, int64_t delayUs) {
+ mBandwidthEstimator->addBandwidthMeasurement(numBytes, delayUs);
+}
+
+size_t LiveSession::getBandwidthIndex(int32_t bandwidthBps) {
+ if (mBandwidthItems.size() < 2) {
+ // shouldn't be here if we only have 1 bandwidth, check
+ // logic to get rid of redundant bandwidth polling
+ ALOGW("getBandwidthIndex() called for single bandwidth playlist!");
return 0;
}
@@ -1014,15 +1261,6 @@
}
if (index < 0) {
- int32_t bandwidthBps;
- if (mHTTPDataSource != NULL
- && mHTTPDataSource->estimateBandwidth(&bandwidthBps)) {
- ALOGV("bandwidth estimated at %.2f kbps", bandwidthBps / 1024.0f);
- } else {
- ALOGV("no bandwidth estimate.");
- return 0; // Pick the lowest bandwidth stream by default.
- }
-
char value[PROPERTY_VALUE_MAX];
if (property_get("media.httplive.max-bw", value, NULL)) {
char *end;
@@ -1039,15 +1277,9 @@
index = mBandwidthItems.size() - 1;
while (index > 0) {
- // consider only 80% of the available bandwidth, but if we are switching up,
- // be even more conservative (70%) to avoid overestimating and immediately
- // switching back.
- size_t adjustedBandwidthBps = bandwidthBps;
- if (index > mCurBandwidthIndex) {
- adjustedBandwidthBps = adjustedBandwidthBps * 7 / 10;
- } else {
- adjustedBandwidthBps = adjustedBandwidthBps * 8 / 10;
- }
+ // be conservative (70%) to avoid overestimating and immediately
+ // switching down again.
+ size_t adjustedBandwidthBps = bandwidthBps * 7 / 10;
if (mBandwidthItems.itemAt(index).mBandwidth <= adjustedBandwidthBps) {
break;
}
@@ -1107,22 +1339,14 @@
return index;
}
-int64_t LiveSession::latestMediaSegmentStartTimeUs() {
- sp<AMessage> audioMeta = mPacketSources.valueFor(STREAMTYPE_AUDIO)->getLatestDequeuedMeta();
- int64_t minSegmentStartTimeUs = -1, videoSegmentStartTimeUs = -1;
- if (audioMeta != NULL) {
- audioMeta->findInt64("segmentStartTimeUs", &minSegmentStartTimeUs);
- }
+HLSTime LiveSession::latestMediaSegmentStartTime() const {
+ HLSTime audioTime(mPacketSources.valueFor(
+ STREAMTYPE_AUDIO)->getLatestDequeuedMeta());
- sp<AMessage> videoMeta = mPacketSources.valueFor(STREAMTYPE_VIDEO)->getLatestDequeuedMeta();
- if (videoMeta != NULL
- && videoMeta->findInt64("segmentStartTimeUs", &videoSegmentStartTimeUs)) {
- if (minSegmentStartTimeUs < 0 || videoSegmentStartTimeUs < minSegmentStartTimeUs) {
- minSegmentStartTimeUs = videoSegmentStartTimeUs;
- }
+ HLSTime videoTime(mPacketSources.valueFor(
+ STREAMTYPE_VIDEO)->getLatestDequeuedMeta());
- }
- return minSegmentStartTimeUs;
+ return audioTime < videoTime ? videoTime : audioTime;
}
status_t LiveSession::onSeek(const sp<AMessage> &msg) {
@@ -1130,7 +1354,7 @@
CHECK(msg->findInt64("timeUs", &timeUs));
if (!mReconfigurationInProgress) {
- changeConfiguration(timeUs, mCurBandwidthIndex);
+ changeConfiguration(timeUs);
return OK;
} else {
return -EWOULDBLOCK;
@@ -1185,8 +1409,7 @@
++mSubtitleGeneration;
status_t err = mPlaylist->selectTrack(index, select);
if (err == OK) {
- sp<AMessage> msg = new AMessage(kWhatChangeConfiguration, id());
- msg->setInt32("bandwidthIndex", mCurBandwidthIndex);
+ sp<AMessage> msg = new AMessage(kWhatChangeConfiguration, this);
msg->setInt32("pickTrack", select);
msg->post();
}
@@ -1201,35 +1424,18 @@
}
}
-bool LiveSession::canSwitchUp() {
- // Allow upwards bandwidth switch when a stream has buffered at least 10 seconds.
- status_t err = OK;
- for (size_t i = 0; i < mPacketSources.size(); ++i) {
- sp<AnotherPacketSource> source = mPacketSources.valueAt(i);
- int64_t dur = source->getBufferedDurationUs(&err);
- if (err == OK && dur > 10000000) {
- return true;
- }
- }
- return false;
-}
-
void LiveSession::changeConfiguration(
- int64_t timeUs, size_t bandwidthIndex, bool pickTrack) {
- // Protect mPacketSources from a swapPacketSource race condition through reconfiguration.
- // (changeConfiguration, onChangeConfiguration2, onChangeConfiguration3).
+ int64_t timeUs, ssize_t bandwidthIndex, bool pickTrack) {
cancelBandwidthSwitch();
CHECK(!mReconfigurationInProgress);
mReconfigurationInProgress = true;
-
- mCurBandwidthIndex = bandwidthIndex;
-
- ALOGV("changeConfiguration => timeUs:%" PRId64 " us, bwIndex:%zu, pickTrack:%d",
- timeUs, bandwidthIndex, pickTrack);
-
- CHECK_LT(bandwidthIndex, mBandwidthItems.size());
- const BandwidthItem &item = mBandwidthItems.itemAt(bandwidthIndex);
+ if (bandwidthIndex >= 0) {
+ mOrigBandwidthIndex = mCurBandwidthIndex;
+ mCurBandwidthIndex = bandwidthIndex;
+ }
+ CHECK_LT(mCurBandwidthIndex, mBandwidthItems.size());
+ const BandwidthItem &item = mBandwidthItems.itemAt(mCurBandwidthIndex);
uint32_t streamMask = 0; // streams that should be fetched by the new fetcher
uint32_t resumeMask = 0; // streams that should be fetched by the original fetcher
@@ -1244,38 +1450,58 @@
// Step 1, stop and discard fetchers that are no longer needed.
// Pause those that we'll reuse.
for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
+ // skip fetchers that are marked mToBeRemoved,
+ // these are done and can't be reused
+ if (mFetcherInfos[i].mToBeRemoved) {
+ continue;
+ }
+
const AString &uri = mFetcherInfos.keyAt(i);
+ sp<PlaylistFetcher> &fetcher = mFetcherInfos.editValueAt(i).mFetcher;
- bool discardFetcher = true;
-
- // If we're seeking all current fetchers are discarded.
- if (timeUs < 0ll) {
- // delay fetcher removal if not picking tracks
- discardFetcher = pickTrack;
-
- for (size_t j = 0; j < kMaxStreams; ++j) {
- StreamType type = indexToType(j);
- if ((streamMask & type) && uri == URIs[j]) {
- resumeMask |= type;
- streamMask &= ~type;
- discardFetcher = false;
- }
+ bool discardFetcher = true, delayRemoval = false;
+ for (size_t j = 0; j < kMaxStreams; ++j) {
+ StreamType type = indexToType(j);
+ if ((streamMask & type) && uri == URIs[j]) {
+ resumeMask |= type;
+ streamMask &= ~type;
+ discardFetcher = false;
}
}
+ // Delay fetcher removal if not picking tracks, AND old fetcher
+ // has stream mask that overlaps new variant. (Okay to discard
+ // old fetcher now, if completely no overlap.)
+ if (discardFetcher && timeUs < 0ll && !pickTrack
+ && (fetcher->getStreamTypeMask() & streamMask)) {
+ discardFetcher = false;
+ delayRemoval = true;
+ }
if (discardFetcher) {
- mFetcherInfos.valueAt(i).mFetcher->stopAsync();
+ fetcher->stopAsync();
} else {
- mFetcherInfos.valueAt(i).mFetcher->pauseAsync();
+ float threshold = -1.0f; // always finish fetching by default
+ if (timeUs >= 0ll) {
+ // seeking, no need to finish fetching
+ threshold = 0.0f;
+ } else if (delayRemoval) {
+ // adapting, abort if remaining of current segment is over threshold
+ threshold = getAbortThreshold(
+ mOrigBandwidthIndex, mCurBandwidthIndex);
+ }
+
+ ALOGV("Pausing with threshold %.3f", threshold);
+
+ fetcher->pauseAsync(threshold);
}
}
sp<AMessage> msg;
if (timeUs < 0ll) {
// skip onChangeConfiguration2 (decoder destruction) if not seeking.
- msg = new AMessage(kWhatChangeConfiguration3, id());
+ msg = new AMessage(kWhatChangeConfiguration3, this);
} else {
- msg = new AMessage(kWhatChangeConfiguration2, id());
+ msg = new AMessage(kWhatChangeConfiguration2, this);
}
msg->setInt32("streamMask", streamMask);
msg->setInt32("resumeMask", resumeMask);
@@ -1296,23 +1522,14 @@
if (mContinuationCounter == 0) {
msg->post();
-
- if (mSeekReplyID != 0) {
- CHECK(mSeekReply != NULL);
- mSeekReply->setInt32("err", OK);
- mSeekReply->postReply(mSeekReplyID);
- mSeekReplyID = 0;
- mSeekReply.clear();
- }
}
}
void LiveSession::onChangeConfiguration(const sp<AMessage> &msg) {
if (!mReconfigurationInProgress) {
- int32_t pickTrack = 0, bandwidthIndex = mCurBandwidthIndex;
+ int32_t pickTrack = 0;
msg->findInt32("pickTrack", &pickTrack);
- msg->findInt32("bandwidthIndex", &bandwidthIndex);
- changeConfiguration(-1ll /* timeUs */, bandwidthIndex, pickTrack);
+ changeConfiguration(-1ll /* timeUs */, -1, pickTrack);
} else {
msg->post(1000000ll); // retry in 1 sec
}
@@ -1323,13 +1540,42 @@
// All fetchers are either suspended or have been removed now.
+ // If we're seeking, clear all packet sources before we report
+ // seek complete, to prevent decoder from pulling stale data.
+ int64_t timeUs;
+ CHECK(msg->findInt64("timeUs", &timeUs));
+
+ if (timeUs >= 0) {
+ mLastSeekTimeUs = timeUs;
+
+ for (size_t i = 0; i < mPacketSources.size(); i++) {
+ mPacketSources.editValueAt(i)->clear();
+ }
+
+ for (size_t i = 0; i < kMaxStreams; ++i) {
+ mStreams[i].mCurDiscontinuitySeq = 0;
+ }
+
+ mDiscontinuityOffsetTimesUs.clear();
+ mDiscontinuityAbsStartTimesUs.clear();
+
+ if (mSeekReplyID != NULL) {
+ CHECK(mSeekReply != NULL);
+ mSeekReply->setInt32("err", OK);
+ mSeekReply->postReply(mSeekReplyID);
+ mSeekReplyID.clear();
+ mSeekReply.clear();
+ }
+
+ // restart buffer polling after seek becauese previous
+ // buffering position is no longer valid.
+ restartPollBuffering();
+ }
+
uint32_t streamMask, resumeMask;
CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask));
CHECK(msg->findInt32("resumeMask", (int32_t *)&resumeMask));
- // currently onChangeConfiguration2 is only called for seeking;
- // remove the following CHECK if using it else where.
- CHECK_EQ(resumeMask, 0);
streamMask |= resumeMask;
AString URIs[kMaxStreams];
@@ -1341,17 +1587,25 @@
}
}
- // Determine which decoders to shutdown on the player side,
- // a decoder has to be shutdown if either
- // 1) its streamtype was active before but now longer isn't.
- // or
- // 2) its streamtype was already active and still is but the URI
- // has changed.
uint32_t changedMask = 0;
for (size_t i = 0; i < kMaxStreams && i != kSubtitleIndex; ++i) {
- if (((mStreamMask & streamMask & indexToType(i))
- && !(URIs[i] == mStreams[i].mUri))
- || (mStreamMask & ~streamMask & indexToType(i))) {
+ // stream URI could change even if onChangeConfiguration2 is only
+ // used for seek. Seek could happen during a bw switch, in this
+ // case bw switch will be cancelled, but the seekTo position will
+ // fetch from the new URI.
+ if ((mStreamMask & streamMask & indexToType(i))
+ && !mStreams[i].mUri.empty()
+ && !(URIs[i] == mStreams[i].mUri)) {
+ ALOGV("stream %zu changed: oldURI %s, newURI %s", i,
+ mStreams[i].mUri.c_str(), URIs[i].c_str());
+ sp<AnotherPacketSource> source = mPacketSources.valueFor(indexToType(i));
+ source->queueDiscontinuity(
+ ATSParser::DISCONTINUITY_FORMATCHANGE, NULL, true);
+ }
+ // Determine which decoders to shutdown on the player side,
+ // a decoder has to be shutdown if its streamtype was active
+ // before but now longer isn't.
+ if ((mStreamMask & ~streamMask & indexToType(i))) {
changedMask |= indexToType(i);
}
}
@@ -1372,7 +1626,7 @@
notify->setInt32("changedMask", changedMask);
msg->setWhat(kWhatChangeConfiguration3);
- msg->setTarget(id());
+ msg->setTarget(this);
notify->setMessage("reply", msg);
notify->post();
@@ -1387,6 +1641,8 @@
CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask));
CHECK(msg->findInt32("resumeMask", (int32_t *)&resumeMask));
+ mNewStreamMask = streamMask | resumeMask;
+
int64_t timeUs;
int32_t pickTrack;
bool switching = false;
@@ -1395,7 +1651,19 @@
if (timeUs < 0ll) {
if (!pickTrack) {
- switching = true;
+ // mSwapMask contains streams that are in both old and new variant,
+ // (in mNewStreamMask & mStreamMask) but with different URIs
+ // (not in resumeMask).
+ // For example, old variant has video and audio in two separate
+ // URIs, and new variant has only audio with unchanged URI. mSwapMask
+ // should be 0 as there is nothing to swap. We only need to stop video,
+ // and resume audio.
+ mSwapMask = mNewStreamMask & mStreamMask & ~resumeMask;
+ switching = (mSwapMask != 0);
+ if (!switching) {
+ ALOGV("#### Finishing Bandwidth Switch Early: %zd => %zd",
+ mOrigBandwidthIndex, mCurBandwidthIndex);
+ }
}
mRealTimeBaseUs = ALooper::GetNowUs() - mLastDequeuedTimeUs;
} else {
@@ -1412,47 +1680,18 @@
}
}
- mNewStreamMask = streamMask | resumeMask;
- if (switching) {
- mSwapMask = mStreamMask & ~resumeMask;
- }
-
// Of all existing fetchers:
// * Resume fetchers that are still needed and assign them original packet sources.
// * Mark otherwise unneeded fetchers for removal.
ALOGV("resuming fetchers for mask 0x%08x", resumeMask);
for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
const AString &uri = mFetcherInfos.keyAt(i);
-
- sp<AnotherPacketSource> sources[kMaxStreams];
- for (size_t j = 0; j < kMaxStreams; ++j) {
- if ((resumeMask & indexToType(j)) && uri == mStreams[j].mUri) {
- sources[j] = mPacketSources.valueFor(indexToType(j));
-
- if (j != kSubtitleIndex) {
- ALOGV("queueing dummy discontinuity for stream type %d", indexToType(j));
- sp<AnotherPacketSource> discontinuityQueue;
- discontinuityQueue = mDiscontinuities.valueFor(indexToType(j));
- discontinuityQueue->queueDiscontinuity(
- ATSParser::DISCONTINUITY_NONE,
- NULL,
- true);
- }
- }
- }
-
- FetcherInfo &info = mFetcherInfos.editValueAt(i);
- if (sources[kAudioIndex] != NULL || sources[kVideoIndex] != NULL
- || sources[kSubtitleIndex] != NULL) {
- info.mFetcher->startAsync(
- sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex]);
- } else {
- info.mToBeRemoved = true;
+ if (!resumeFetcher(uri, resumeMask, timeUs)) {
+ mFetcherInfos.editValueAt(i).mToBeRemoved = true;
}
}
// streamMask now only contains the types that need a new fetcher created.
-
if (streamMask != 0) {
ALOGV("creating new fetchers for mask 0x%08x", streamMask);
}
@@ -1470,13 +1709,12 @@
sp<PlaylistFetcher> fetcher = addFetcher(uri.c_str());
CHECK(fetcher != NULL);
- int64_t startTimeUs = -1;
- int64_t segmentStartTimeUs = -1ll;
- int32_t discontinuitySeq = -1;
+ HLSTime startTime;
+ SeekMode seekMode = kSeekModeExactPosition;
sp<AnotherPacketSource> sources[kMaxStreams];
- if (i == kSubtitleIndex) {
- segmentStartTimeUs = latestMediaSegmentStartTimeUs();
+ if (i == kSubtitleIndex || (!pickTrack && !switching)) {
+ startTime = latestMediaSegmentStartTime();
}
// TRICKY: looping from i as earlier streams are already removed from streamMask
@@ -1486,63 +1724,50 @@
sources[j] = mPacketSources.valueFor(indexToType(j));
if (timeUs >= 0) {
- sources[j]->clear();
- startTimeUs = timeUs;
-
- sp<AnotherPacketSource> discontinuityQueue;
- sp<AMessage> extra = new AMessage;
- extra->setInt64("timeUs", timeUs);
- discontinuityQueue = mDiscontinuities.valueFor(indexToType(j));
- discontinuityQueue->queueDiscontinuity(
- ATSParser::DISCONTINUITY_TIME, extra, true);
+ startTime.mTimeUs = timeUs;
} else {
int32_t type;
sp<AMessage> meta;
- if (pickTrack) {
- // selecting
+ if (!switching) {
+ // selecting, or adapting but no swap required
meta = sources[j]->getLatestDequeuedMeta();
} else {
- // adapting
+ // adapting and swap required
meta = sources[j]->getLatestEnqueuedMeta();
- }
-
- if (meta != NULL && !meta->findInt32("discontinuity", &type)) {
- int64_t tmpUs;
- int64_t tmpSegmentUs;
-
- CHECK(meta->findInt64("timeUs", &tmpUs));
- CHECK(meta->findInt64("segmentStartTimeUs", &tmpSegmentUs));
- if (startTimeUs < 0 || tmpSegmentUs < segmentStartTimeUs) {
- startTimeUs = tmpUs;
- segmentStartTimeUs = tmpSegmentUs;
- } else if (tmpSegmentUs == segmentStartTimeUs && tmpUs < startTimeUs) {
- startTimeUs = tmpUs;
- }
-
- int32_t seq;
- CHECK(meta->findInt32("discontinuitySeq", &seq));
- if (discontinuitySeq < 0 || seq < discontinuitySeq) {
- discontinuitySeq = seq;
+ if (meta != NULL && mCurBandwidthIndex > mOrigBandwidthIndex) {
+ // switching up
+ meta = sources[j]->getMetaAfterLastDequeued(mUpSwitchMargin);
}
}
- if (pickTrack) {
- // selecting track, queue discontinuities before content
+ if (j != kSubtitleIndex && meta != NULL
+ && !meta->findInt32("discontinuity", &type)) {
+ HLSTime tmpTime(meta);
+ if (startTime < tmpTime) {
+ startTime = tmpTime;
+ }
+ }
+
+ if (!switching) {
+ // selecting, or adapting but no swap required
sources[j]->clear();
if (j == kSubtitleIndex) {
break;
}
- sp<AnotherPacketSource> discontinuityQueue;
- discontinuityQueue = mDiscontinuities.valueFor(indexToType(j));
- discontinuityQueue->queueDiscontinuity(
- ATSParser::DISCONTINUITY_FORMATCHANGE, NULL, true);
+
+ ALOGV("stream[%zu]: queue format change", j);
+ sources[j]->queueDiscontinuity(
+ ATSParser::DISCONTINUITY_FORMAT_ONLY, NULL, true);
} else {
- // adapting, queue discontinuities after resume
+ // switching, queue discontinuities after resume
sources[j] = mPacketSources2.valueFor(indexToType(j));
sources[j]->clear();
- uint32_t extraStreams = mNewStreamMask & (~mStreamMask);
- if (extraStreams & indexToType(j)) {
- sources[j]->queueAccessUnit(createFormatChangeBuffer(/*swap*/ false));
+ // the new fetcher might be providing streams that used to be
+ // provided by two different fetchers, if one of the fetcher
+ // paused in the middle while the other somehow paused in next
+ // seg, we have to start from next seg.
+ if (seekMode < mStreams[j].mSeekMode) {
+ seekMode = mStreams[j].mSeekMode;
}
}
}
@@ -1551,54 +1776,89 @@
}
}
+ // Set the target segment start time to the middle point of the
+ // segment where the last sample was.
+ // This gives a better guess if segments of the two variants are not
+ // perfectly aligned. (If the corresponding segment in new variant
+ // starts slightly later than that in the old variant, we still want
+ // to pick that segment, not the one before)
fetcher->startAsync(
sources[kAudioIndex],
sources[kVideoIndex],
sources[kSubtitleIndex],
- startTimeUs < 0 ? mLastSeekTimeUs : startTimeUs,
- segmentStartTimeUs,
- discontinuitySeq,
- switching);
+ startTime.mTimeUs < 0 ? mLastSeekTimeUs : startTime.mTimeUs,
+ startTime.getSegmentTimeUs(true /* midpoint */),
+ startTime.mSeq,
+ seekMode);
}
// All fetchers have now been started, the configuration change
// has completed.
- cancelCheckBandwidthEvent();
- scheduleCheckBandwidthEvent();
-
ALOGV("XXX configuration change completed.");
mReconfigurationInProgress = false;
if (switching) {
mSwitchInProgress = true;
} else {
mStreamMask = mNewStreamMask;
+ mOrigBandwidthIndex = mCurBandwidthIndex;
}
- if (mDisconnectReplyID != 0) {
+ if (mDisconnectReplyID != NULL) {
finishDisconnect();
}
}
-void LiveSession::onSwapped(const sp<AMessage> &msg) {
- int32_t switchGeneration;
- CHECK(msg->findInt32("switchGeneration", &switchGeneration));
- if (switchGeneration != mSwitchGeneration) {
+void LiveSession::swapPacketSource(StreamType stream) {
+ ALOGV("swapPacketSource: stream = %d", stream);
+
+ // transfer packets from source2 to source
+ sp<AnotherPacketSource> &aps = mPacketSources.editValueFor(stream);
+ sp<AnotherPacketSource> &aps2 = mPacketSources2.editValueFor(stream);
+
+ // queue discontinuity in mPacketSource
+ aps->queueDiscontinuity(ATSParser::DISCONTINUITY_FORMAT_ONLY, NULL, false);
+
+ // queue packets in mPacketSource2 to mPacketSource
+ status_t finalResult = OK;
+ sp<ABuffer> accessUnit;
+ while (aps2->hasBufferAvailable(&finalResult) && finalResult == OK &&
+ OK == aps2->dequeueAccessUnit(&accessUnit)) {
+ aps->queueAccessUnit(accessUnit);
+ }
+ aps2->clear();
+}
+
+void LiveSession::tryToFinishBandwidthSwitch(const AString &oldUri) {
+ if (!mSwitchInProgress) {
return;
}
- int32_t stream;
- CHECK(msg->findInt32("stream", &stream));
-
- ssize_t idx = typeToIndex(stream);
- CHECK(idx >= 0);
- if ((mNewStreamMask & stream) && mStreams[idx].mNewUri.empty()) {
- ALOGW("swapping stream type %d %s to empty stream", stream, mStreams[idx].mUri.c_str());
+ ssize_t index = mFetcherInfos.indexOfKey(oldUri);
+ if (index < 0 || !mFetcherInfos[index].mToBeRemoved) {
+ return;
}
- mStreams[idx].mUri = mStreams[idx].mNewUri;
- mStreams[idx].mNewUri.clear();
- mSwapMask &= ~stream;
+ // Swap packet source of streams provided by old variant
+ for (size_t idx = 0; idx < kMaxStreams; idx++) {
+ StreamType stream = indexToType(idx);
+ if ((mSwapMask & stream) && (oldUri == mStreams[idx].mUri)) {
+ swapPacketSource(stream);
+
+ if ((mNewStreamMask & stream) && mStreams[idx].mNewUri.empty()) {
+ ALOGW("swapping stream type %d %s to empty stream",
+ stream, mStreams[idx].mUri.c_str());
+ }
+ mStreams[idx].mUri = mStreams[idx].mNewUri;
+ mStreams[idx].mNewUri.clear();
+
+ mSwapMask &= ~stream;
+ }
+ }
+
+ mFetcherInfos.editValueAt(index).mFetcher->stopAsync(false /* clear */);
+
+ ALOGV("tryToFinishBandwidthSwitch: mSwapMask=%x", mSwapMask);
if (mSwapMask != 0) {
return;
}
@@ -1606,157 +1866,310 @@
// Check if new variant contains extra streams.
uint32_t extraStreams = mNewStreamMask & (~mStreamMask);
while (extraStreams) {
- StreamType extraStream = (StreamType) (extraStreams & ~(extraStreams - 1));
- swapPacketSource(extraStream);
- extraStreams &= ~extraStream;
+ StreamType stream = (StreamType) (extraStreams & ~(extraStreams - 1));
+ extraStreams &= ~stream;
- idx = typeToIndex(extraStream);
+ swapPacketSource(stream);
+
+ ssize_t idx = typeToIndex(stream);
CHECK(idx >= 0);
if (mStreams[idx].mNewUri.empty()) {
ALOGW("swapping extra stream type %d %s to empty stream",
- extraStream, mStreams[idx].mUri.c_str());
+ stream, mStreams[idx].mUri.c_str());
}
mStreams[idx].mUri = mStreams[idx].mNewUri;
mStreams[idx].mNewUri.clear();
}
- tryToFinishBandwidthSwitch();
-}
-
-void LiveSession::onCheckSwitchDown() {
- if (mSwitchDownMonitor == NULL) {
- return;
- }
-
- if (mSwitchInProgress || mReconfigurationInProgress) {
- ALOGV("Switch/Reconfig in progress, defer switch down");
- mSwitchDownMonitor->post(1000000ll);
- return;
- }
-
- for (size_t i = 0; i < kMaxStreams; ++i) {
- int32_t targetDuration;
- sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(indexToType(i));
- sp<AMessage> meta = packetSource->getLatestDequeuedMeta();
-
- if (meta != NULL && meta->findInt32("targetDuration", &targetDuration) ) {
- int64_t bufferedDurationUs = packetSource->getEstimatedDurationUs();
- int64_t targetDurationUs = targetDuration * 1000000ll;
-
- if (bufferedDurationUs < targetDurationUs / 3) {
- (new AMessage(kWhatSwitchDown, id()))->post();
- break;
- }
+ // Restart new fetcher (it was paused after the first 47k block)
+ // and let it fetch into mPacketSources (not mPacketSources2)
+ for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
+ FetcherInfo &info = mFetcherInfos.editValueAt(i);
+ if (info.mToBeResumed) {
+ resumeFetcher(mFetcherInfos.keyAt(i), mNewStreamMask);
+ info.mToBeResumed = false;
}
}
- mSwitchDownMonitor->post(1000000ll);
+ ALOGI("#### Finished Bandwidth Switch: %zd => %zd",
+ mOrigBandwidthIndex, mCurBandwidthIndex);
+
+ mStreamMask = mNewStreamMask;
+ mSwitchInProgress = false;
+ mOrigBandwidthIndex = mCurBandwidthIndex;
+
+ restartPollBuffering();
}
-void LiveSession::onSwitchDown() {
- if (mReconfigurationInProgress || mSwitchInProgress || mCurBandwidthIndex == 0) {
- return;
- }
-
- ssize_t bandwidthIndex = getBandwidthIndex();
- if (bandwidthIndex < mCurBandwidthIndex) {
- changeConfiguration(-1, bandwidthIndex, false);
- return;
- }
-
+void LiveSession::schedulePollBuffering() {
+ sp<AMessage> msg = new AMessage(kWhatPollBuffering, this);
+ msg->setInt32("generation", mPollBufferingGeneration);
+ msg->post(1000000ll);
}
-// Mark switch done when:
-// 1. all old buffers are swapped out
-void LiveSession::tryToFinishBandwidthSwitch() {
+void LiveSession::cancelPollBuffering() {
+ ++mPollBufferingGeneration;
+ mPrevBufferPercentage = -1;
+}
+
+void LiveSession::restartPollBuffering() {
+ cancelPollBuffering();
+ onPollBuffering();
+}
+
+void LiveSession::onPollBuffering() {
+ ALOGV("onPollBuffering: mSwitchInProgress %d, mReconfigurationInProgress %d, "
+ "mInPreparationPhase %d, mCurBandwidthIndex %zd, mStreamMask 0x%x",
+ mSwitchInProgress, mReconfigurationInProgress,
+ mInPreparationPhase, mCurBandwidthIndex, mStreamMask);
+
+ bool underflow, ready, down, up;
+ if (checkBuffering(underflow, ready, down, up)) {
+ if (mInPreparationPhase && ready) {
+ postPrepared(OK);
+ }
+
+ // don't switch before we report prepared
+ if (!mInPreparationPhase) {
+ if (ready) {
+ stopBufferingIfNecessary();
+ } else if (underflow) {
+ startBufferingIfNecessary();
+ }
+ switchBandwidthIfNeeded(up, down);
+ }
+
+ }
+
+ schedulePollBuffering();
+}
+
+void LiveSession::cancelBandwidthSwitch(bool resume) {
+ ALOGV("cancelBandwidthSwitch: mSwitchGen(%d)++, orig %zd, cur %zd",
+ mSwitchGeneration, mOrigBandwidthIndex, mCurBandwidthIndex);
if (!mSwitchInProgress) {
return;
}
- bool needToRemoveFetchers = false;
- for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
- if (mFetcherInfos.valueAt(i).mToBeRemoved) {
- needToRemoveFetchers = true;
- break;
- }
- }
-
- if (!needToRemoveFetchers && mSwapMask == 0) {
- ALOGI("mSwitchInProgress = false");
- mStreamMask = mNewStreamMask;
- mSwitchInProgress = false;
- }
-}
-
-void LiveSession::scheduleCheckBandwidthEvent() {
- sp<AMessage> msg = new AMessage(kWhatCheckBandwidth, id());
- msg->setInt32("generation", mCheckBandwidthGeneration);
- msg->post(10000000ll);
-}
-
-void LiveSession::cancelCheckBandwidthEvent() {
- ++mCheckBandwidthGeneration;
-}
-
-void LiveSession::cancelBandwidthSwitch() {
- Mutex::Autolock lock(mSwapMutex);
- mSwitchGeneration++;
- mSwitchInProgress = false;
- mSwapMask = 0;
-
for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
FetcherInfo& info = mFetcherInfos.editValueAt(i);
if (info.mToBeRemoved) {
info.mToBeRemoved = false;
+ if (resume) {
+ resumeFetcher(mFetcherInfos.keyAt(i), mSwapMask);
+ }
}
}
for (size_t i = 0; i < kMaxStreams; ++i) {
- if (!mStreams[i].mNewUri.empty()) {
- ssize_t j = mFetcherInfos.indexOfKey(mStreams[i].mNewUri);
- if (j < 0) {
- mStreams[i].mNewUri.clear();
+ AString newUri = mStreams[i].mNewUri;
+ if (!newUri.empty()) {
+ // clear all mNewUri matching this newUri
+ for (size_t j = i; j < kMaxStreams; ++j) {
+ if (mStreams[j].mNewUri == newUri) {
+ mStreams[j].mNewUri.clear();
+ }
+ }
+ ALOGV("stopping newUri = %s", newUri.c_str());
+ ssize_t index = mFetcherInfos.indexOfKey(newUri);
+ if (index < 0) {
+ ALOGE("did not find fetcher for newUri: %s", newUri.c_str());
continue;
}
-
- const FetcherInfo &info = mFetcherInfos.valueAt(j);
+ FetcherInfo &info = mFetcherInfos.editValueAt(index);
+ info.mToBeRemoved = true;
info.mFetcher->stopAsync();
- mFetcherInfos.removeItemsAt(j);
- mStreams[i].mNewUri.clear();
}
}
+
+ ALOGI("#### Canceled Bandwidth Switch: %zd => %zd",
+ mCurBandwidthIndex, mOrigBandwidthIndex);
+
+ mSwitchGeneration++;
+ mSwitchInProgress = false;
+ mCurBandwidthIndex = mOrigBandwidthIndex;
+ mSwapMask = 0;
}
-bool LiveSession::canSwitchBandwidthTo(size_t bandwidthIndex) {
- if (mReconfigurationInProgress || mSwitchInProgress) {
+bool LiveSession::checkBuffering(
+ bool &underflow, bool &ready, bool &down, bool &up) {
+ underflow = ready = down = up = false;
+
+ if (mReconfigurationInProgress) {
+ ALOGV("Switch/Reconfig in progress, defer buffer polling");
return false;
}
- if (mCurBandwidthIndex < 0) {
+ size_t activeCount, underflowCount, readyCount, downCount, upCount;
+ activeCount = underflowCount = readyCount = downCount = upCount =0;
+ int32_t minBufferPercent = -1;
+ int64_t durationUs;
+ if (getDuration(&durationUs) != OK) {
+ durationUs = -1;
+ }
+ for (size_t i = 0; i < mPacketSources.size(); ++i) {
+ // we don't check subtitles for buffering level
+ if (!(mStreamMask & mPacketSources.keyAt(i)
+ & (STREAMTYPE_AUDIO | STREAMTYPE_VIDEO))) {
+ continue;
+ }
+ // ignore streams that never had any packet queued.
+ // (it's possible that the variant only has audio or video)
+ sp<AMessage> meta = mPacketSources[i]->getLatestEnqueuedMeta();
+ if (meta == NULL) {
+ continue;
+ }
+
+ int64_t bufferedDurationUs =
+ mPacketSources[i]->getEstimatedDurationUs();
+ ALOGV("source[%zu]: buffered %lld us", i, (long long)bufferedDurationUs);
+ if (durationUs >= 0) {
+ int32_t percent;
+ if (mPacketSources[i]->isFinished(0 /* duration */)) {
+ percent = 100;
+ } else {
+ percent = (int32_t)(100.0 * (mLastDequeuedTimeUs + bufferedDurationUs) / durationUs);
+ }
+ if (minBufferPercent < 0 || percent < minBufferPercent) {
+ minBufferPercent = percent;
+ }
+ }
+
+ ++activeCount;
+ int64_t readyMark = mInPreparationPhase ? kPrepareMarkUs : kReadyMarkUs;
+ if (bufferedDurationUs > readyMark
+ || mPacketSources[i]->isFinished(0)) {
+ ++readyCount;
+ }
+ if (!mPacketSources[i]->isFinished(0)) {
+ if (bufferedDurationUs < kUnderflowMarkUs) {
+ ++underflowCount;
+ }
+ if (bufferedDurationUs > mUpSwitchMark) {
+ ++upCount;
+ }
+ if (bufferedDurationUs < mDownSwitchMark) {
+ ++downCount;
+ }
+ }
+ }
+
+ if (minBufferPercent >= 0) {
+ notifyBufferingUpdate(minBufferPercent);
+ }
+
+ if (activeCount > 0) {
+ up = (upCount == activeCount);
+ down = (downCount > 0);
+ ready = (readyCount == activeCount);
+ underflow = (underflowCount > 0);
return true;
}
- if (bandwidthIndex == (size_t)mCurBandwidthIndex) {
- return false;
- } else if (bandwidthIndex > (size_t)mCurBandwidthIndex) {
- return canSwitchUp();
- } else {
- return true;
+ return false;
+}
+
+void LiveSession::startBufferingIfNecessary() {
+ ALOGV("startBufferingIfNecessary: mInPreparationPhase=%d, mBuffering=%d",
+ mInPreparationPhase, mBuffering);
+ if (!mBuffering) {
+ mBuffering = true;
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatBufferingStart);
+ notify->post();
}
}
-void LiveSession::onCheckBandwidth(const sp<AMessage> &msg) {
- size_t bandwidthIndex = getBandwidthIndex();
- if (canSwitchBandwidthTo(bandwidthIndex)) {
- changeConfiguration(-1ll /* timeUs */, bandwidthIndex);
- } else {
- // Come back and check again 10 seconds later in case there is nothing to do now.
- // If we DO change configuration, once that completes it'll schedule a new
- // check bandwidth event with an incremented mCheckBandwidthGeneration.
- msg->post(10000000ll);
+void LiveSession::stopBufferingIfNecessary() {
+ ALOGV("stopBufferingIfNecessary: mInPreparationPhase=%d, mBuffering=%d",
+ mInPreparationPhase, mBuffering);
+
+ if (mBuffering) {
+ mBuffering = false;
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatBufferingEnd);
+ notify->post();
}
}
+void LiveSession::notifyBufferingUpdate(int32_t percentage) {
+ if (percentage < mPrevBufferPercentage) {
+ percentage = mPrevBufferPercentage;
+ } else if (percentage > 100) {
+ percentage = 100;
+ }
+
+ mPrevBufferPercentage = percentage;
+
+ ALOGV("notifyBufferingUpdate: percentage=%d%%", percentage);
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatBufferingUpdate);
+ notify->setInt32("percentage", percentage);
+ notify->post();
+}
+
+void LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) {
+ // no need to check bandwidth if we only have 1 bandwidth settings
+ if (mSwitchInProgress || mBandwidthItems.size() < 2) {
+ return;
+ }
+
+ int32_t bandwidthBps;
+ if (mBandwidthEstimator->estimateBandwidth(&bandwidthBps)) {
+ ALOGV("bandwidth estimated at %.2f kbps", bandwidthBps / 1024.0f);
+ mLastBandwidthBps = bandwidthBps;
+ } else {
+ ALOGV("no bandwidth estimate.");
+ return;
+ }
+
+ int32_t curBandwidth = mBandwidthItems.itemAt(mCurBandwidthIndex).mBandwidth;
+ // canSwithDown and canSwitchUp can't both be true.
+ // we only want to switch up when measured bw is 120% higher than current variant,
+ // and we only want to switch down when measured bw is below current variant.
+ bool canSwithDown = bufferLow
+ && (bandwidthBps < (int32_t)curBandwidth);
+ bool canSwitchUp = bufferHigh
+ && (bandwidthBps > (int32_t)curBandwidth * 12 / 10);
+
+ if (canSwithDown || canSwitchUp) {
+ ssize_t bandwidthIndex = getBandwidthIndex(bandwidthBps);
+
+ // it's possible that we're checking for canSwitchUp case, but the returned
+ // bandwidthIndex is < mCurBandwidthIndex, as getBandwidthIndex() only uses 70%
+ // of measured bw. In that case we don't want to do anything, since we have
+ // both enough buffer and enough bw.
+ if (bandwidthIndex == mCurBandwidthIndex
+ || (canSwitchUp && bandwidthIndex < mCurBandwidthIndex)
+ || (canSwithDown && bandwidthIndex > mCurBandwidthIndex)) {
+ return;
+ }
+
+ ALOGI("#### Starting Bandwidth Switch: %zd => %zd",
+ mCurBandwidthIndex, bandwidthIndex);
+ changeConfiguration(-1, bandwidthIndex, false);
+ }
+}
+
+void LiveSession::postError(status_t err) {
+ // if we reached EOS, notify buffering of 100%
+ if (err == ERROR_END_OF_STREAM) {
+ notifyBufferingUpdate(100);
+ }
+ // we'll stop buffer polling now, before that notify
+ // stop buffering to stop the spinning icon
+ stopBufferingIfNecessary();
+ cancelPollBuffering();
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatError);
+ notify->setInt32("err", err);
+ notify->post();
+}
+
void LiveSession::postPrepared(status_t err) {
CHECK(mInPreparationPhase);
@@ -1764,6 +2177,8 @@
if (err == OK || err == ERROR_END_OF_STREAM) {
notify->setInt32("what", kWhatPrepared);
} else {
+ cancelPollBuffering();
+
notify->setInt32("what", kWhatPreparationFailed);
notify->setInt32("err", err);
}
@@ -1771,10 +2186,8 @@
notify->post();
mInPreparationPhase = false;
-
- mSwitchDownMonitor = new AMessage(kWhatCheckSwitchDown, id());
- mSwitchDownMonitor->post();
}
+
} // namespace android
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index 2d3a25a..b5e31c9 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -26,6 +26,7 @@
namespace android {
struct ABuffer;
+struct AReplyToken;
struct AnotherPacketSource;
struct DataSource;
struct HTTPBase;
@@ -33,16 +34,13 @@
struct LiveDataSource;
struct M3UParser;
struct PlaylistFetcher;
+struct HLSTime;
struct LiveSession : public AHandler {
enum Flags {
// Don't log any URLs.
kFlagIncognito = 1,
};
- LiveSession(
- const sp<AMessage> ¬ify,
- uint32_t flags,
- const sp<IMediaHTTPService> &httpService);
enum StreamIndex {
kAudioIndex = 0,
@@ -56,10 +54,24 @@
STREAMTYPE_VIDEO = 1 << kVideoIndex,
STREAMTYPE_SUBTITLES = 1 << kSubtitleIndex,
};
+
+ enum SeekMode {
+ kSeekModeExactPosition = 0, // used for seeking
+ kSeekModeNextSample = 1, // used for seamless switching
+ kSeekModeNextSegment = 2, // used for seamless switching
+ };
+
+ LiveSession(
+ const sp<AMessage> ¬ify,
+ uint32_t flags,
+ const sp<IMediaHTTPService> &httpService);
+
status_t dequeueAccessUnit(StreamType stream, sp<ABuffer> *accessUnit);
status_t getStreamFormat(StreamType stream, sp<AMessage> *format);
+ sp<HTTPBase> getHTTPDataSource();
+
void connectAsync(
const char *url,
const KeyedVector<String8, String8> *headers = NULL);
@@ -78,18 +90,18 @@
bool isSeekable() const;
bool hasDynamicDuration() const;
+ static const char *getKeyForStream(StreamType type);
+
enum {
kWhatStreamsChanged,
kWhatError,
kWhatPrepared,
kWhatPreparationFailed,
+ kWhatBufferingStart,
+ kWhatBufferingEnd,
+ kWhatBufferingUpdate,
};
- // create a format-change discontinuity
- //
- // swap:
- // whether is format-change discontinuity should trigger a buffer swap
- sp<ABuffer> createFormatChangeBuffer(bool swap = true);
protected:
virtual ~LiveSession();
@@ -103,18 +115,25 @@
kWhatDisconnect = 'disc',
kWhatSeek = 'seek',
kWhatFetcherNotify = 'notf',
- kWhatCheckBandwidth = 'bndw',
kWhatChangeConfiguration = 'chC0',
kWhatChangeConfiguration2 = 'chC2',
kWhatChangeConfiguration3 = 'chC3',
kWhatFinishDisconnect2 = 'fin2',
- kWhatSwapped = 'swap',
- kWhatCheckSwitchDown = 'ckSD',
- kWhatSwitchDown = 'sDwn',
+ kWhatPollBuffering = 'poll',
};
- static const size_t kBandwidthHistoryBytes;
+ // Bandwidth Switch Mark Defaults
+ static const int64_t kUpSwitchMarkUs;
+ static const int64_t kDownSwitchMarkUs;
+ static const int64_t kUpSwitchMarginUs;
+ static const int64_t kResumeThresholdUs;
+ // Buffer Prepare/Ready/Underflow Marks
+ static const int64_t kReadyMarkUs;
+ static const int64_t kPrepareMarkUs;
+ static const int64_t kUnderflowMarkUs;
+
+ struct BandwidthEstimator;
struct BandwidthItem {
size_t mPlaylistIndex;
unsigned long mBandwidth;
@@ -123,23 +142,22 @@
struct FetcherInfo {
sp<PlaylistFetcher> mFetcher;
int64_t mDurationUs;
- bool mIsPrepared;
bool mToBeRemoved;
+ bool mToBeResumed;
};
struct StreamItem {
const char *mType;
AString mUri, mNewUri;
+ SeekMode mSeekMode;
size_t mCurDiscontinuitySeq;
int64_t mLastDequeuedTimeUs;
int64_t mLastSampleDurationUs;
StreamItem()
- : mType(""),
- mCurDiscontinuitySeq(0),
- mLastDequeuedTimeUs(0),
- mLastSampleDurationUs(0) {}
+ : StreamItem("") {}
StreamItem(const char *type)
: mType(type),
+ mSeekMode(kSeekModeExactPosition),
mCurDiscontinuitySeq(0),
mLastDequeuedTimeUs(0),
mLastSampleDurationUs(0) {}
@@ -155,8 +173,10 @@
uint32_t mFlags;
sp<IMediaHTTPService> mHTTPService;
+ bool mBuffering;
bool mInPreparationPhase;
- bool mBuffering[kMaxStreams];
+ int32_t mPollBufferingGeneration;
+ int32_t mPrevBufferPercentage;
sp<HTTPBase> mHTTPDataSource;
KeyedVector<String8, String8> mExtraHeaders;
@@ -165,9 +185,13 @@
Vector<BandwidthItem> mBandwidthItems;
ssize_t mCurBandwidthIndex;
+ ssize_t mOrigBandwidthIndex;
+ int32_t mLastBandwidthBps;
+ sp<BandwidthEstimator> mBandwidthEstimator;
sp<M3UParser> mPlaylist;
+ sp<ALooper> mFetcherLooper;
KeyedVector<AString, FetcherInfo> mFetcherInfos;
uint32_t mStreamMask;
@@ -180,17 +204,10 @@
// we use this to track reconfiguration progress.
uint32_t mSwapMask;
- KeyedVector<StreamType, sp<AnotherPacketSource> > mDiscontinuities;
KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources;
// A second set of packet sources that buffer content for the variant we're switching to.
KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources2;
- // A mutex used to serialize two sets of events:
- // * the swapping of packet sources in dequeueAccessUnit on the player thread, AND
- // * a forced bandwidth switch termination in cancelSwitch on the live looper.
- Mutex mSwapMutex;
-
- int32_t mCheckBandwidthGeneration;
int32_t mSwitchGeneration;
int32_t mSubtitleGeneration;
@@ -203,13 +220,16 @@
bool mReconfigurationInProgress;
bool mSwitchInProgress;
- uint32_t mDisconnectReplyID;
- uint32_t mSeekReplyID;
+ int64_t mUpSwitchMark;
+ int64_t mDownSwitchMark;
+ int64_t mUpSwitchMargin;
+
+ sp<AReplyToken> mDisconnectReplyID;
+ sp<AReplyToken> mSeekReplyID;
bool mFirstTimeUsValid;
int64_t mFirstTimeUs;
int64_t mLastSeekTimeUs;
- sp<AMessage> mSwitchDownMonitor;
KeyedVector<size_t, int64_t> mDiscontinuityAbsStartTimesUs;
KeyedVector<size_t, int64_t> mDiscontinuityOffsetTimesUs;
@@ -238,45 +258,54 @@
uint32_t block_size = 0,
/* reuse DataSource if doing partial fetch */
sp<DataSource> *source = NULL,
- String8 *actualUrl = NULL);
+ String8 *actualUrl = NULL,
+ /* force connect http even when resuing DataSource */
+ bool forceConnectHTTP = false);
sp<M3UParser> fetchPlaylist(
const char *url, uint8_t *curPlaylistHash, bool *unchanged);
- size_t getBandwidthIndex();
- int64_t latestMediaSegmentStartTimeUs();
+ bool resumeFetcher(
+ const AString &uri, uint32_t streamMask,
+ int64_t timeUs = -1ll, bool newUri = false);
+
+ float getAbortThreshold(
+ ssize_t currentBWIndex, ssize_t targetBWIndex) const;
+ void addBandwidthMeasurement(size_t numBytes, int64_t delayUs);
+ size_t getBandwidthIndex(int32_t bandwidthBps);
+ HLSTime latestMediaSegmentStartTime() const;
static int SortByBandwidth(const BandwidthItem *, const BandwidthItem *);
static StreamType indexToType(int idx);
static ssize_t typeToIndex(int32_t type);
void changeConfiguration(
- int64_t timeUs, size_t bandwidthIndex, bool pickTrack = false);
+ int64_t timeUs, ssize_t bwIndex = -1, bool pickTrack = false);
void onChangeConfiguration(const sp<AMessage> &msg);
void onChangeConfiguration2(const sp<AMessage> &msg);
void onChangeConfiguration3(const sp<AMessage> &msg);
- void onSwapped(const sp<AMessage> &msg);
- void onCheckSwitchDown();
- void onSwitchDown();
- void tryToFinishBandwidthSwitch();
- void scheduleCheckBandwidthEvent();
- void cancelCheckBandwidthEvent();
+ void swapPacketSource(StreamType stream);
+ void tryToFinishBandwidthSwitch(const AString &oldUri);
+ void cancelBandwidthSwitch(bool resume = false);
+ bool checkSwitchProgress(
+ sp<AMessage> &msg, int64_t delayUs, bool *needResumeUntil);
- // cancelBandwidthSwitch is atomic wrt swapPacketSource; call it to prevent packet sources
- // from being swapped out on stale discontinuities while manipulating
- // mPacketSources/mPacketSources2.
- void cancelBandwidthSwitch();
+ void switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow);
- bool canSwitchBandwidthTo(size_t bandwidthIndex);
- void onCheckBandwidth(const sp<AMessage> &msg);
+ void schedulePollBuffering();
+ void cancelPollBuffering();
+ void restartPollBuffering();
+ void onPollBuffering();
+ bool checkBuffering(bool &underflow, bool &ready, bool &down, bool &up);
+ void startBufferingIfNecessary();
+ void stopBufferingIfNecessary();
+ void notifyBufferingUpdate(int32_t percentage);
void finishDisconnect();
void postPrepared(status_t err);
-
- void swapPacketSource(StreamType stream);
- bool canSwitchUp();
+ void postError(status_t err);
DISALLOW_EVIL_CONSTRUCTORS(LiveSession);
};
diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 997b694..7bb7f2ca 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -251,6 +251,7 @@
mIsComplete(false),
mIsEvent(false),
mDiscontinuitySeq(0),
+ mDiscontinuityCount(0),
mSelectedIndex(-1) {
mInitCheck = parse(data, size);
}
@@ -394,7 +395,9 @@
bool M3UParser::getTypeURI(size_t index, const char *key, AString *uri) const {
if (!mIsVariantPlaylist) {
- *uri = mBaseURI;
+ if (uri != NULL) {
+ *uri = mBaseURI;
+ }
// Assume media without any more specific attribute contains
// audio and video, but no subtitles.
@@ -407,7 +410,9 @@
AString groupID;
if (!meta->findString(key, &groupID)) {
- *uri = mItems.itemAt(index).mURI;
+ if (uri != NULL) {
+ *uri = mItems.itemAt(index).mURI;
+ }
AString codecs;
if (!meta->findString("codecs", &codecs)) {
@@ -433,18 +438,26 @@
}
}
- sp<MediaGroup> group = mMediaGroups.valueFor(groupID);
- if (!group->getActiveURI(uri)) {
- return false;
- }
+ // if uri == NULL, we're only checking if the type is present,
+ // don't care about the active URI (or if there is an active one)
+ if (uri != NULL) {
+ sp<MediaGroup> group = mMediaGroups.valueFor(groupID);
+ if (!group->getActiveURI(uri)) {
+ return false;
+ }
- if ((*uri).empty()) {
- *uri = mItems.itemAt(index).mURI;
+ if ((*uri).empty()) {
+ *uri = mItems.itemAt(index).mURI;
+ }
}
return true;
}
+bool M3UParser::hasType(size_t index, const char *key) const {
+ return getTypeURI(index, key, NULL /* uri */);
+}
+
static bool MakeURL(const char *baseURL, const char *url, AString *out) {
out->clear();
@@ -582,6 +595,7 @@
itemMeta = new AMessage;
}
itemMeta->setInt32("discontinuity", true);
+ ++mDiscontinuityCount;
} else if (line.startsWith("#EXT-X-STREAM-INF")) {
if (mMeta != NULL) {
return ERROR_MALFORMED;
@@ -609,6 +623,9 @@
} else if (line.startsWith("#EXT-X-MEDIA")) {
err = parseMedia(line);
} else if (line.startsWith("#EXT-X-DISCONTINUITY-SEQUENCE")) {
+ if (mIsVariantPlaylist) {
+ return ERROR_MALFORMED;
+ }
size_t seq;
err = parseDiscontinuitySequence(line, &seq);
if (err == OK) {
@@ -628,6 +645,8 @@
|| !itemMeta->findInt64("durationUs", &durationUs)) {
return ERROR_MALFORMED;
}
+ itemMeta->setInt32("discontinuity-sequence",
+ mDiscontinuitySeq + mDiscontinuityCount);
}
mItems.push();
@@ -644,6 +663,14 @@
++lineNo;
}
+ // error checking of all fields that's required to appear once
+ // (currently only checking "target-duration")
+ int32_t targetDurationSecs;
+ if (!mIsVariantPlaylist && (mMeta == NULL || !mMeta->findInt32(
+ "target-duration", &targetDurationSecs))) {
+ return ERROR_MALFORMED;
+ }
+
return OK;
}
diff --git a/media/libstagefright/httplive/M3UParser.h b/media/libstagefright/httplive/M3UParser.h
index 1cad060..fef361f 100644
--- a/media/libstagefright/httplive/M3UParser.h
+++ b/media/libstagefright/httplive/M3UParser.h
@@ -50,6 +50,7 @@
ssize_t getSelectedTrack(media_track_type /* type */) const;
bool getTypeURI(size_t index, const char *key, AString *uri) const;
+ bool hasType(size_t index, const char *key) const;
protected:
virtual ~M3UParser();
@@ -70,6 +71,7 @@
bool mIsComplete;
bool mIsEvent;
size_t mDiscontinuitySeq;
+ int32_t mDiscontinuityCount;
sp<AMessage> mMeta;
Vector<Item> mItems;
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 1227600..368612d 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -33,6 +33,7 @@
#include <media/stagefright/foundation/ABitReader.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AUtils.h>
#include <media/stagefright/foundation/hexdump.h>
#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaDefs.h>
@@ -47,11 +48,96 @@
namespace android {
// static
-const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll;
+const int64_t PlaylistFetcher::kMinBufferedDurationUs = 30000000ll;
const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll;
// LCM of 188 (size of a TS packet) & 1k works well
const int32_t PlaylistFetcher::kDownloadBlockSize = 47 * 1024;
-const int32_t PlaylistFetcher::kNumSkipFrames = 5;
+
+struct PlaylistFetcher::DownloadState : public RefBase {
+ DownloadState();
+ void resetState();
+ bool hasSavedState() const;
+ void restoreState(
+ AString &uri,
+ sp<AMessage> &itemMeta,
+ sp<ABuffer> &buffer,
+ sp<ABuffer> &tsBuffer,
+ int32_t &firstSeqNumberInPlaylist,
+ int32_t &lastSeqNumberInPlaylist);
+ void saveState(
+ AString &uri,
+ sp<AMessage> &itemMeta,
+ sp<ABuffer> &buffer,
+ sp<ABuffer> &tsBuffer,
+ int32_t &firstSeqNumberInPlaylist,
+ int32_t &lastSeqNumberInPlaylist);
+
+private:
+ bool mHasSavedState;
+ AString mUri;
+ sp<AMessage> mItemMeta;
+ sp<ABuffer> mBuffer;
+ sp<ABuffer> mTsBuffer;
+ int32_t mFirstSeqNumberInPlaylist;
+ int32_t mLastSeqNumberInPlaylist;
+};
+
+PlaylistFetcher::DownloadState::DownloadState() {
+ resetState();
+}
+
+bool PlaylistFetcher::DownloadState::hasSavedState() const {
+ return mHasSavedState;
+}
+
+void PlaylistFetcher::DownloadState::resetState() {
+ mHasSavedState = false;
+
+ mUri.clear();
+ mItemMeta = NULL;
+ mBuffer = NULL;
+ mTsBuffer = NULL;
+ mFirstSeqNumberInPlaylist = 0;
+ mLastSeqNumberInPlaylist = 0;
+}
+
+void PlaylistFetcher::DownloadState::restoreState(
+ AString &uri,
+ sp<AMessage> &itemMeta,
+ sp<ABuffer> &buffer,
+ sp<ABuffer> &tsBuffer,
+ int32_t &firstSeqNumberInPlaylist,
+ int32_t &lastSeqNumberInPlaylist) {
+ if (!mHasSavedState) {
+ return;
+ }
+
+ uri = mUri;
+ itemMeta = mItemMeta;
+ buffer = mBuffer;
+ tsBuffer = mTsBuffer;
+ firstSeqNumberInPlaylist = mFirstSeqNumberInPlaylist;
+ lastSeqNumberInPlaylist = mLastSeqNumberInPlaylist;
+
+ resetState();
+}
+
+void PlaylistFetcher::DownloadState::saveState(
+ AString &uri,
+ sp<AMessage> &itemMeta,
+ sp<ABuffer> &buffer,
+ sp<ABuffer> &tsBuffer,
+ int32_t &firstSeqNumberInPlaylist,
+ int32_t &lastSeqNumberInPlaylist) {
+ mHasSavedState = true;
+
+ mUri = uri;
+ mItemMeta = itemMeta;
+ mBuffer = buffer;
+ mTsBuffer = tsBuffer;
+ mFirstSeqNumberInPlaylist = firstSeqNumberInPlaylist;
+ mLastSeqNumberInPlaylist = lastSeqNumberInPlaylist;
+}
PlaylistFetcher::PlaylistFetcher(
const sp<AMessage> ¬ify,
@@ -59,7 +145,6 @@
const char *uri,
int32_t subtitleGeneration)
: mNotify(notify),
- mStartTimeUsNotify(notify->dup()),
mSession(session),
mURI(uri),
mStreamTypeMask(0),
@@ -71,18 +156,21 @@
mSeqNumber(-1),
mNumRetries(0),
mStartup(true),
- mAdaptive(false),
- mPrepared(false),
+ mIDRFound(false),
+ mSeekMode(LiveSession::kSeekModeExactPosition),
+ mTimeChangeSignaled(false),
mNextPTSTimeUs(-1ll),
mMonitorQueueGeneration(0),
mSubtitleGeneration(subtitleGeneration),
+ mLastDiscontinuitySeq(-1ll),
mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY),
mFirstPTSValid(false),
- mAbsoluteTimeAnchorUs(0ll),
- mVideoBuffer(new AnotherPacketSource(NULL)) {
+ mFirstTimeUs(-1ll),
+ mVideoBuffer(new AnotherPacketSource(NULL)),
+ mThresholdRatio(-1.0f),
+ mDownloadState(new DownloadState()) {
memset(mPlaylistHash, 0, sizeof(mPlaylistHash));
- mStartTimeUsNotify->setInt32("what", kWhatStartedAt);
- mStartTimeUsNotify->setInt32("streamMask", 0);
+ mHTTPDataSource = mSession->getHTTPDataSource();
}
PlaylistFetcher::~PlaylistFetcher() {
@@ -119,6 +207,32 @@
return segmentStartUs;
}
+int64_t PlaylistFetcher::getSegmentDurationUs(int32_t seqNumber) const {
+ CHECK(mPlaylist != NULL);
+
+ int32_t firstSeqNumberInPlaylist;
+ if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
+ "media-sequence", &firstSeqNumberInPlaylist)) {
+ firstSeqNumberInPlaylist = 0;
+ }
+
+ int32_t lastSeqNumberInPlaylist =
+ firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1;
+
+ CHECK_GE(seqNumber, firstSeqNumberInPlaylist);
+ CHECK_LE(seqNumber, lastSeqNumberInPlaylist);
+
+ int32_t index = seqNumber - firstSeqNumberInPlaylist;
+ sp<AMessage> itemMeta;
+ CHECK(mPlaylist->itemAt(
+ index, NULL /* uri */, &itemMeta));
+
+ int64_t itemDurationUs;
+ CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
+
+ return itemDurationUs;
+}
+
int64_t PlaylistFetcher::delayUsToRefreshPlaylist() const {
int64_t nowUs = ALooper::GetNowUs();
@@ -325,7 +439,7 @@
ALOGV("Need to refresh playlist in %" PRId64 , maxDelayUs);
delayUs = maxDelayUs;
}
- sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id());
+ sp<AMessage> msg = new AMessage(kWhatMonitorQueue, this);
msg->setInt32("generation", mMonitorQueueGeneration);
msg->post(delayUs);
}
@@ -334,6 +448,14 @@
++mMonitorQueueGeneration;
}
+void PlaylistFetcher::setStoppingThreshold(float thresholdRatio) {
+ AutoMutex _l(mThresholdLock);
+ if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) {
+ return;
+ }
+ mThresholdRatio = thresholdRatio;
+}
+
void PlaylistFetcher::startAsync(
const sp<AnotherPacketSource> &audioSource,
const sp<AnotherPacketSource> &videoSource,
@@ -341,8 +463,8 @@
int64_t startTimeUs,
int64_t segmentStartTimeUs,
int32_t startDiscontinuitySeq,
- bool adaptive) {
- sp<AMessage> msg = new AMessage(kWhatStart, id());
+ LiveSession::SeekMode seekMode) {
+ sp<AMessage> msg = new AMessage(kWhatStart, this);
uint32_t streamTypeMask = 0ul;
@@ -365,22 +487,27 @@
msg->setInt64("startTimeUs", startTimeUs);
msg->setInt64("segmentStartTimeUs", segmentStartTimeUs);
msg->setInt32("startDiscontinuitySeq", startDiscontinuitySeq);
- msg->setInt32("adaptive", adaptive);
+ msg->setInt32("seekMode", seekMode);
msg->post();
}
-void PlaylistFetcher::pauseAsync() {
- (new AMessage(kWhatPause, id()))->post();
+void PlaylistFetcher::pauseAsync(float thresholdRatio) {
+ if (thresholdRatio >= 0.0f) {
+ setStoppingThreshold(thresholdRatio);
+ }
+ (new AMessage(kWhatPause, this))->post();
}
void PlaylistFetcher::stopAsync(bool clear) {
- sp<AMessage> msg = new AMessage(kWhatStop, id());
+ setStoppingThreshold(0.0f);
+
+ sp<AMessage> msg = new AMessage(kWhatStop, this);
msg->setInt32("clear", clear);
msg->post();
}
void PlaylistFetcher::resumeUntilAsync(const sp<AMessage> ¶ms) {
- AMessage* msg = new AMessage(kWhatResumeUntil, id());
+ AMessage* msg = new AMessage(kWhatResumeUntil, this);
msg->setMessage("params", params);
msg->post();
}
@@ -404,6 +531,10 @@
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kWhatPaused);
+ notify->setInt32("seekMode",
+ mDownloadState->hasSavedState()
+ ? LiveSession::kSeekModeNextSample
+ : LiveSession::kSeekModeNextSegment);
notify->post();
break;
}
@@ -450,6 +581,10 @@
status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) {
mPacketSources.clear();
+ mStopParams.clear();
+ mStartTimeUsNotify = mNotify->dup();
+ mStartTimeUsNotify->setInt32("what", kWhatStartedAt);
+ mStartTimeUsNotify->setString("uri", mURI);
uint32_t streamTypeMask;
CHECK(msg->findInt32("streamTypeMask", (int32_t *)&streamTypeMask));
@@ -457,11 +592,11 @@
int64_t startTimeUs;
int64_t segmentStartTimeUs;
int32_t startDiscontinuitySeq;
- int32_t adaptive;
+ int32_t seekMode;
CHECK(msg->findInt64("startTimeUs", &startTimeUs));
CHECK(msg->findInt64("segmentStartTimeUs", &segmentStartTimeUs));
CHECK(msg->findInt32("startDiscontinuitySeq", &startDiscontinuitySeq));
- CHECK(msg->findInt32("adaptive", &adaptive));
+ CHECK(msg->findInt32("seekMode", &seekMode));
if (streamTypeMask & LiveSession::STREAMTYPE_AUDIO) {
void *ptr;
@@ -493,14 +628,26 @@
mStreamTypeMask = streamTypeMask;
mSegmentStartTimeUs = segmentStartTimeUs;
- mDiscontinuitySeq = startDiscontinuitySeq;
+
+ if (startDiscontinuitySeq >= 0) {
+ mDiscontinuitySeq = startDiscontinuitySeq;
+ }
+
+ mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY;
+ mSeekMode = (LiveSession::SeekMode) seekMode;
+
+ if (startTimeUs >= 0 || mSeekMode == LiveSession::kSeekModeNextSample) {
+ mStartup = true;
+ mIDRFound = false;
+ mVideoBuffer->clear();
+ }
if (startTimeUs >= 0) {
mStartTimeUs = startTimeUs;
+ mFirstPTSValid = false;
mSeqNumber = -1;
- mStartup = true;
- mPrepared = false;
- mAdaptive = adaptive;
+ mTimeChangeSignaled = false;
+ mDownloadState->resetState();
}
postMonitorQueue();
@@ -510,6 +657,9 @@
void PlaylistFetcher::onPause() {
cancelMonitorQueue();
+ mLastDiscontinuitySeq = mDiscontinuitySeq;
+
+ setStoppingThreshold(-1.0f);
}
void PlaylistFetcher::onStop(const sp<AMessage> &msg) {
@@ -524,8 +674,14 @@
}
}
+ // close off the connection after use
+ mHTTPDataSource->disconnect();
+
+ mDownloadState->resetState();
mPacketSources.clear();
mStreamTypeMask = 0;
+
+ setStoppingThreshold(-1.0f);
}
// Resume until we have reached the boundary timestamps listed in `msg`; when
@@ -535,57 +691,18 @@
sp<AMessage> params;
CHECK(msg->findMessage("params", ¶ms));
- bool stop = false;
- for (size_t i = 0; i < mPacketSources.size(); i++) {
- sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
-
- const char *stopKey;
- int streamType = mPacketSources.keyAt(i);
- switch (streamType) {
- case LiveSession::STREAMTYPE_VIDEO:
- stopKey = "timeUsVideo";
- break;
-
- case LiveSession::STREAMTYPE_AUDIO:
- stopKey = "timeUsAudio";
- break;
-
- case LiveSession::STREAMTYPE_SUBTITLES:
- stopKey = "timeUsSubtitle";
- break;
-
- default:
- TRESPASS();
- }
-
- // Don't resume if we would stop within a resume threshold.
- int32_t discontinuitySeq;
- int64_t latestTimeUs = 0, stopTimeUs = 0;
- sp<AMessage> latestMeta = packetSource->getLatestEnqueuedMeta();
- if (latestMeta != NULL
- && latestMeta->findInt32("discontinuitySeq", &discontinuitySeq)
- && discontinuitySeq == mDiscontinuitySeq
- && latestMeta->findInt64("timeUs", &latestTimeUs)
- && params->findInt64(stopKey, &stopTimeUs)
- && stopTimeUs - latestTimeUs < resumeThreshold(latestMeta)) {
- stop = true;
- }
- }
-
- if (stop) {
- for (size_t i = 0; i < mPacketSources.size(); i++) {
- mPacketSources.valueAt(i)->queueAccessUnit(mSession->createFormatChangeBuffer());
- }
- stopAsync(/* clear = */ false);
- return OK;
- }
-
mStopParams = params;
- postMonitorQueue();
+ onDownloadNext();
return OK;
}
+void PlaylistFetcher::notifyStopReached() {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatStopReached);
+ notify->post();
+}
+
void PlaylistFetcher::notifyError(status_t err) {
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kWhatError);
@@ -605,7 +722,12 @@
void PlaylistFetcher::onMonitorQueue() {
bool downloadMore = false;
- refreshPlaylist();
+
+ // in the middle of an unfinished download, delay
+ // playlist refresh as it'll change seq numbers
+ if (!mDownloadState->hasSavedState()) {
+ refreshPlaylist();
+ }
int32_t targetDurationSecs;
int64_t targetDurationUs = kMinBufferedDurationUs;
@@ -619,27 +741,23 @@
targetDurationUs = targetDurationSecs * 1000000ll;
}
- // buffer at least 3 times the target duration, or up to 10 seconds
- int64_t durationToBufferUs = targetDurationUs * 3;
- if (durationToBufferUs > kMinBufferedDurationUs) {
- durationToBufferUs = kMinBufferedDurationUs;
- }
-
int64_t bufferedDurationUs = 0ll;
- status_t finalResult = NOT_ENOUGH_DATA;
+ status_t finalResult = OK;
if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) {
sp<AnotherPacketSource> packetSource =
mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES);
bufferedDurationUs =
packetSource->getBufferedDurationUs(&finalResult);
- finalResult = OK;
} else {
- // Use max stream duration to prevent us from waiting on a non-existent stream;
- // when we cannot make out from the manifest what streams are included in a playlist
- // we might assume extra streams.
+ // Use min stream duration, but ignore streams that never have any packet
+ // enqueued to prevent us from waiting on a non-existent stream;
+ // when we cannot make out from the manifest what streams are included in
+ // a playlist we might assume extra streams.
+ bufferedDurationUs = -1ll;
for (size_t i = 0; i < mPacketSources.size(); ++i) {
- if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) {
+ if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0
+ || mPacketSources[i]->getLatestEnqueuedMeta() == NULL) {
continue;
}
@@ -647,46 +765,36 @@
mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult);
ALOGV("buffered %" PRId64 " for stream %d",
bufferedStreamDurationUs, mPacketSources.keyAt(i));
- if (bufferedStreamDurationUs > bufferedDurationUs) {
+ if (bufferedDurationUs == -1ll
+ || bufferedStreamDurationUs < bufferedDurationUs) {
bufferedDurationUs = bufferedStreamDurationUs;
}
}
- }
- downloadMore = (bufferedDurationUs < durationToBufferUs);
-
- // signal start if buffered up at least the target size
- if (!mPrepared && bufferedDurationUs > targetDurationUs && downloadMore) {
- mPrepared = true;
-
- ALOGV("prepared, buffered=%" PRId64 " > %" PRId64 "",
- bufferedDurationUs, targetDurationUs);
- sp<AMessage> msg = mNotify->dup();
- msg->setInt32("what", kWhatTemporarilyDoneFetching);
- msg->post();
+ if (bufferedDurationUs == -1ll) {
+ bufferedDurationUs = 0ll;
+ }
}
- if (finalResult == OK && downloadMore) {
+ if (finalResult == OK && bufferedDurationUs < kMinBufferedDurationUs) {
ALOGV("monitoring, buffered=%" PRId64 " < %" PRId64 "",
- bufferedDurationUs, durationToBufferUs);
+ bufferedDurationUs, kMinBufferedDurationUs);
// delay the next download slightly; hopefully this gives other concurrent fetchers
// a better chance to run.
// onDownloadNext();
- sp<AMessage> msg = new AMessage(kWhatDownloadNext, id());
+ sp<AMessage> msg = new AMessage(kWhatDownloadNext, this);
msg->setInt32("generation", mMonitorQueueGeneration);
msg->post(1000l);
} else {
- // Nothing to do yet, try again in a second.
-
- sp<AMessage> msg = mNotify->dup();
- msg->setInt32("what", kWhatTemporarilyDoneFetching);
- msg->post();
-
- int64_t delayUs = mPrepared ? kMaxMonitorDelayUs : targetDurationUs / 2;
+ // We'd like to maintain buffering above durationToBufferUs, so try
+ // again when buffer just about to go below durationToBufferUs
+ // (or after targetDurationUs / 2, whichever is smaller).
+ int64_t delayUs = bufferedDurationUs - kMinBufferedDurationUs + 1000000ll;
+ if (delayUs > targetDurationUs / 2) {
+ delayUs = targetDurationUs / 2;
+ }
ALOGV("pausing for %" PRId64 ", buffered=%" PRId64 " > %" PRId64 "",
- delayUs, bufferedDurationUs, durationToBufferUs);
- // :TRICKY: need to enforce minimum delay because the delay to
- // refresh the playlist will become 0
- postMonitorQueue(delayUs, mPrepared ? targetDurationUs * 2 : 0);
+ delayUs, bufferedDurationUs, kMinBufferedDurationUs);
+ postMonitorQueue(delayUs);
}
}
@@ -715,6 +823,13 @@
if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
updateDuration();
}
+ // Notify LiveSession to use target-duration based buffering level
+ // for up/down switch. Default LiveSession::kUpSwitchMark may not
+ // be reachable for live streams, as our max buffering amount is
+ // limited to 3 segments.
+ if (!mPlaylist->isComplete()) {
+ updateTargetDuration();
+ }
}
mLastPlaylistFetchTimeUs = ALooper::GetNowUs();
@@ -727,10 +842,69 @@
return buffer->size() > 0 && buffer->data()[0] == 0x47;
}
-void PlaylistFetcher::onDownloadNext() {
+bool PlaylistFetcher::shouldPauseDownload() {
+ if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) {
+ // doesn't apply to subtitles
+ return false;
+ }
+
+ // Calculate threshold to abort current download
+ int32_t targetDurationSecs;
+ CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
+ int64_t targetDurationUs = targetDurationSecs * 1000000ll;
+ int64_t thresholdUs = -1;
+ {
+ AutoMutex _l(mThresholdLock);
+ thresholdUs = (mThresholdRatio < 0.0f) ?
+ -1ll : mThresholdRatio * targetDurationUs;
+ }
+
+ if (thresholdUs < 0) {
+ // never abort
+ return false;
+ } else if (thresholdUs == 0) {
+ // immediately abort
+ return true;
+ }
+
+ // now we have a positive thresholdUs, abort if remaining
+ // portion to download is over that threshold.
+ if (mSegmentFirstPTS < 0) {
+ // this means we haven't even find the first access unit,
+ // abort now as we must be very far away from the end.
+ return true;
+ }
+ int64_t lastEnqueueUs = mSegmentFirstPTS;
+ for (size_t i = 0; i < mPacketSources.size(); ++i) {
+ if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) {
+ continue;
+ }
+ sp<AMessage> meta = mPacketSources[i]->getLatestEnqueuedMeta();
+ int32_t type;
+ if (meta == NULL || meta->findInt32("discontinuity", &type)) {
+ continue;
+ }
+ int64_t tmpUs;
+ CHECK(meta->findInt64("timeUs", &tmpUs));
+ if (tmpUs > lastEnqueueUs) {
+ lastEnqueueUs = tmpUs;
+ }
+ }
+ lastEnqueueUs -= mSegmentFirstPTS;
+ if (targetDurationUs - lastEnqueueUs > thresholdUs) {
+ return true;
+ }
+ return false;
+}
+
+bool PlaylistFetcher::initDownloadState(
+ AString &uri,
+ sp<AMessage> &itemMeta,
+ int32_t &firstSeqNumberInPlaylist,
+ int32_t &lastSeqNumberInPlaylist) {
status_t err = refreshPlaylist();
- int32_t firstSeqNumberInPlaylist = 0;
- int32_t lastSeqNumberInPlaylist = 0;
+ firstSeqNumberInPlaylist = 0;
+ lastSeqNumberInPlaylist = 0;
bool discontinuity = false;
if (mPlaylist != NULL) {
@@ -746,6 +920,8 @@
}
}
+ mSegmentFirstPTS = -1ll;
+
if (mPlaylist != NULL && mSeqNumber < 0) {
CHECK_GE(mStartTimeUs, 0ll);
@@ -773,7 +949,8 @@
// timestamps coming from the media container) is used to determine the position
// inside a segments.
mSeqNumber = getSeqNumberForTime(mSegmentStartTimeUs);
- if (mAdaptive) {
+ if (mStreamTypeMask != LiveSession::STREAMTYPE_SUBTITLES
+ && mSeekMode != LiveSession::kSeekModeNextSample) {
// avoid double fetch/decode
mSeqNumber += 1;
}
@@ -823,12 +1000,12 @@
mSeqNumber, firstSeqNumberInPlaylist,
lastSeqNumberInPlaylist, delayUs, mNumRetries);
postMonitorQueue(delayUs);
- return;
+ return false;
}
if (err != OK) {
notifyError(err);
- return;
+ return false;
}
// we've missed the boat, let's start 3 segments prior to the latest sequence
@@ -843,12 +1020,8 @@
// but since the segments we are supposed to fetch have already rolled off
// the playlist, i.e. we have already missed the boat, we inevitably have to
// skip.
- for (size_t i = 0; i < mPacketSources.size(); i++) {
- sp<ABuffer> formatChange = mSession->createFormatChangeBuffer();
- mPacketSources.valueAt(i)->queueAccessUnit(formatChange);
- }
- stopAsync(/* clear = */ false);
- return;
+ notifyStopReached();
+ return false;
}
mSeqNumber = lastSeqNumberInPlaylist - 3;
if (mSeqNumber < firstSeqNumberInPlaylist) {
@@ -858,45 +1031,49 @@
// fall through
} else {
- ALOGE("Cannot find sequence number %d in playlist "
- "(contains %d - %d)",
- mSeqNumber, firstSeqNumberInPlaylist,
- firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1);
+ if (mPlaylist != NULL) {
+ ALOGE("Cannot find sequence number %d in playlist "
+ "(contains %d - %d)",
+ mSeqNumber, firstSeqNumberInPlaylist,
+ firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1);
- notifyError(ERROR_END_OF_STREAM);
- return;
+ notifyError(ERROR_END_OF_STREAM);
+ } else {
+ // It's possible that we were never able to download the playlist.
+ // In this case we should notify error, instead of EOS, as EOS during
+ // prepare means we succeeded in downloading everything.
+ ALOGE("Failed to download playlist!");
+ notifyError(ERROR_IO);
+ }
+
+ return false;
}
}
mNumRetries = 0;
- AString uri;
- sp<AMessage> itemMeta;
CHECK(mPlaylist->itemAt(
mSeqNumber - firstSeqNumberInPlaylist,
&uri,
&itemMeta));
+ CHECK(itemMeta->findInt32("discontinuity-sequence", &mDiscontinuitySeq));
+
int32_t val;
if (itemMeta->findInt32("discontinuity", &val) && val != 0) {
- mDiscontinuitySeq++;
+ discontinuity = true;
+ } else if (mLastDiscontinuitySeq >= 0
+ && mDiscontinuitySeq != mLastDiscontinuitySeq) {
+ // Seek jumped to a new discontinuity sequence. We need to signal
+ // a format change to decoder. Decoder needs to shutdown and be
+ // created again if seamless format change is unsupported.
+ ALOGV("saw discontinuity: mStartup %d, mLastDiscontinuitySeq %d, "
+ "mDiscontinuitySeq %d, mStartTimeUs %lld",
+ mStartup, mLastDiscontinuitySeq, mDiscontinuitySeq, (long long)mStartTimeUs);
discontinuity = true;
}
+ mLastDiscontinuitySeq = -1;
- int64_t range_offset, range_length;
- if (!itemMeta->findInt64("range-offset", &range_offset)
- || !itemMeta->findInt64("range-length", &range_length)) {
- range_offset = 0;
- range_length = -1;
- }
-
- ALOGV("fetching segment %d from (%d .. %d)",
- mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist);
-
- ALOGV("fetching '%s'", uri.c_str());
-
- sp<DataSource> source;
- sp<ABuffer> buffer, tsBuffer;
// decrypt a junk buffer to prefetch key; since a session uses only one http connection,
// this avoids interleaved connections to the key and segment file.
{
@@ -906,16 +1083,122 @@
true /* first */);
if (err != OK) {
notifyError(err);
- return;
+ return false;
}
}
+ if ((mStartup && !mTimeChangeSignaled) || discontinuity) {
+ // We need to signal a time discontinuity to ATSParser on the
+ // first segment after start, or on a discontinuity segment.
+ // Setting mNextPTSTimeUs informs extractAndQueueAccessUnitsXX()
+ // to send the time discontinuity.
+ if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
+ // If this was a live event this made no sense since
+ // we don't have access to all the segment before the current
+ // one.
+ mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber);
+ }
+
+ // Setting mTimeChangeSignaled to true, so that if start time
+ // searching goes into 2nd segment (without a discontinuity),
+ // we don't reset time again. It causes corruption when pending
+ // data in ATSParser is cleared.
+ mTimeChangeSignaled = true;
+ }
+
+ if (discontinuity) {
+ ALOGI("queueing discontinuity (explicit=%d)", discontinuity);
+
+ // Signal a format discontinuity to ATSParser to clear partial data
+ // from previous streams. Not doing this causes bitstream corruption.
+ if (mTSParser != NULL) {
+ mTSParser->signalDiscontinuity(
+ ATSParser::DISCONTINUITY_FORMATCHANGE, NULL /* extra */);
+ }
+
+ queueDiscontinuity(
+ ATSParser::DISCONTINUITY_FORMATCHANGE,
+ NULL /* extra */);
+
+ if (mStartup && mStartTimeUsRelative && mFirstPTSValid) {
+ // This means we guessed mStartTimeUs to be in the previous
+ // segment (likely very close to the end), but either video or
+ // audio has not found start by the end of that segment.
+ //
+ // If this new segment is not a discontinuity, keep searching.
+ //
+ // If this new segment even got a discontinuity marker, just
+ // set mStartTimeUs=0, and take all samples from now on.
+ mStartTimeUs = 0;
+ mFirstPTSValid = false;
+ }
+ }
+
+ ALOGV("fetching segment %d from (%d .. %d)",
+ mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist);
+ return true;
+}
+
+void PlaylistFetcher::onDownloadNext() {
+ AString uri;
+ sp<AMessage> itemMeta;
+ sp<ABuffer> buffer;
+ sp<ABuffer> tsBuffer;
+ int32_t firstSeqNumberInPlaylist = 0;
+ int32_t lastSeqNumberInPlaylist = 0;
+ bool connectHTTP = true;
+
+ if (mDownloadState->hasSavedState()) {
+ mDownloadState->restoreState(
+ uri,
+ itemMeta,
+ buffer,
+ tsBuffer,
+ firstSeqNumberInPlaylist,
+ lastSeqNumberInPlaylist);
+ connectHTTP = false;
+ ALOGV("resuming: '%s'", uri.c_str());
+ } else {
+ if (!initDownloadState(
+ uri,
+ itemMeta,
+ firstSeqNumberInPlaylist,
+ lastSeqNumberInPlaylist)) {
+ return;
+ }
+ ALOGV("fetching: '%s'", uri.c_str());
+ }
+
+ int64_t range_offset, range_length;
+ if (!itemMeta->findInt64("range-offset", &range_offset)
+ || !itemMeta->findInt64("range-length", &range_length)) {
+ range_offset = 0;
+ range_length = -1;
+ }
+
// block-wise download
- bool startup = mStartup;
+ bool shouldPause = false;
ssize_t bytesRead;
do {
+ sp<DataSource> source = mHTTPDataSource;
+
+ int64_t startUs = ALooper::GetNowUs();
bytesRead = mSession->fetchFile(
- uri.c_str(), &buffer, range_offset, range_length, kDownloadBlockSize, &source);
+ uri.c_str(), &buffer, range_offset, range_length, kDownloadBlockSize,
+ &source, NULL, connectHTTP);
+
+ // add sample for bandwidth estimation, excluding samples from subtitles (as
+ // its too small), or during startup/resumeUntil (when we could have more than
+ // one connection open which affects bandwidth)
+ if (!mStartup && mStopParams == NULL && bytesRead > 0
+ && (mStreamTypeMask
+ & (LiveSession::STREAMTYPE_AUDIO
+ | LiveSession::STREAMTYPE_VIDEO))) {
+ int64_t delayUs = ALooper::GetNowUs() - startUs;
+ mSession->addBandwidthMeasurement(bytesRead, delayUs);
+ }
+
+ connectHTTP = false;
if (bytesRead < 0) {
status_t err = bytesRead;
@@ -941,28 +1224,7 @@
return;
}
- if (startup || discontinuity) {
- // Signal discontinuity.
-
- if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
- // If this was a live event this made no sense since
- // we don't have access to all the segment before the current
- // one.
- mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber);
- }
-
- if (discontinuity) {
- ALOGI("queueing discontinuity (explicit=%d)", discontinuity);
-
- queueDiscontinuity(
- ATSParser::DISCONTINUITY_FORMATCHANGE,
- NULL /* extra */);
-
- discontinuity = false;
- }
-
- startup = false;
- }
+ bool startUp = mStartup; // save current start up state
err = OK;
if (bufferStartsWithTsSyncByte(buffer)) {
@@ -976,7 +1238,6 @@
tsBuffer->setRange(tsOff, tsSize);
}
tsBuffer->setRange(tsBuffer->offset(), tsBuffer->size() + bytesRead);
-
err = extractAndQueueAccessUnitsFromTs(tsBuffer);
}
@@ -991,13 +1252,35 @@
return;
} else if (err == ERROR_OUT_OF_RANGE) {
// reached stopping point
- stopAsync(/* clear = */ false);
+ notifyStopReached();
return;
} else if (err != OK) {
notifyError(err);
return;
}
-
+ // If we're switching, post start notification
+ // this should only be posted when the last chunk is full processed by TSParser
+ if (mSeekMode != LiveSession::kSeekModeExactPosition && startUp != mStartup) {
+ CHECK(mStartTimeUsNotify != NULL);
+ mStartTimeUsNotify->post();
+ mStartTimeUsNotify.clear();
+ shouldPause = true;
+ }
+ if (shouldPause || shouldPauseDownload()) {
+ // save state and return if this is not the last chunk,
+ // leaving the fetcher in paused state.
+ if (bytesRead != 0) {
+ mDownloadState->saveState(
+ uri,
+ itemMeta,
+ buffer,
+ tsBuffer,
+ firstSeqNumberInPlaylist,
+ lastSeqNumberInPlaylist);
+ return;
+ }
+ shouldPause = true;
+ }
} while (bytesRead != 0);
if (bufferStartsWithTsSyncByte(buffer)) {
@@ -1034,7 +1317,6 @@
return;
}
- err = OK;
if (tsBuffer != NULL) {
AString method;
CHECK(buffer->meta()->findString("cipher-method", &method));
@@ -1048,30 +1330,40 @@
}
// bulk extract non-ts files
+ bool startUp = mStartup;
if (tsBuffer == NULL) {
- err = extractAndQueueAccessUnits(buffer, itemMeta);
+ status_t err = extractAndQueueAccessUnits(buffer, itemMeta);
if (err == -EAGAIN) {
// starting sequence number too low/high
postMonitorQueue();
return;
} else if (err == ERROR_OUT_OF_RANGE) {
// reached stopping point
- stopAsync(/* clear = */false);
+ notifyStopReached();
+ return;
+ } else if (err != OK) {
+ notifyError(err);
return;
}
}
- if (err != OK) {
- notifyError(err);
- return;
- }
-
++mSeqNumber;
- postMonitorQueue();
+ // if adapting, pause after found the next starting point
+ if (mSeekMode != LiveSession::kSeekModeExactPosition && startUp != mStartup) {
+ CHECK(mStartTimeUsNotify != NULL);
+ mStartTimeUsNotify->post();
+ mStartTimeUsNotify.clear();
+ shouldPause = true;
+ }
+
+ if (!shouldPause) {
+ postMonitorQueue();
+ }
}
-int32_t PlaylistFetcher::getSeqNumberWithAnchorTime(int64_t anchorTimeUs) const {
+int32_t PlaylistFetcher::getSeqNumberWithAnchorTime(
+ int64_t anchorTimeUs, int64_t targetDiffUs) const {
int32_t firstSeqNumberInPlaylist, lastSeqNumberInPlaylist;
if (mPlaylist->meta() == NULL
|| !mPlaylist->meta()->findInt32("media-sequence", &firstSeqNumberInPlaylist)) {
@@ -1080,7 +1372,8 @@
lastSeqNumberInPlaylist = firstSeqNumberInPlaylist + mPlaylist->size() - 1;
int32_t index = mSeqNumber - firstSeqNumberInPlaylist - 1;
- while (index >= 0 && anchorTimeUs > mStartTimeUs) {
+ // adjust anchorTimeUs to within targetDiffUs from mStartTimeUs
+ while (index >= 0 && anchorTimeUs - mStartTimeUs > targetDiffUs) {
sp<AMessage> itemMeta;
CHECK(mPlaylist->itemAt(index, NULL /* uri */, &itemMeta));
@@ -1101,28 +1394,22 @@
int32_t PlaylistFetcher::getSeqNumberForDiscontinuity(size_t discontinuitySeq) const {
int32_t firstSeqNumberInPlaylist;
- if (mPlaylist->meta() == NULL
- || !mPlaylist->meta()->findInt32("media-sequence", &firstSeqNumberInPlaylist)) {
+ if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
+ "media-sequence", &firstSeqNumberInPlaylist)) {
firstSeqNumberInPlaylist = 0;
}
- size_t curDiscontinuitySeq = mPlaylist->getDiscontinuitySeq();
- if (discontinuitySeq < curDiscontinuitySeq) {
- return firstSeqNumberInPlaylist <= 0 ? 0 : (firstSeqNumberInPlaylist - 1);
- }
-
size_t index = 0;
while (index < mPlaylist->size()) {
sp<AMessage> itemMeta;
CHECK(mPlaylist->itemAt( index, NULL /* uri */, &itemMeta));
-
- int64_t discontinuity;
- if (itemMeta->findInt64("discontinuity", &discontinuity)) {
- curDiscontinuitySeq++;
- }
-
+ size_t curDiscontinuitySeq;
+ CHECK(itemMeta->findInt32("discontinuity-sequence", (int32_t *)&curDiscontinuitySeq));
+ int32_t seqNumber = firstSeqNumberInPlaylist + index;
if (curDiscontinuitySeq == discontinuitySeq) {
- return firstSeqNumberInPlaylist + index;
+ return seqNumber;
+ } else if (curDiscontinuitySeq > discontinuitySeq) {
+ return seqNumber <= 0 ? 0 : seqNumber - 1;
}
++index;
@@ -1182,6 +1469,7 @@
accessUnit->meta()->setInt32("discontinuitySeq", mDiscontinuitySeq);
accessUnit->meta()->setInt64("segmentStartTimeUs", getSegmentStartTimeUs(mSeqNumber));
+ accessUnit->meta()->setInt64("segmentDurationUs", getSegmentDurationUs(mSeqNumber));
return accessUnit;
}
@@ -1197,12 +1485,16 @@
// ATSParser from skewing the timestamps of access units.
extra->setInt64(IStreamListener::kKeyMediaTimeUs, 0);
+ // When adapting, signal a recent media time to the parser,
+ // so that PTS wrap around is handled for the new variant.
+ if (mStartTimeUs >= 0 && !mStartTimeUsRelative) {
+ extra->setInt64(IStreamListener::kKeyRecentMediaTimeUs, mStartTimeUs);
+ }
+
mTSParser->signalDiscontinuity(
ATSParser::DISCONTINUITY_TIME, extra);
- mAbsoluteTimeAnchorUs = mNextPTSTimeUs;
mNextPTSTimeUs = -1ll;
- mFirstPTSValid = false;
}
size_t offset = 0;
@@ -1222,30 +1514,15 @@
for (size_t i = mPacketSources.size(); i-- > 0;) {
sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
- const char *key;
- ATSParser::SourceType type;
const LiveSession::StreamType stream = mPacketSources.keyAt(i);
- switch (stream) {
- case LiveSession::STREAMTYPE_VIDEO:
- type = ATSParser::VIDEO;
- key = "timeUsVideo";
- break;
-
- case LiveSession::STREAMTYPE_AUDIO:
- type = ATSParser::AUDIO;
- key = "timeUsAudio";
- break;
-
- case LiveSession::STREAMTYPE_SUBTITLES:
- {
- ALOGE("MPEG2 Transport streams do not contain subtitles.");
- return ERROR_MALFORMED;
- break;
- }
-
- default:
- TRESPASS();
+ if (stream == LiveSession::STREAMTYPE_SUBTITLES) {
+ ALOGE("MPEG2 Transport streams do not contain subtitles.");
+ return ERROR_MALFORMED;
}
+ const char *key = LiveSession::getKeyForStream(stream);
+ ATSParser::SourceType type =
+ (stream == LiveSession::STREAMTYPE_AUDIO) ?
+ ATSParser::AUDIO : ATSParser::VIDEO;
sp<AnotherPacketSource> source =
static_cast<AnotherPacketSource *>(
@@ -1255,110 +1532,112 @@
continue;
}
- int64_t timeUs;
+ const char *mime;
+ sp<MetaData> format = source->getFormat();
+ bool isAvc = format != NULL && format->findCString(kKeyMIMEType, &mime)
+ && !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC);
+
sp<ABuffer> accessUnit;
status_t finalResult;
while (source->hasBufferAvailable(&finalResult)
&& source->dequeueAccessUnit(&accessUnit) == OK) {
+ int64_t timeUs;
CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
+ if (mSegmentFirstPTS < 0ll) {
+ mSegmentFirstPTS = timeUs;
+ if (!mStartTimeUsRelative) {
+ int32_t firstSeqNumberInPlaylist;
+ if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
+ "media-sequence", &firstSeqNumberInPlaylist)) {
+ firstSeqNumberInPlaylist = 0;
+ }
+
+ int32_t targetDurationSecs;
+ CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
+ int64_t targetDurationUs = targetDurationSecs * 1000000ll;
+ // mStartup
+ // mStartup is true until we have queued a packet for all the streams
+ // we are fetching. We queue packets whose timestamps are greater than
+ // mStartTimeUs.
+ // mSegmentStartTimeUs >= 0
+ // mSegmentStartTimeUs is non-negative when adapting or switching tracks
+ // mSeqNumber > firstSeqNumberInPlaylist
+ // don't decrement mSeqNumber if it already points to the 1st segment
+ // timeUs - mStartTimeUs > targetDurationUs:
+ // This and the 2 above conditions should only happen when adapting in a live
+ // stream; the old fetcher has already fetched to mStartTimeUs; the new fetcher
+ // would start fetching after timeUs, which should be greater than mStartTimeUs;
+ // the old fetcher would then continue fetching data until timeUs. We don't want
+ // timeUs to be too far ahead of mStartTimeUs because we want the old fetcher to
+ // stop as early as possible. The definition of being "too far ahead" is
+ // arbitrary; here we use targetDurationUs as threshold.
+ int64_t targetDiffUs = (mSeekMode == LiveSession::kSeekModeNextSample
+ ? 0 : targetDurationUs);
+ if (mStartup && mSegmentStartTimeUs >= 0
+ && mSeqNumber > firstSeqNumberInPlaylist
+ && timeUs - mStartTimeUs > targetDiffUs) {
+ // we just guessed a starting timestamp that is too high when adapting in a
+ // live stream; re-adjust based on the actual timestamp extracted from the
+ // media segment; if we didn't move backward after the re-adjustment
+ // (newSeqNumber), start at least 1 segment prior.
+ int32_t newSeqNumber = getSeqNumberWithAnchorTime(
+ timeUs, targetDiffUs);
+ if (newSeqNumber >= mSeqNumber) {
+ --mSeqNumber;
+ } else {
+ mSeqNumber = newSeqNumber;
+ }
+ mStartTimeUsNotify = mNotify->dup();
+ mStartTimeUsNotify->setInt32("what", kWhatStartedAt);
+ mStartTimeUsNotify->setString("uri", mURI);
+ mIDRFound = false;
+ return -EAGAIN;
+ }
+ }
+ }
if (mStartup) {
if (!mFirstPTSValid) {
mFirstTimeUs = timeUs;
mFirstPTSValid = true;
}
+ bool startTimeReached = true;
if (mStartTimeUsRelative) {
timeUs -= mFirstTimeUs;
if (timeUs < 0) {
timeUs = 0;
}
+ startTimeReached = (timeUs >= mStartTimeUs);
}
- if (timeUs < mStartTimeUs) {
- // buffer up to the closest preceding IDR frame
- ALOGV("timeUs %" PRId64 " us < mStartTimeUs %" PRId64 " us",
- timeUs, mStartTimeUs);
- const char *mime;
- sp<MetaData> format = source->getFormat();
- bool isAvc = false;
- if (format != NULL && format->findCString(kKeyMIMEType, &mime)
- && !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC)) {
- isAvc = true;
- }
- if (isAvc && IsIDR(accessUnit)) {
- mVideoBuffer->clear();
- }
+ if (!startTimeReached || (isAvc && !mIDRFound)) {
+ // buffer up to the closest preceding IDR frame in the next segement,
+ // or the closest succeeding IDR frame after the exact position
if (isAvc) {
- mVideoBuffer->queueAccessUnit(accessUnit);
+ if (IsIDR(accessUnit)) {
+ mVideoBuffer->clear();
+ mIDRFound = true;
+ }
+ if (mIDRFound && mStartTimeUsRelative && !startTimeReached) {
+ mVideoBuffer->queueAccessUnit(accessUnit);
+ }
}
-
- continue;
+ if (!startTimeReached || (isAvc && !mIDRFound)) {
+ continue;
+ }
}
}
- CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
- if (mStartTimeUsNotify != NULL && timeUs > mStartTimeUs) {
- int32_t firstSeqNumberInPlaylist;
- if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
- "media-sequence", &firstSeqNumberInPlaylist)) {
- firstSeqNumberInPlaylist = 0;
- }
-
- int32_t targetDurationSecs;
- CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
- int64_t targetDurationUs = targetDurationSecs * 1000000ll;
- // mStartup
- // mStartup is true until we have queued a packet for all the streams
- // we are fetching. We queue packets whose timestamps are greater than
- // mStartTimeUs.
- // mSegmentStartTimeUs >= 0
- // mSegmentStartTimeUs is non-negative when adapting or switching tracks
- // mSeqNumber > firstSeqNumberInPlaylist
- // don't decrement mSeqNumber if it already points to the 1st segment
- // timeUs - mStartTimeUs > targetDurationUs:
- // This and the 2 above conditions should only happen when adapting in a live
- // stream; the old fetcher has already fetched to mStartTimeUs; the new fetcher
- // would start fetching after timeUs, which should be greater than mStartTimeUs;
- // the old fetcher would then continue fetching data until timeUs. We don't want
- // timeUs to be too far ahead of mStartTimeUs because we want the old fetcher to
- // stop as early as possible. The definition of being "too far ahead" is
- // arbitrary; here we use targetDurationUs as threshold.
- if (mStartup && mSegmentStartTimeUs >= 0
- && mSeqNumber > firstSeqNumberInPlaylist
- && timeUs - mStartTimeUs > targetDurationUs) {
- // we just guessed a starting timestamp that is too high when adapting in a
- // live stream; re-adjust based on the actual timestamp extracted from the
- // media segment; if we didn't move backward after the re-adjustment
- // (newSeqNumber), start at least 1 segment prior.
- int32_t newSeqNumber = getSeqNumberWithAnchorTime(timeUs);
- if (newSeqNumber >= mSeqNumber) {
- --mSeqNumber;
- } else {
- mSeqNumber = newSeqNumber;
- }
- mStartTimeUsNotify = mNotify->dup();
- mStartTimeUsNotify->setInt32("what", kWhatStartedAt);
- return -EAGAIN;
- }
-
- int32_t seq;
- if (!mStartTimeUsNotify->findInt32("discontinuitySeq", &seq)) {
- mStartTimeUsNotify->setInt32("discontinuitySeq", mDiscontinuitySeq);
- }
- int64_t startTimeUs;
- if (!mStartTimeUsNotify->findInt64(key, &startTimeUs)) {
- mStartTimeUsNotify->setInt64(key, timeUs);
-
- uint32_t streamMask = 0;
- mStartTimeUsNotify->findInt32("streamMask", (int32_t *) &streamMask);
+ if (mStartTimeUsNotify != NULL) {
+ uint32_t streamMask = 0;
+ mStartTimeUsNotify->findInt32("streamMask", (int32_t *) &streamMask);
+ if (!(streamMask & mPacketSources.keyAt(i))) {
streamMask |= mPacketSources.keyAt(i);
mStartTimeUsNotify->setInt32("streamMask", streamMask);
if (streamMask == mStreamTypeMask) {
mStartup = false;
- mStartTimeUsNotify->post();
- mStartTimeUsNotify.clear();
}
}
}
@@ -1372,7 +1651,6 @@
|| !mStopParams->findInt64(key, &stopTimeUs)
|| (discontinuitySeq == mDiscontinuitySeq
&& timeUs >= stopTimeUs)) {
- packetSource->queueAccessUnit(mSession->createFormatChangeBuffer());
mStreamTypeMask &= ~stream;
mPacketSources.removeItemsAt(i);
break;
@@ -1467,8 +1745,6 @@
}
if (mNextPTSTimeUs >= 0ll) {
- mFirstPTSValid = false;
- mAbsoluteTimeAnchorUs = mNextPTSTimeUs;
mNextPTSTimeUs = -1ll;
}
@@ -1569,7 +1845,7 @@
CHECK(packetSource->getFormat()->findInt32(kKeySampleRate, &sampleRate));
int64_t timeUs = (PTS * 100ll) / 9ll;
- if (!mFirstPTSValid) {
+ if (mStartup && !mFirstPTSValid) {
mFirstPTSValid = true;
mFirstTimeUs = timeUs;
}
@@ -1621,10 +1897,13 @@
CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
int64_t targetDurationUs = targetDurationSecs * 1000000ll;
+ int64_t targetDiffUs =(mSeekMode == LiveSession::kSeekModeNextSample
+ ? 0 : targetDurationUs);
// Duplicated logic from how we handle .ts playlists.
if (mStartup && mSegmentStartTimeUs >= 0
- && timeUs - mStartTimeUs > targetDurationUs) {
- int32_t newSeqNumber = getSeqNumberWithAnchorTime(timeUs);
+ && timeUs - mStartTimeUs > targetDiffUs) {
+ int32_t newSeqNumber = getSeqNumberWithAnchorTime(
+ timeUs, targetDiffUs);
if (newSeqNumber >= mSeqNumber) {
--mSeqNumber;
} else {
@@ -1633,11 +1912,7 @@
return -EAGAIN;
}
- mStartTimeUsNotify->setInt64("timeUsAudio", timeUs);
- mStartTimeUsNotify->setInt32("discontinuitySeq", mDiscontinuitySeq);
mStartTimeUsNotify->setInt32("streamMask", LiveSession::STREAMTYPE_AUDIO);
- mStartTimeUsNotify->post();
- mStartTimeUsNotify.clear();
mStartup = false;
}
}
@@ -1650,7 +1925,6 @@
|| discontinuitySeq > mDiscontinuitySeq
|| !mStopParams->findInt64("timeUsAudio", &stopTimeUs)
|| (discontinuitySeq == mDiscontinuitySeq && unitTimeUs >= stopTimeUs)) {
- packetSource->queueAccessUnit(mSession->createFormatChangeBuffer());
mStreamTypeMask = 0;
mPacketSources.clear();
return ERROR_OUT_OF_RANGE;
@@ -1687,33 +1961,15 @@
msg->post();
}
-int64_t PlaylistFetcher::resumeThreshold(const sp<AMessage> &msg) {
- int64_t durationUs;
- if (msg->findInt64("durationUs", &durationUs) && durationUs > 0) {
- return kNumSkipFrames * durationUs;
- }
+void PlaylistFetcher::updateTargetDuration() {
+ int32_t targetDurationSecs;
+ CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
+ int64_t targetDurationUs = targetDurationSecs * 1000000ll;
- sp<RefBase> obj;
- msg->findObject("format", &obj);
- MetaData *format = static_cast<MetaData *>(obj.get());
-
- const char *mime;
- CHECK(format->findCString(kKeyMIMEType, &mime));
- bool audio = !strncasecmp(mime, "audio/", 6);
- if (audio) {
- // Assumes 1000 samples per frame.
- int32_t sampleRate;
- CHECK(format->findInt32(kKeySampleRate, &sampleRate));
- return kNumSkipFrames /* frames */ * 1000 /* samples */
- * (1000000 / sampleRate) /* sample duration (us) */;
- } else {
- int32_t frameRate;
- if (format->findInt32(kKeyFrameRate, &frameRate) && frameRate > 0) {
- return kNumSkipFrames * (1000000 / frameRate);
- }
- }
-
- return 500000ll;
+ sp<AMessage> msg = mNotify->dup();
+ msg->setInt32("what", kWhatTargetDurationUpdate);
+ msg->setInt64("targetDurationUs", targetDurationUs);
+ msg->post();
}
} // namespace android
diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h
index 4e15f85..dab56df 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.h
+++ b/media/libstagefright/httplive/PlaylistFetcher.h
@@ -36,6 +36,7 @@
struct PlaylistFetcher : public AHandler {
static const int64_t kMinBufferedDurationUs;
static const int32_t kDownloadBlockSize;
+ static const int64_t kFetcherResumeThreshold;
enum {
kWhatStarted,
@@ -43,10 +44,11 @@
kWhatStopped,
kWhatError,
kWhatDurationUpdate,
- kWhatTemporarilyDoneFetching,
+ kWhatTargetDurationUpdate,
kWhatPrepared,
kWhatPreparationFailed,
kWhatStartedAt,
+ kWhatStopReached,
};
PlaylistFetcher(
@@ -64,10 +66,10 @@
int64_t startTimeUs = -1ll, // starting timestamps
int64_t segmentStartTimeUs = -1ll, // starting position within playlist
// startTimeUs!=segmentStartTimeUs only when playlist is live
- int32_t startDiscontinuitySeq = 0,
- bool adaptive = false);
+ int32_t startDiscontinuitySeq = -1,
+ LiveSession::SeekMode seekMode = LiveSession::kSeekModeExactPosition);
- void pauseAsync();
+ void pauseAsync(float thresholdRatio);
void stopAsync(bool clear = true);
@@ -95,6 +97,8 @@
kWhatDownloadNext = 'dlnx',
};
+ struct DownloadState;
+
static const int64_t kMaxMonitorDelayUs;
static const int32_t kNumSkipFrames;
@@ -105,6 +109,7 @@
sp<AMessage> mNotify;
sp<AMessage> mStartTimeUsNotify;
+ sp<HTTPBase> mHTTPDataSource;
sp<LiveSession> mSession;
AString mURI;
@@ -116,7 +121,7 @@
// adapting or switching tracks.
int64_t mSegmentStartTimeUs;
- ssize_t mDiscontinuitySeq;
+ int32_t mDiscontinuitySeq;
bool mStartTimeUsRelative;
sp<AMessage> mStopParams; // message containing the latest timestamps we should fetch.
@@ -130,13 +135,16 @@
int32_t mSeqNumber;
int32_t mNumRetries;
bool mStartup;
- bool mAdaptive;
- bool mPrepared;
+ bool mIDRFound;
+ int32_t mSeekMode;
+ bool mTimeChangeSignaled;
int64_t mNextPTSTimeUs;
int32_t mMonitorQueueGeneration;
const int32_t mSubtitleGeneration;
+ int32_t mLastDiscontinuitySeq;
+
enum RefreshState {
INITIAL_MINIMUM_RELOAD_DELAY,
FIRST_UNCHANGED_RELOAD_ATTEMPT,
@@ -150,9 +158,8 @@
sp<ATSParser> mTSParser;
bool mFirstPTSValid;
- uint64_t mFirstPTS;
int64_t mFirstTimeUs;
- int64_t mAbsoluteTimeAnchorUs;
+ int64_t mSegmentFirstPTS;
sp<AnotherPacketSource> mVideoBuffer;
// Stores the initialization vector to decrypt the next block of cipher text, which can
@@ -160,6 +167,11 @@
// the last block of cipher text (cipher-block chaining).
unsigned char mAESInitVec[16];
+ Mutex mThresholdLock;
+ float mThresholdRatio;
+
+ sp<DownloadState> mDownloadState;
+
// Set first to true if decrypting the first segment of a playlist segment. When
// first is true, reset the initialization vector based on the available
// information in the manifest; otherwise, use the initialization vector as
@@ -175,6 +187,8 @@
void postMonitorQueue(int64_t delayUs = 0, int64_t minDelayUs = 0);
void cancelMonitorQueue();
+ void setStoppingThreshold(float thresholdRatio);
+ bool shouldPauseDownload();
int64_t delayUsToRefreshPlaylist() const;
status_t refreshPlaylist();
@@ -182,12 +196,19 @@
// Returns the media time in us of the segment specified by seqNumber.
// This is computed by summing the durations of all segments before it.
int64_t getSegmentStartTimeUs(int32_t seqNumber) const;
+ // Returns the duration time in us of the segment specified.
+ int64_t getSegmentDurationUs(int32_t seqNumber) const;
status_t onStart(const sp<AMessage> &msg);
void onPause();
void onStop(const sp<AMessage> &msg);
void onMonitorQueue();
void onDownloadNext();
+ bool initDownloadState(
+ AString &uri,
+ sp<AMessage> &itemMeta,
+ int32_t &firstSeqNumberInPlaylist,
+ int32_t &lastSeqNumberInPlaylist);
// Resume a fetcher to continue until the stopping point stored in msg.
status_t onResumeUntil(const sp<AMessage> &msg);
@@ -201,20 +222,19 @@
status_t extractAndQueueAccessUnits(
const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta);
+ void notifyStopReached();
void notifyError(status_t err);
void queueDiscontinuity(
ATSParser::DiscontinuityType type, const sp<AMessage> &extra);
- int32_t getSeqNumberWithAnchorTime(int64_t anchorTimeUs) const;
+ int32_t getSeqNumberWithAnchorTime(
+ int64_t anchorTimeUs, int64_t targetDurationUs) const;
int32_t getSeqNumberForDiscontinuity(size_t discontinuitySeq) const;
int32_t getSeqNumberForTime(int64_t timeUs) const;
void updateDuration();
-
- // Before resuming a fetcher in onResume, check the remaining duration is longer than that
- // returned by resumeThreshold.
- int64_t resumeThreshold(const sp<AMessage> &msg);
+ void updateTargetDuration();
DISALLOW_EVIL_CONSTRUCTORS(PlaylistFetcher);
};
diff --git a/media/libstagefright/include/MPEG4Extractor.h b/media/libstagefright/include/MPEG4Extractor.h
index 1fe6fcf..8c16251 100644
--- a/media/libstagefright/include/MPEG4Extractor.h
+++ b/media/libstagefright/include/MPEG4Extractor.h
@@ -83,6 +83,8 @@
Vector<SidxEntry> mSidxEntries;
off64_t mMoofOffset;
+ bool mMoofFound;
+ bool mMdatFound;
Vector<PsshInfo> mPssh;
diff --git a/media/libstagefright/include/avc_utils.h b/media/libstagefright/include/avc_utils.h
index c270bc1..dafa07e 100644
--- a/media/libstagefright/include/avc_utils.h
+++ b/media/libstagefright/include/avc_utils.h
@@ -36,6 +36,11 @@
kAVCProfileCAVLC444Intra = 0x2c
};
+struct NALPosition {
+ size_t nalOffset;
+ size_t nalSize;
+};
+
// Optionally returns sample aspect ratio as well.
void FindAVCDimensions(
const sp<ABuffer> &seqParamSet,
diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 482ccff..0a868bc 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -35,6 +35,7 @@
#include <media/stagefright/Utils.h>
#include <media/IStreamSource.h>
#include <utils/KeyedVector.h>
+#include <utils/Vector.h>
#include <inttypes.h>
@@ -47,7 +48,8 @@
static const size_t kTSPacketSize = 188;
struct ATSParser::Program : public RefBase {
- Program(ATSParser *parser, unsigned programNumber, unsigned programMapPID);
+ Program(ATSParser *parser, unsigned programNumber, unsigned programMapPID,
+ int64_t lastRecoveredPTS);
bool parsePSISection(
unsigned pid, ABitReader *br, status_t *err);
@@ -86,14 +88,22 @@
}
private:
+ struct StreamInfo {
+ unsigned mType;
+ unsigned mPID;
+ };
+
ATSParser *mParser;
unsigned mProgramNumber;
unsigned mProgramMapPID;
KeyedVector<unsigned, sp<Stream> > mStreams;
bool mFirstPTSValid;
uint64_t mFirstPTS;
+ int64_t mLastRecoveredPTS;
status_t parseProgramMap(ABitReader *br);
+ int64_t recoverPTS(uint64_t PTS_33bit);
+ bool switchPIDs(const Vector<StreamInfo> &infos);
DISALLOW_EVIL_CONSTRUCTORS(Program);
};
@@ -157,10 +167,12 @@
PSISection();
status_t append(const void *data, size_t size);
+ void setSkipBytes(uint8_t skip);
void clear();
bool isComplete() const;
bool isEmpty() const;
+ bool isCRCOkay() const;
const uint8_t *data() const;
size_t size() const;
@@ -170,6 +182,8 @@
private:
sp<ABuffer> mBuffer;
+ uint8_t mSkipBytes;
+ static uint32_t CRC_TABLE[];
DISALLOW_EVIL_CONSTRUCTORS(PSISection);
};
@@ -177,12 +191,14 @@
////////////////////////////////////////////////////////////////////////////////
ATSParser::Program::Program(
- ATSParser *parser, unsigned programNumber, unsigned programMapPID)
+ ATSParser *parser, unsigned programNumber, unsigned programMapPID,
+ int64_t lastRecoveredPTS)
: mParser(parser),
mProgramNumber(programNumber),
mProgramMapPID(programMapPID),
mFirstPTSValid(false),
- mFirstPTS(0) {
+ mFirstPTS(0),
+ mLastRecoveredPTS(lastRecoveredPTS) {
ALOGV("new program number %u", programNumber);
}
@@ -237,10 +253,71 @@
}
}
-struct StreamInfo {
- unsigned mType;
- unsigned mPID;
-};
+bool ATSParser::Program::switchPIDs(const Vector<StreamInfo> &infos) {
+ bool success = false;
+
+ if (mStreams.size() == infos.size()) {
+ // build type->PIDs map for old and new mapping
+ size_t i;
+ KeyedVector<int32_t, Vector<int32_t> > oldType2PIDs, newType2PIDs;
+ for (i = 0; i < mStreams.size(); ++i) {
+ ssize_t index = oldType2PIDs.indexOfKey(mStreams[i]->type());
+ if (index < 0) {
+ oldType2PIDs.add(mStreams[i]->type(), Vector<int32_t>());
+ }
+ oldType2PIDs.editValueFor(mStreams[i]->type()).push_back(mStreams[i]->pid());
+ }
+ for (i = 0; i < infos.size(); ++i) {
+ ssize_t index = newType2PIDs.indexOfKey(infos[i].mType);
+ if (index < 0) {
+ newType2PIDs.add(infos[i].mType, Vector<int32_t>());
+ }
+ newType2PIDs.editValueFor(infos[i].mType).push_back(infos[i].mPID);
+ }
+
+ // we can recover if the number of streams for each type hasn't changed
+ if (oldType2PIDs.size() == newType2PIDs.size()) {
+ success = true;
+ for (i = 0; i < oldType2PIDs.size(); ++i) {
+ // KeyedVector is sorted, we just compare key and size of each index
+ if (oldType2PIDs.keyAt(i) != newType2PIDs.keyAt(i)
+ || oldType2PIDs[i].size() != newType2PIDs[i].size()) {
+ success = false;
+ break;
+ }
+ }
+ }
+
+ if (success) {
+ // save current streams to temp
+ KeyedVector<int32_t, sp<Stream> > temp;
+ for (i = 0; i < mStreams.size(); ++i) {
+ temp.add(mStreams.keyAt(i), mStreams.editValueAt(i));
+ }
+
+ mStreams.clear();
+ for (i = 0; i < temp.size(); ++i) {
+ // The two checks below shouldn't happen,
+ // we already checked above the stream count matches
+ ssize_t index = newType2PIDs.indexOfKey(temp[i]->type());
+ CHECK(index >= 0);
+ Vector<int32_t> &newPIDs = newType2PIDs.editValueAt(index);
+ CHECK(newPIDs.size() > 0);
+
+ // get the next PID for temp[i]->type() in the new PID map
+ Vector<int32_t>::iterator it = newPIDs.begin();
+
+ // change the PID of the stream, and add it back
+ temp.editValueAt(i)->setPID(*it);
+ mStreams.add(temp[i]->pid(), temp.editValueAt(i));
+
+ // removed the used PID
+ newPIDs.erase(it);
+ }
+ }
+ }
+ return success;
+}
status_t ATSParser::Program::parseProgramMap(ABitReader *br) {
unsigned table_id = br->getBits(8);
@@ -369,39 +446,8 @@
}
#endif
- // The only case we can recover from is if we have two streams
- // and they switched PIDs.
-
- bool success = false;
-
- if (mStreams.size() == 2 && infos.size() == 2) {
- const StreamInfo &info1 = infos.itemAt(0);
- const StreamInfo &info2 = infos.itemAt(1);
-
- sp<Stream> s1 = mStreams.editValueAt(0);
- sp<Stream> s2 = mStreams.editValueAt(1);
-
- bool caseA =
- info1.mPID == s1->pid() && info1.mType == s2->type()
- && info2.mPID == s2->pid() && info2.mType == s1->type();
-
- bool caseB =
- info1.mPID == s2->pid() && info1.mType == s1->type()
- && info2.mPID == s1->pid() && info2.mType == s2->type();
-
- if (caseA || caseB) {
- unsigned pid1 = s1->pid();
- unsigned pid2 = s2->pid();
- s1->setPID(pid2);
- s2->setPID(pid1);
-
- mStreams.clear();
- mStreams.add(s1->pid(), s1);
- mStreams.add(s2->pid(), s2);
-
- success = true;
- }
- }
+ // we can recover if number of streams for each type remain the same
+ bool success = switchPIDs(infos);
if (!success) {
ALOGI("Stream PIDs changed and we cannot recover.");
@@ -425,6 +471,32 @@
return OK;
}
+int64_t ATSParser::Program::recoverPTS(uint64_t PTS_33bit) {
+ // We only have the lower 33-bit of the PTS. It could overflow within a
+ // reasonable amount of time. To handle the wrap-around, use fancy math
+ // to get an extended PTS that is within [-0xffffffff, 0xffffffff]
+ // of the latest recovered PTS.
+ if (mLastRecoveredPTS < 0ll) {
+ // Use the original 33bit number for 1st frame, the reason is that
+ // if 1st frame wraps to negative that's far away from 0, we could
+ // never start. Only start wrapping around from 2nd frame.
+ mLastRecoveredPTS = static_cast<int64_t>(PTS_33bit);
+ } else {
+ mLastRecoveredPTS = static_cast<int64_t>(
+ ((mLastRecoveredPTS - PTS_33bit + 0x100000000ll)
+ & 0xfffffffe00000000ull) | PTS_33bit);
+ // We start from 0, but recovered PTS could be slightly below 0.
+ // Clamp it to 0 as rest of the pipeline doesn't take negative pts.
+ // (eg. video is read first and starts at 0, but audio starts at 0xfffffff0)
+ if (mLastRecoveredPTS < 0ll) {
+ ALOGI("Clamping negative recovered PTS (%" PRId64 ") to 0", mLastRecoveredPTS);
+ mLastRecoveredPTS = 0ll;
+ }
+ }
+
+ return mLastRecoveredPTS;
+}
+
sp<MediaSource> ATSParser::Program::getSource(SourceType type) {
size_t index = (type == AUDIO) ? 0 : 0;
@@ -455,6 +527,8 @@
}
int64_t ATSParser::Program::convertPTSToTimestamp(uint64_t PTS) {
+ PTS = recoverPTS(PTS);
+
if (!(mParser->mFlags & TS_TIMESTAMPS_ARE_ABSOLUTE)) {
if (!mFirstPTSValid) {
mFirstPTSValid = true;
@@ -969,6 +1043,7 @@
mAbsoluteTimeAnchorUs(-1ll),
mTimeOffsetValid(false),
mTimeOffsetUs(0ll),
+ mLastRecoveredPTS(-1ll),
mNumTSPacketsParsed(0),
mNumPCRs(0) {
mPSISections.add(0 /* PID */, new PSISection);
@@ -987,11 +1062,21 @@
void ATSParser::signalDiscontinuity(
DiscontinuityType type, const sp<AMessage> &extra) {
int64_t mediaTimeUs;
- if ((type & DISCONTINUITY_TIME)
- && extra != NULL
- && extra->findInt64(
- IStreamListener::kKeyMediaTimeUs, &mediaTimeUs)) {
- mAbsoluteTimeAnchorUs = mediaTimeUs;
+ if ((type & DISCONTINUITY_TIME) && extra != NULL) {
+ if (extra->findInt64(IStreamListener::kKeyMediaTimeUs, &mediaTimeUs)) {
+ mAbsoluteTimeAnchorUs = mediaTimeUs;
+ }
+ if ((mFlags & TS_TIMESTAMPS_ARE_ABSOLUTE)
+ && extra->findInt64(
+ IStreamListener::kKeyRecentMediaTimeUs, &mediaTimeUs)) {
+ if (mAbsoluteTimeAnchorUs >= 0ll) {
+ mediaTimeUs -= mAbsoluteTimeAnchorUs;
+ }
+ if (mTimeOffsetValid) {
+ mediaTimeUs -= mTimeOffsetUs;
+ }
+ mLastRecoveredPTS = (mediaTimeUs * 9) / 100;
+ }
} else if (type == DISCONTINUITY_ABSOLUTE_TIME) {
int64_t timeUs;
CHECK(extra->findInt64("timeUs", &timeUs));
@@ -1075,7 +1160,7 @@
if (!found) {
mPrograms.push(
- new Program(this, program_number, programMapPID));
+ new Program(this, program_number, programMapPID, mLastRecoveredPTS));
}
if (mPSISections.indexOfKey(programMapPID) < 0) {
@@ -1098,10 +1183,12 @@
if (payload_unit_start_indicator) {
if (!section->isEmpty()) {
- return ERROR_UNSUPPORTED;
+ ALOGW("parsePID encounters payload_unit_start_indicator when section is not empty");
+ section->clear();
}
unsigned skip = br->getBits(8);
+ section->setSkipBytes(skip + 1); // skip filler bytes + pointer field itself
br->skipBits(skip * 8);
}
@@ -1116,6 +1203,9 @@
return OK;
}
+ if (!section->isCRCOkay()) {
+ return BAD_VALUE;
+ }
ABitReader sectionBits(section->data(), section->size());
if (PID == 0) {
@@ -1338,7 +1428,79 @@
////////////////////////////////////////////////////////////////////////////////
-ATSParser::PSISection::PSISection() {
+
+// CRC32 used for PSI section. The table was generated by following command:
+// $ python pycrc.py --model crc-32-mpeg --algorithm table-driven --generate c
+// Visit http://www.tty1.net/pycrc/index_en.html for more details.
+uint32_t ATSParser::PSISection::CRC_TABLE[] = {
+ 0x00000000, 0x04c11db7, 0x09823b6e, 0x0d4326d9,
+ 0x130476dc, 0x17c56b6b, 0x1a864db2, 0x1e475005,
+ 0x2608edb8, 0x22c9f00f, 0x2f8ad6d6, 0x2b4bcb61,
+ 0x350c9b64, 0x31cd86d3, 0x3c8ea00a, 0x384fbdbd,
+ 0x4c11db70, 0x48d0c6c7, 0x4593e01e, 0x4152fda9,
+ 0x5f15adac, 0x5bd4b01b, 0x569796c2, 0x52568b75,
+ 0x6a1936c8, 0x6ed82b7f, 0x639b0da6, 0x675a1011,
+ 0x791d4014, 0x7ddc5da3, 0x709f7b7a, 0x745e66cd,
+ 0x9823b6e0, 0x9ce2ab57, 0x91a18d8e, 0x95609039,
+ 0x8b27c03c, 0x8fe6dd8b, 0x82a5fb52, 0x8664e6e5,
+ 0xbe2b5b58, 0xbaea46ef, 0xb7a96036, 0xb3687d81,
+ 0xad2f2d84, 0xa9ee3033, 0xa4ad16ea, 0xa06c0b5d,
+ 0xd4326d90, 0xd0f37027, 0xddb056fe, 0xd9714b49,
+ 0xc7361b4c, 0xc3f706fb, 0xceb42022, 0xca753d95,
+ 0xf23a8028, 0xf6fb9d9f, 0xfbb8bb46, 0xff79a6f1,
+ 0xe13ef6f4, 0xe5ffeb43, 0xe8bccd9a, 0xec7dd02d,
+ 0x34867077, 0x30476dc0, 0x3d044b19, 0x39c556ae,
+ 0x278206ab, 0x23431b1c, 0x2e003dc5, 0x2ac12072,
+ 0x128e9dcf, 0x164f8078, 0x1b0ca6a1, 0x1fcdbb16,
+ 0x018aeb13, 0x054bf6a4, 0x0808d07d, 0x0cc9cdca,
+ 0x7897ab07, 0x7c56b6b0, 0x71159069, 0x75d48dde,
+ 0x6b93dddb, 0x6f52c06c, 0x6211e6b5, 0x66d0fb02,
+ 0x5e9f46bf, 0x5a5e5b08, 0x571d7dd1, 0x53dc6066,
+ 0x4d9b3063, 0x495a2dd4, 0x44190b0d, 0x40d816ba,
+ 0xaca5c697, 0xa864db20, 0xa527fdf9, 0xa1e6e04e,
+ 0xbfa1b04b, 0xbb60adfc, 0xb6238b25, 0xb2e29692,
+ 0x8aad2b2f, 0x8e6c3698, 0x832f1041, 0x87ee0df6,
+ 0x99a95df3, 0x9d684044, 0x902b669d, 0x94ea7b2a,
+ 0xe0b41de7, 0xe4750050, 0xe9362689, 0xedf73b3e,
+ 0xf3b06b3b, 0xf771768c, 0xfa325055, 0xfef34de2,
+ 0xc6bcf05f, 0xc27dede8, 0xcf3ecb31, 0xcbffd686,
+ 0xd5b88683, 0xd1799b34, 0xdc3abded, 0xd8fba05a,
+ 0x690ce0ee, 0x6dcdfd59, 0x608edb80, 0x644fc637,
+ 0x7a089632, 0x7ec98b85, 0x738aad5c, 0x774bb0eb,
+ 0x4f040d56, 0x4bc510e1, 0x46863638, 0x42472b8f,
+ 0x5c007b8a, 0x58c1663d, 0x558240e4, 0x51435d53,
+ 0x251d3b9e, 0x21dc2629, 0x2c9f00f0, 0x285e1d47,
+ 0x36194d42, 0x32d850f5, 0x3f9b762c, 0x3b5a6b9b,
+ 0x0315d626, 0x07d4cb91, 0x0a97ed48, 0x0e56f0ff,
+ 0x1011a0fa, 0x14d0bd4d, 0x19939b94, 0x1d528623,
+ 0xf12f560e, 0xf5ee4bb9, 0xf8ad6d60, 0xfc6c70d7,
+ 0xe22b20d2, 0xe6ea3d65, 0xeba91bbc, 0xef68060b,
+ 0xd727bbb6, 0xd3e6a601, 0xdea580d8, 0xda649d6f,
+ 0xc423cd6a, 0xc0e2d0dd, 0xcda1f604, 0xc960ebb3,
+ 0xbd3e8d7e, 0xb9ff90c9, 0xb4bcb610, 0xb07daba7,
+ 0xae3afba2, 0xaafbe615, 0xa7b8c0cc, 0xa379dd7b,
+ 0x9b3660c6, 0x9ff77d71, 0x92b45ba8, 0x9675461f,
+ 0x8832161a, 0x8cf30bad, 0x81b02d74, 0x857130c3,
+ 0x5d8a9099, 0x594b8d2e, 0x5408abf7, 0x50c9b640,
+ 0x4e8ee645, 0x4a4ffbf2, 0x470cdd2b, 0x43cdc09c,
+ 0x7b827d21, 0x7f436096, 0x7200464f, 0x76c15bf8,
+ 0x68860bfd, 0x6c47164a, 0x61043093, 0x65c52d24,
+ 0x119b4be9, 0x155a565e, 0x18197087, 0x1cd86d30,
+ 0x029f3d35, 0x065e2082, 0x0b1d065b, 0x0fdc1bec,
+ 0x3793a651, 0x3352bbe6, 0x3e119d3f, 0x3ad08088,
+ 0x2497d08d, 0x2056cd3a, 0x2d15ebe3, 0x29d4f654,
+ 0xc5a92679, 0xc1683bce, 0xcc2b1d17, 0xc8ea00a0,
+ 0xd6ad50a5, 0xd26c4d12, 0xdf2f6bcb, 0xdbee767c,
+ 0xe3a1cbc1, 0xe760d676, 0xea23f0af, 0xeee2ed18,
+ 0xf0a5bd1d, 0xf464a0aa, 0xf9278673, 0xfde69bc4,
+ 0x89b8fd09, 0x8d79e0be, 0x803ac667, 0x84fbdbd0,
+ 0x9abc8bd5, 0x9e7d9662, 0x933eb0bb, 0x97ffad0c,
+ 0xafb010b1, 0xab710d06, 0xa6322bdf, 0xa2f33668,
+ 0xbcb4666d, 0xb8757bda, 0xb5365d03, 0xb1f740b4
+ };
+
+ATSParser::PSISection::PSISection() :
+ mSkipBytes(0) {
}
ATSParser::PSISection::~PSISection() {
@@ -1369,10 +1531,15 @@
return OK;
}
+void ATSParser::PSISection::setSkipBytes(uint8_t skip) {
+ mSkipBytes = skip;
+}
+
void ATSParser::PSISection::clear() {
if (mBuffer != NULL) {
mBuffer->setRange(0, 0);
}
+ mSkipBytes = 0;
}
bool ATSParser::PSISection::isComplete() const {
@@ -1396,4 +1563,30 @@
return mBuffer == NULL ? 0 : mBuffer->size();
}
+bool ATSParser::PSISection::isCRCOkay() const {
+ if (!isComplete()) {
+ return false;
+ }
+ uint8_t* data = mBuffer->data();
+
+ // Return true if section_syntax_indicator says no section follows the field section_length.
+ if ((data[1] & 0x80) == 0) {
+ return true;
+ }
+
+ unsigned sectionLength = U16_AT(data + 1) & 0xfff;
+ ALOGV("sectionLength %u, skip %u", sectionLength, mSkipBytes);
+
+ // Skip the preceding field present when payload start indicator is on.
+ sectionLength -= mSkipBytes;
+
+ uint32_t crc = 0xffffffff;
+ for(unsigned i = 0; i < sectionLength + 4 /* crc */; i++) {
+ uint8_t b = data[i];
+ int index = ((crc >> 24) ^ (b & 0xff)) & 0xff;
+ crc = CRC_TABLE[index] ^ (crc << 8);
+ }
+ ALOGV("crc: %08x\n", crc);
+ return (crc == 0);
+}
} // namespace android
diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h
index 75d76dc..a1405bd 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.h
+++ b/media/libstagefright/mpeg2ts/ATSParser.h
@@ -46,6 +46,9 @@
DISCONTINUITY_AUDIO_FORMAT
| DISCONTINUITY_VIDEO_FORMAT
| DISCONTINUITY_TIME,
+ DISCONTINUITY_FORMAT_ONLY =
+ DISCONTINUITY_AUDIO_FORMAT
+ | DISCONTINUITY_VIDEO_FORMAT,
};
enum Flags {
@@ -115,6 +118,7 @@
bool mTimeOffsetValid;
int64_t mTimeOffsetUs;
+ int64_t mLastRecoveredPTS;
size_t mNumTSPacketsParsed;
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index f266fe7..c5bb41b 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -19,6 +19,8 @@
#include "AnotherPacketSource.h"
+#include "include/avc_utils.h"
+
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
@@ -27,6 +29,7 @@
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
#include <utils/Vector.h>
#include <inttypes.h>
@@ -38,6 +41,7 @@
AnotherPacketSource::AnotherPacketSource(const sp<MetaData> &meta)
: mIsAudio(false),
mIsVideo(false),
+ mEnabled(true),
mFormat(NULL),
mLastQueuedTimeUs(0),
mEOSResult(OK),
@@ -48,7 +52,10 @@
}
void AnotherPacketSource::setFormat(const sp<MetaData> &meta) {
- CHECK(mFormat == NULL);
+ if (mFormat != NULL) {
+ // Only allowed to be set once. Requires explicit clear to reset.
+ return;
+ }
mIsAudio = false;
mIsVideo = false;
@@ -91,13 +98,12 @@
while (it != mBuffers.end()) {
sp<ABuffer> buffer = *it;
int32_t discontinuity;
- if (buffer->meta()->findInt32("discontinuity", &discontinuity)) {
- break;
- }
-
- sp<RefBase> object;
- if (buffer->meta()->findObject("format", &object)) {
- return mFormat = static_cast<MetaData*>(object.get());
+ if (!buffer->meta()->findInt32("discontinuity", &discontinuity)) {
+ sp<RefBase> object;
+ if (buffer->meta()->findObject("format", &object)) {
+ setFormat(static_cast<MetaData*>(object.get()));
+ return mFormat;
+ }
}
++it;
@@ -131,7 +137,7 @@
sp<RefBase> object;
if ((*buffer)->meta()->findObject("format", &object)) {
- mFormat = static_cast<MetaData*>(object.get());
+ setFormat(static_cast<MetaData*>(object.get()));
}
return OK;
@@ -153,7 +159,6 @@
const sp<ABuffer> buffer = *mBuffers.begin();
mBuffers.erase(mBuffers.begin());
- mLatestDequeuedMeta = buffer->meta()->dup();
int32_t discontinuity;
if (buffer->meta()->findInt32("discontinuity", &discontinuity)) {
@@ -164,9 +169,11 @@
return INFO_DISCONTINUITY;
}
+ mLatestDequeuedMeta = buffer->meta()->dup();
+
sp<RefBase> object;
if (buffer->meta()->findObject("format", &object)) {
- mFormat = static_cast<MetaData*>(object.get());
+ setFormat(static_cast<MetaData*>(object.get()));
}
int64_t timeUs;
@@ -203,20 +210,26 @@
return;
}
- int64_t lastQueuedTimeUs;
- CHECK(buffer->meta()->findInt64("timeUs", &lastQueuedTimeUs));
- mLastQueuedTimeUs = lastQueuedTimeUs;
- ALOGV("queueAccessUnit timeUs=%" PRIi64 " us (%.2f secs)", mLastQueuedTimeUs, mLastQueuedTimeUs / 1E6);
-
Mutex::Autolock autoLock(mLock);
mBuffers.push_back(buffer);
mCondition.signal();
int32_t discontinuity;
if (buffer->meta()->findInt32("discontinuity", &discontinuity)) {
+ // discontinuity handling needs to be consistent with queueDiscontinuity()
++mQueuedDiscontinuityCount;
+ mLastQueuedTimeUs = 0ll;
+ mEOSResult = OK;
+ mLatestEnqueuedMeta = NULL;
+ return;
}
+ int64_t lastQueuedTimeUs;
+ CHECK(buffer->meta()->findInt64("timeUs", &lastQueuedTimeUs));
+ mLastQueuedTimeUs = lastQueuedTimeUs;
+ ALOGV("queueAccessUnit timeUs=%" PRIi64 " us (%.2f secs)",
+ mLastQueuedTimeUs, mLastQueuedTimeUs / 1E6);
+
if (mLatestEnqueuedMeta == NULL) {
mLatestEnqueuedMeta = buffer->meta()->dup();
} else {
@@ -296,6 +309,10 @@
bool AnotherPacketSource::hasBufferAvailable(status_t *finalResult) {
Mutex::Autolock autoLock(mLock);
+ *finalResult = OK;
+ if (!mEnabled) {
+ return false;
+ }
if (!mBuffers.empty()) {
return true;
}
@@ -304,6 +321,24 @@
return false;
}
+bool AnotherPacketSource::hasDataBufferAvailable(status_t *finalResult) {
+ Mutex::Autolock autoLock(mLock);
+ *finalResult = OK;
+ if (!mEnabled) {
+ return false;
+ }
+ List<sp<ABuffer> >::iterator it;
+ for (it = mBuffers.begin(); it != mBuffers.end(); it++) {
+ int32_t discontinuity;
+ if (!(*it)->meta()->findInt32("discontinuity", &discontinuity)) {
+ return true;
+ }
+ }
+
+ *finalResult = mEOSResult;
+ return false;
+}
+
int64_t AnotherPacketSource::getBufferedDurationUs(status_t *finalResult) {
Mutex::Autolock autoLock(mLock);
return getBufferedDurationUs_l(finalResult);
@@ -422,4 +457,152 @@
return mLatestDequeuedMeta;
}
+void AnotherPacketSource::enable(bool enable) {
+ Mutex::Autolock autoLock(mLock);
+ mEnabled = enable;
+}
+
+/*
+ * returns the sample meta that's delayUs after queue head
+ * (NULL if such sample is unavailable)
+ */
+sp<AMessage> AnotherPacketSource::getMetaAfterLastDequeued(int64_t delayUs) {
+ Mutex::Autolock autoLock(mLock);
+ int64_t firstUs = -1;
+ int64_t lastUs = -1;
+ int64_t durationUs = 0;
+
+ List<sp<ABuffer> >::iterator it;
+ for (it = mBuffers.begin(); it != mBuffers.end(); ++it) {
+ const sp<ABuffer> &buffer = *it;
+ int32_t discontinuity;
+ if (buffer->meta()->findInt32("discontinuity", &discontinuity)) {
+ durationUs += lastUs - firstUs;
+ firstUs = -1;
+ lastUs = -1;
+ continue;
+ }
+ int64_t timeUs;
+ if (buffer->meta()->findInt64("timeUs", &timeUs)) {
+ if (firstUs < 0) {
+ firstUs = timeUs;
+ }
+ if (lastUs < 0 || timeUs > lastUs) {
+ lastUs = timeUs;
+ }
+ if (durationUs + (lastUs - firstUs) >= delayUs) {
+ return buffer->meta();
+ }
+ }
+ }
+ return NULL;
+}
+
+/*
+ * removes samples with time equal or after meta
+ */
+void AnotherPacketSource::trimBuffersAfterMeta(
+ const sp<AMessage> &meta) {
+ if (meta == NULL) {
+ ALOGW("trimming with NULL meta, ignoring");
+ return;
+ }
+
+ Mutex::Autolock autoLock(mLock);
+ if (mBuffers.empty()) {
+ return;
+ }
+
+ HLSTime stopTime(meta);
+ ALOGV("trimBuffersAfterMeta: discontinuitySeq %zu, timeUs %lld",
+ stopTime.mSeq, (long long)stopTime.mTimeUs);
+
+ List<sp<ABuffer> >::iterator it;
+ sp<AMessage> newLatestEnqueuedMeta = NULL;
+ int64_t newLastQueuedTimeUs = 0;
+ size_t newDiscontinuityCount = 0;
+ for (it = mBuffers.begin(); it != mBuffers.end(); ++it) {
+ const sp<ABuffer> &buffer = *it;
+ int32_t discontinuity;
+ if (buffer->meta()->findInt32("discontinuity", &discontinuity)) {
+ newDiscontinuityCount++;
+ continue;
+ }
+
+ HLSTime curTime(buffer->meta());
+ if (!(curTime < stopTime)) {
+ ALOGV("trimming from %lld (inclusive) to end",
+ (long long)curTime.mTimeUs);
+ break;
+ }
+ newLatestEnqueuedMeta = buffer->meta();
+ newLastQueuedTimeUs = curTime.mTimeUs;
+ }
+ mBuffers.erase(it, mBuffers.end());
+ mLatestEnqueuedMeta = newLatestEnqueuedMeta;
+ mLastQueuedTimeUs = newLastQueuedTimeUs;
+ mQueuedDiscontinuityCount = newDiscontinuityCount;
+}
+
+/*
+ * removes samples with time equal or before meta;
+ * returns first sample left in the queue.
+ *
+ * (for AVC, if trim happens, the samples left will always start
+ * at next IDR.)
+ */
+sp<AMessage> AnotherPacketSource::trimBuffersBeforeMeta(
+ const sp<AMessage> &meta) {
+ HLSTime startTime(meta);
+ ALOGV("trimBuffersBeforeMeta: discontinuitySeq %zu, timeUs %lld",
+ startTime.mSeq, (long long)startTime.mTimeUs);
+
+ sp<AMessage> firstMeta;
+ Mutex::Autolock autoLock(mLock);
+ if (mBuffers.empty()) {
+ return NULL;
+ }
+
+ sp<MetaData> format;
+ bool isAvc = false;
+
+ List<sp<ABuffer> >::iterator it;
+ size_t discontinuityCount = 0;
+ for (it = mBuffers.begin(); it != mBuffers.end(); ++it) {
+ const sp<ABuffer> &buffer = *it;
+ int32_t discontinuity;
+ if (buffer->meta()->findInt32("discontinuity", &discontinuity)) {
+ format = NULL;
+ isAvc = false;
+ discontinuityCount++;
+ continue;
+ }
+ if (format == NULL) {
+ sp<RefBase> object;
+ if (buffer->meta()->findObject("format", &object)) {
+ const char* mime;
+ format = static_cast<MetaData*>(object.get());
+ isAvc = format != NULL
+ && format->findCString(kKeyMIMEType, &mime)
+ && !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC);
+ }
+ }
+ if (isAvc && !IsIDR(buffer)) {
+ continue;
+ }
+
+ HLSTime curTime(buffer->meta());
+ if (startTime < curTime) {
+ ALOGV("trimming from beginning to %lld (not inclusive)",
+ (long long)curTime.mTimeUs);
+ firstMeta = buffer->meta();
+ break;
+ }
+ }
+ mBuffers.erase(mBuffers.begin(), it);
+ mQueuedDiscontinuityCount -= discontinuityCount;
+ mLatestDequeuedMeta = NULL;
+ return firstMeta;
+}
+
} // namespace android
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.h b/media/libstagefright/mpeg2ts/AnotherPacketSource.h
index 809a858..fa7dd6a 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.h
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.h
@@ -43,8 +43,12 @@
void clear();
+ // Returns true if we have any packets including discontinuities
bool hasBufferAvailable(status_t *finalResult);
+ // Returns true if we have packets that's not discontinuities
+ bool hasDataBufferAvailable(status_t *finalResult);
+
// Returns the difference between the last and the first queued
// presentation timestamps since the last discontinuity (if any).
int64_t getBufferedDurationUs(status_t *finalResult);
@@ -66,8 +70,14 @@
bool isFinished(int64_t duration) const;
+ void enable(bool enable);
+
sp<AMessage> getLatestEnqueuedMeta();
sp<AMessage> getLatestDequeuedMeta();
+ sp<AMessage> getMetaAfterLastDequeued(int64_t delayUs);
+
+ void trimBuffersAfterMeta(const sp<AMessage> &meta);
+ sp<AMessage> trimBuffersBeforeMeta(const sp<AMessage> &meta);
protected:
virtual ~AnotherPacketSource();
@@ -78,6 +88,7 @@
bool mIsAudio;
bool mIsVideo;
+ bool mEnabled;
sp<MetaData> mFormat;
int64_t mLastQueuedTimeUs;
List<sp<ABuffer> > mBuffers;
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index a605595..b17985c 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -257,8 +257,8 @@
uint8_t *ptr = (uint8_t *)data;
ssize_t startOffset = -1;
- for (size_t i = 0; i + 3 < size; ++i) {
- if (!memcmp("\x00\x00\x00\x01", &ptr[i], 4)) {
+ for (size_t i = 0; i + 2 < size; ++i) {
+ if (!memcmp("\x00\x00\x01", &ptr[i], 3)) {
startOffset = i;
break;
}
@@ -617,8 +617,6 @@
// having to interpolate.
// The final AAC frame may well extend into the next RangeInfo but
// that's ok.
- // TODO: the logic commented above is skipped because codec cannot take
- // arbitrary sized input buffers;
size_t offset = 0;
while (offset < info.mLength) {
if (offset + 7 > mBuffer->size()) {
@@ -683,12 +681,9 @@
size_t headerSize __unused = protection_absent ? 7 : 9;
offset += aac_frame_length;
- // TODO: move back to concatenation when codec can support arbitrary input buffers.
- // For now only queue a single buffer
- break;
}
- int64_t timeUs = fetchTimestampAAC(offset);
+ int64_t timeUs = fetchTimestamp(offset);
sp<ABuffer> accessUnit = new ABuffer(offset);
memcpy(accessUnit->data(), mBuffer->data(), offset);
@@ -735,50 +730,6 @@
return timeUs;
}
-// TODO: avoid interpolating timestamps once codec supports arbitrary sized input buffers
-int64_t ElementaryStreamQueue::fetchTimestampAAC(size_t size) {
- int64_t timeUs = -1;
- bool first = true;
-
- size_t samplesize = size;
- while (size > 0) {
- CHECK(!mRangeInfos.empty());
-
- RangeInfo *info = &*mRangeInfos.begin();
-
- if (first) {
- timeUs = info->mTimestampUs;
- first = false;
- }
-
- if (info->mLength > size) {
- int32_t sampleRate;
- CHECK(mFormat->findInt32(kKeySampleRate, &sampleRate));
- info->mLength -= size;
- size_t numSamples = 1024 * size / samplesize;
- info->mTimestampUs += numSamples * 1000000ll / sampleRate;
- size = 0;
- } else {
- size -= info->mLength;
-
- mRangeInfos.erase(mRangeInfos.begin());
- info = NULL;
- }
-
- }
-
- if (timeUs == 0ll) {
- ALOGV("Returning 0 timestamp");
- }
-
- return timeUs;
-}
-
-struct NALPosition {
- size_t nalOffset;
- size_t nalSize;
-};
-
sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitH264() {
const uint8_t *data = mBuffer->data();
@@ -786,6 +737,7 @@
Vector<NALPosition> nals;
size_t totalSize = 0;
+ size_t seiCount = 0;
status_t err;
const uint8_t *nalStart;
@@ -815,6 +767,9 @@
// next frame.
flush = true;
+ } else if (nalType == 6 && nalSize > 0) {
+ // found non-zero sized SEI
+ ++seiCount;
}
if (flush) {
@@ -823,21 +778,29 @@
size_t auSize = 4 * nals.size() + totalSize;
sp<ABuffer> accessUnit = new ABuffer(auSize);
+ sp<ABuffer> sei;
+
+ if (seiCount > 0) {
+ sei = new ABuffer(seiCount * sizeof(NALPosition));
+ accessUnit->meta()->setBuffer("sei", sei);
+ }
#if !LOG_NDEBUG
AString out;
#endif
size_t dstOffset = 0;
+ size_t seiIndex = 0;
for (size_t i = 0; i < nals.size(); ++i) {
const NALPosition &pos = nals.itemAt(i);
unsigned nalType = mBuffer->data()[pos.nalOffset] & 0x1f;
- if (nalType == 6) {
- sp<ABuffer> sei = new ABuffer(pos.nalSize);
- memcpy(sei->data(), mBuffer->data() + pos.nalOffset, pos.nalSize);
- accessUnit->meta()->setBuffer("sei", sei);
+ if (nalType == 6 && pos.nalSize > 0) {
+ CHECK_LT(seiIndex, sei->size() / sizeof(NALPosition));
+ NALPosition &seiPos = ((NALPosition *)sei->data())[seiIndex++];
+ seiPos.nalOffset = dstOffset + 4;
+ seiPos.nalSize = pos.nalSize;
}
#if !LOG_NDEBUG
diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h
index eb4b1c9..45b4624 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.h
+++ b/media/libstagefright/mpeg2ts/ESQueue.h
@@ -77,7 +77,6 @@
// consume a logical (compressed) access unit of size "size",
// returns its timestamp in us (or -1 if no time information).
int64_t fetchTimestamp(size_t size);
- int64_t fetchTimestampAAC(size_t size);
DISALLOW_EVIL_CONSTRUCTORS(ElementaryStreamQueue);
};
diff --git a/media/libstagefright/omx/Android.mk b/media/libstagefright/omx/Android.mk
index aaa8334..07ea605 100644
--- a/media/libstagefright/omx/Android.mk
+++ b/media/libstagefright/omx/Android.mk
@@ -1,11 +1,8 @@
LOCAL_PATH:= $(call my-dir)
include $(CLEAR_VARS)
-ifeq ($(TARGET_DEVICE), manta)
- LOCAL_CFLAGS += -DSURFACE_IS_BGR32
-endif
-
LOCAL_SRC_FILES:= \
+ FrameDropper.cpp \
GraphicBufferSource.cpp \
OMX.cpp \
OMXMaster.cpp \
diff --git a/media/libstagefright/omx/FrameDropper.cpp b/media/libstagefright/omx/FrameDropper.cpp
new file mode 100644
index 0000000..9fba0b7
--- /dev/null
+++ b/media/libstagefright/omx/FrameDropper.cpp
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2015 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 "FrameDropper"
+#include <utils/Log.h>
+
+#include "FrameDropper.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+
+namespace android {
+
+static const int64_t kMaxJitterUs = 2000;
+
+FrameDropper::FrameDropper()
+ : mDesiredMinTimeUs(-1),
+ mMinIntervalUs(0) {
+}
+
+FrameDropper::~FrameDropper() {
+}
+
+status_t FrameDropper::setMaxFrameRate(float maxFrameRate) {
+ if (maxFrameRate <= 0) {
+ ALOGE("framerate should be positive but got %f.", maxFrameRate);
+ return BAD_VALUE;
+ }
+ mMinIntervalUs = (int64_t) (1000000.0f / maxFrameRate);
+ return OK;
+}
+
+bool FrameDropper::shouldDrop(int64_t timeUs) {
+ if (mMinIntervalUs <= 0) {
+ return false;
+ }
+
+ if (mDesiredMinTimeUs < 0) {
+ mDesiredMinTimeUs = timeUs + mMinIntervalUs;
+ ALOGV("first frame %lld, next desired frame %lld", timeUs, mDesiredMinTimeUs);
+ return false;
+ }
+
+ if (timeUs < (mDesiredMinTimeUs - kMaxJitterUs)) {
+ ALOGV("drop frame %lld, desired frame %lld, diff %lld",
+ timeUs, mDesiredMinTimeUs, mDesiredMinTimeUs - timeUs);
+ return true;
+ }
+
+ int64_t n = (timeUs - mDesiredMinTimeUs + kMaxJitterUs) / mMinIntervalUs;
+ mDesiredMinTimeUs += (n + 1) * mMinIntervalUs;
+ ALOGV("keep frame %lld, next desired frame %lld, diff %lld",
+ timeUs, mDesiredMinTimeUs, mDesiredMinTimeUs - timeUs);
+ return false;
+}
+
+} // namespace android
diff --git a/media/libstagefright/omx/FrameDropper.h b/media/libstagefright/omx/FrameDropper.h
new file mode 100644
index 0000000..c5a6d4b
--- /dev/null
+++ b/media/libstagefright/omx/FrameDropper.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (C) 2015 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 FRAME_DROPPER_H_
+
+#define FRAME_DROPPER_H_
+
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+struct FrameDropper : public RefBase {
+ // No frames will be dropped until a valid max frame rate is set.
+ FrameDropper();
+
+ // maxFrameRate required to be positive.
+ status_t setMaxFrameRate(float maxFrameRate);
+
+ // Returns false if max frame rate has not been set via setMaxFrameRate.
+ bool shouldDrop(int64_t timeUs);
+
+protected:
+ virtual ~FrameDropper();
+
+private:
+ int64_t mDesiredMinTimeUs;
+ int64_t mMinIntervalUs;
+
+ DISALLOW_EVIL_CONSTRUCTORS(FrameDropper);
+};
+
+} // namespace android
+
+#endif // FRAME_DROPPER_H_
diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index 2945644..477cfc6 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -31,6 +31,7 @@
#include <gui/BufferItem.h>
#include <inttypes.h>
+#include "FrameDropper.h"
namespace android {
@@ -54,9 +55,9 @@
mRepeatAfterUs(-1ll),
mRepeatLastFrameGeneration(0),
mRepeatLastFrameTimestamp(-1ll),
- mLatestSubmittedBufferId(-1),
- mLatestSubmittedBufferFrameNum(0),
- mLatestSubmittedBufferUseCount(0),
+ mLatestBufferId(-1),
+ mLatestBufferFrameNum(0),
+ mLatestBufferUseCount(0),
mRepeatBufferDeferred(false),
mTimePerCaptureUs(-1ll),
mTimePerFrameUs(-1ll),
@@ -153,9 +154,9 @@
mLooper->registerHandler(mReflector);
mLooper->start();
- if (mLatestSubmittedBufferId >= 0) {
+ if (mLatestBufferId >= 0) {
sp<AMessage> msg =
- new AMessage(kWhatRepeatLastFrame, mReflector->id());
+ new AMessage(kWhatRepeatLastFrame, mReflector);
msg->setInt32("generation", ++mRepeatLastFrameGeneration);
msg->post(mRepeatAfterUs);
@@ -288,8 +289,8 @@
ALOGV("cbi %d matches bq slot %d, handle=%p",
cbi, id, mBufferSlot[id]->handle);
- if (id == mLatestSubmittedBufferId) {
- CHECK_GT(mLatestSubmittedBufferUseCount--, 0);
+ if (id == mLatestBufferId) {
+ CHECK_GT(mLatestBufferUseCount--, 0);
} else {
mConsumer->releaseBuffer(id, codecBuffer.mFrameNumber,
EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE);
@@ -314,11 +315,11 @@
ALOGV("buffer freed, EOS pending");
submitEndOfInputStream_l();
} else if (mRepeatBufferDeferred) {
- bool success = repeatLatestSubmittedBuffer_l();
+ bool success = repeatLatestBuffer_l();
if (success) {
- ALOGV("deferred repeatLatestSubmittedBuffer_l SUCCESS");
+ ALOGV("deferred repeatLatestBuffer_l SUCCESS");
} else {
- ALOGV("deferred repeatLatestSubmittedBuffer_l FAILURE");
+ ALOGV("deferred repeatLatestBuffer_l FAILURE");
}
mRepeatBufferDeferred = false;
}
@@ -383,12 +384,12 @@
mSuspended = false;
if (mExecuting && mNumFramesAvailable == 0 && mRepeatBufferDeferred) {
- if (repeatLatestSubmittedBuffer_l()) {
- ALOGV("suspend/deferred repeatLatestSubmittedBuffer_l SUCCESS");
+ if (repeatLatestBuffer_l()) {
+ ALOGV("suspend/deferred repeatLatestBuffer_l SUCCESS");
mRepeatBufferDeferred = false;
} else {
- ALOGV("suspend/deferred repeatLatestSubmittedBuffer_l FAILURE");
+ ALOGV("suspend/deferred repeatLatestBuffer_l FAILURE");
}
}
}
@@ -442,12 +443,22 @@
// only submit sample if start time is unspecified, or sample
// is queued after the specified start time
+ bool dropped = false;
if (mSkipFramesBeforeNs < 0ll || item.mTimestamp >= mSkipFramesBeforeNs) {
// if start time is set, offset time stamp by start time
if (mSkipFramesBeforeNs > 0) {
item.mTimestamp -= mSkipFramesBeforeNs;
}
- err = submitBuffer_l(item, cbi);
+
+ int64_t timeUs = item.mTimestamp / 1000;
+ if (mFrameDropper != NULL && mFrameDropper->shouldDrop(timeUs)) {
+ ALOGV("skipping frame (%lld) to meet max framerate", static_cast<long long>(timeUs));
+ // set err to OK so that the skipped frame can still be saved as the lastest frame
+ err = OK;
+ dropped = true;
+ } else {
+ err = submitBuffer_l(item, cbi);
+ }
}
if (err != OK) {
@@ -456,46 +467,46 @@
EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE);
} else {
ALOGV("buffer submitted (bq %d, cbi %d)", item.mBuf, cbi);
- setLatestSubmittedBuffer_l(item);
+ setLatestBuffer_l(item, dropped);
}
return true;
}
-bool GraphicBufferSource::repeatLatestSubmittedBuffer_l() {
+bool GraphicBufferSource::repeatLatestBuffer_l() {
CHECK(mExecuting && mNumFramesAvailable == 0);
- if (mLatestSubmittedBufferId < 0 || mSuspended) {
+ if (mLatestBufferId < 0 || mSuspended) {
return false;
}
- if (mBufferSlot[mLatestSubmittedBufferId] == NULL) {
+ if (mBufferSlot[mLatestBufferId] == NULL) {
// This can happen if the remote side disconnects, causing
// onBuffersReleased() to NULL out our copy of the slots. The
// buffer is gone, so we have nothing to show.
//
// To be on the safe side we try to release the buffer.
- ALOGD("repeatLatestSubmittedBuffer_l: slot was NULL");
+ ALOGD("repeatLatestBuffer_l: slot was NULL");
mConsumer->releaseBuffer(
- mLatestSubmittedBufferId,
- mLatestSubmittedBufferFrameNum,
+ mLatestBufferId,
+ mLatestBufferFrameNum,
EGL_NO_DISPLAY,
EGL_NO_SYNC_KHR,
Fence::NO_FENCE);
- mLatestSubmittedBufferId = -1;
- mLatestSubmittedBufferFrameNum = 0;
+ mLatestBufferId = -1;
+ mLatestBufferFrameNum = 0;
return false;
}
int cbi = findAvailableCodecBuffer_l();
if (cbi < 0) {
// No buffers available, bail.
- ALOGV("repeatLatestSubmittedBuffer_l: no codec buffers.");
+ ALOGV("repeatLatestBuffer_l: no codec buffers.");
return false;
}
BufferItem item;
- item.mBuf = mLatestSubmittedBufferId;
- item.mFrameNumber = mLatestSubmittedBufferFrameNum;
+ item.mBuf = mLatestBufferId;
+ item.mFrameNumber = mLatestBufferFrameNum;
item.mTimestamp = mRepeatLastFrameTimestamp;
status_t err = submitBuffer_l(item, cbi);
@@ -504,7 +515,7 @@
return false;
}
- ++mLatestSubmittedBufferUseCount;
+ ++mLatestBufferUseCount;
/* repeat last frame up to kRepeatLastFrameCount times.
* in case of static scene, a single repeat might not get rid of encoder
@@ -514,7 +525,7 @@
mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
if (mReflector != NULL) {
- sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
+ sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector);
msg->setInt32("generation", ++mRepeatLastFrameGeneration);
msg->post(mRepeatAfterUs);
}
@@ -523,31 +534,31 @@
return true;
}
-void GraphicBufferSource::setLatestSubmittedBuffer_l(
- const BufferItem &item) {
- ALOGV("setLatestSubmittedBuffer_l");
+void GraphicBufferSource::setLatestBuffer_l(
+ const BufferItem &item, bool dropped) {
+ ALOGV("setLatestBuffer_l");
- if (mLatestSubmittedBufferId >= 0) {
- if (mLatestSubmittedBufferUseCount == 0) {
+ if (mLatestBufferId >= 0) {
+ if (mLatestBufferUseCount == 0) {
mConsumer->releaseBuffer(
- mLatestSubmittedBufferId,
- mLatestSubmittedBufferFrameNum,
+ mLatestBufferId,
+ mLatestBufferFrameNum,
EGL_NO_DISPLAY,
EGL_NO_SYNC_KHR,
Fence::NO_FENCE);
}
}
- mLatestSubmittedBufferId = item.mBuf;
- mLatestSubmittedBufferFrameNum = item.mFrameNumber;
+ mLatestBufferId = item.mBuf;
+ mLatestBufferFrameNum = item.mFrameNumber;
mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
- mLatestSubmittedBufferUseCount = 1;
+ mLatestBufferUseCount = dropped ? 0 : 1;
mRepeatBufferDeferred = false;
mRepeatLastFrameCount = kRepeatLastFrameCount;
if (mReflector != NULL) {
- sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
+ sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector);
msg->setInt32("generation", ++mRepeatLastFrameGeneration);
msg->post(mRepeatAfterUs);
}
@@ -842,6 +853,23 @@
return OK;
}
+status_t GraphicBufferSource::setMaxFps(float maxFps) {
+ Mutex::Autolock autoLock(mMutex);
+
+ if (mExecuting) {
+ return INVALID_OPERATION;
+ }
+
+ mFrameDropper = new FrameDropper();
+ status_t err = mFrameDropper->setMaxFrameRate(maxFps);
+ if (err != OK) {
+ mFrameDropper.clear();
+ return err;
+ }
+
+ return OK;
+}
+
void GraphicBufferSource::setSkipFramesBeforeUs(int64_t skipFramesBeforeUs) {
Mutex::Autolock autoLock(mMutex);
@@ -880,12 +908,12 @@
break;
}
- bool success = repeatLatestSubmittedBuffer_l();
+ bool success = repeatLatestBuffer_l();
if (success) {
- ALOGV("repeatLatestSubmittedBuffer_l SUCCESS");
+ ALOGV("repeatLatestBuffer_l SUCCESS");
} else {
- ALOGV("repeatLatestSubmittedBuffer_l FAILURE");
+ ALOGV("repeatLatestBuffer_l FAILURE");
mRepeatBufferDeferred = true;
}
break;
diff --git a/media/libstagefright/omx/GraphicBufferSource.h b/media/libstagefright/omx/GraphicBufferSource.h
index 401bbc3..1067472 100644
--- a/media/libstagefright/omx/GraphicBufferSource.h
+++ b/media/libstagefright/omx/GraphicBufferSource.h
@@ -30,6 +30,8 @@
namespace android {
+class FrameDropper;
+
/*
* This class is used to feed OMX codecs from a Surface via BufferQueue.
*
@@ -119,6 +121,9 @@
// of suspension on input.
status_t setMaxTimestampGapUs(int64_t maxGapUs);
+ // When set, the max frame rate fed to the encoder will be capped at maxFps.
+ status_t setMaxFps(float maxFps);
+
// Sets the time lapse (or slow motion) parameters.
// data[0] is the time (us) between two frames for playback
// data[1] is the time (us) between two frames for capture
@@ -193,8 +198,8 @@
// doing anything if we don't have a codec buffer available.
void submitEndOfInputStream_l();
- void setLatestSubmittedBuffer_l(const BufferItem &item);
- bool repeatLatestSubmittedBuffer_l();
+ void setLatestBuffer_l(const BufferItem &item, bool dropped);
+ bool repeatLatestBuffer_l();
int64_t getTimestamp(const BufferItem &item);
// Lock, covers all member variables.
@@ -250,6 +255,8 @@
int64_t mPrevModifiedTimeUs;
int64_t mSkipFramesBeforeNs;
+ sp<FrameDropper> mFrameDropper;
+
sp<ALooper> mLooper;
sp<AHandlerReflector<GraphicBufferSource> > mReflector;
@@ -258,11 +265,11 @@
int64_t mRepeatLastFrameTimestamp;
int32_t mRepeatLastFrameCount;
- int mLatestSubmittedBufferId;
- uint64_t mLatestSubmittedBufferFrameNum;
- int32_t mLatestSubmittedBufferUseCount;
+ int mLatestBufferId;
+ uint64_t mLatestBufferFrameNum;
+ int32_t mLatestBufferUseCount;
- // The previously submitted buffer should've been repeated but
+ // The previous buffer should've been repeated but
// no codec buffer was available at the time.
bool mRepeatBufferDeferred;
diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index ab7419f..4779d6a 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -1075,6 +1075,7 @@
case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY:
return "REPEAT_PREVIOUS_FRAME_DELAY";
case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP: return "MAX_TIMESTAMP_GAP";
+ case IOMX::INTERNAL_OPTION_MAX_FPS: return "MAX_FPS";
case IOMX::INTERNAL_OPTION_START_TIME: return "START_TIME";
case IOMX::INTERNAL_OPTION_TIME_LAPSE: return "TIME_LAPSE";
default: return def;
@@ -1092,6 +1093,7 @@
case IOMX::INTERNAL_OPTION_SUSPEND:
case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY:
case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP:
+ case IOMX::INTERNAL_OPTION_MAX_FPS:
case IOMX::INTERNAL_OPTION_START_TIME:
case IOMX::INTERNAL_OPTION_TIME_LAPSE:
{
@@ -1129,6 +1131,14 @@
int64_t maxGapUs = *(int64_t *)data;
CLOG_CONFIG(setInternalOption, "gapUs=%lld", (long long)maxGapUs);
return bufferSource->setMaxTimestampGapUs(maxGapUs);
+ } else if (type == IOMX::INTERNAL_OPTION_MAX_FPS) {
+ if (size != sizeof(float)) {
+ return INVALID_OPERATION;
+ }
+
+ float maxFps = *(float *)data;
+ CLOG_CONFIG(setInternalOption, "maxFps=%f", maxFps);
+ return bufferSource->setMaxFps(maxFps);
} else if (type == IOMX::INTERNAL_OPTION_START_TIME) {
if (size != sizeof(int64_t)) {
return INVALID_OPERATION;
diff --git a/media/libstagefright/omx/SimpleSoftOMXComponent.cpp b/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
index 7f99dcd..801a1bd 100644
--- a/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
+++ b/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
@@ -58,7 +58,7 @@
OMX_COMMANDTYPE cmd, OMX_U32 param, OMX_PTR data) {
CHECK(data == NULL);
- sp<AMessage> msg = new AMessage(kWhatSendCommand, mHandler->id());
+ sp<AMessage> msg = new AMessage(kWhatSendCommand, mHandler);
msg->setInt32("cmd", cmd);
msg->setInt32("param", param);
msg->post();
@@ -307,7 +307,7 @@
OMX_ERRORTYPE SimpleSoftOMXComponent::emptyThisBuffer(
OMX_BUFFERHEADERTYPE *buffer) {
- sp<AMessage> msg = new AMessage(kWhatEmptyThisBuffer, mHandler->id());
+ sp<AMessage> msg = new AMessage(kWhatEmptyThisBuffer, mHandler);
msg->setPointer("header", buffer);
msg->post();
@@ -316,7 +316,7 @@
OMX_ERRORTYPE SimpleSoftOMXComponent::fillThisBuffer(
OMX_BUFFERHEADERTYPE *buffer) {
- sp<AMessage> msg = new AMessage(kWhatFillThisBuffer, mHandler->id());
+ sp<AMessage> msg = new AMessage(kWhatFillThisBuffer, mHandler);
msg->setPointer("header", buffer);
msg->post();
diff --git a/media/libstagefright/omx/tests/Android.mk b/media/libstagefright/omx/tests/Android.mk
index 447b29e..9be637a 100644
--- a/media/libstagefright/omx/tests/Android.mk
+++ b/media/libstagefright/omx/tests/Android.mk
@@ -20,3 +20,21 @@
LOCAL_32_BIT_ONLY := true
include $(BUILD_EXECUTABLE)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := FrameDropper_test
+
+LOCAL_MODULE_TAGS := tests
+
+LOCAL_SRC_FILES := \
+ FrameDropper_test.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright_omx \
+ libutils \
+
+LOCAL_C_INCLUDES := \
+ frameworks/av/media/libstagefright/omx \
+
+include $(BUILD_NATIVE_TEST)
diff --git a/media/libstagefright/omx/tests/FrameDropper_test.cpp b/media/libstagefright/omx/tests/FrameDropper_test.cpp
new file mode 100644
index 0000000..4ac72c4
--- /dev/null
+++ b/media/libstagefright/omx/tests/FrameDropper_test.cpp
@@ -0,0 +1,136 @@
+/*
+ * Copyright (C) 2015 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 "FrameDropper_test"
+#include <utils/Log.h>
+
+#include <gtest/gtest.h>
+
+#include "FrameDropper.h"
+#include <media/stagefright/foundation/ADebug.h>
+
+namespace android {
+
+struct TestFrame {
+ int64_t timeUs;
+ bool shouldDrop;
+};
+
+static const TestFrame testFrames20Fps[] = {
+ {1000000, false}, {1050000, false}, {1100000, false}, {1150000, false},
+ {1200000, false}, {1250000, false}, {1300000, false}, {1350000, false},
+ {1400000, false}, {1450000, false}, {1500000, false}, {1550000, false},
+ {1600000, false}, {1650000, false}, {1700000, false}, {1750000, false},
+ {1800000, false}, {1850000, false}, {1900000, false}, {1950000, false},
+};
+
+static const TestFrame testFrames30Fps[] = {
+ {1000000, false}, {1033333, false}, {1066667, false}, {1100000, false},
+ {1133333, false}, {1166667, false}, {1200000, false}, {1233333, false},
+ {1266667, false}, {1300000, false}, {1333333, false}, {1366667, false},
+ {1400000, false}, {1433333, false}, {1466667, false}, {1500000, false},
+ {1533333, false}, {1566667, false}, {1600000, false}, {1633333, false},
+};
+
+static const TestFrame testFrames40Fps[] = {
+ {1000000, false}, {1025000, true}, {1050000, false}, {1075000, false},
+ {1100000, false}, {1125000, true}, {1150000, false}, {1175000, false},
+ {1200000, false}, {1225000, true}, {1250000, false}, {1275000, false},
+ {1300000, false}, {1325000, true}, {1350000, false}, {1375000, false},
+ {1400000, false}, {1425000, true}, {1450000, false}, {1475000, false},
+};
+
+static const TestFrame testFrames60Fps[] = {
+ {1000000, false}, {1016667, true}, {1033333, false}, {1050000, true},
+ {1066667, false}, {1083333, true}, {1100000, false}, {1116667, true},
+ {1133333, false}, {1150000, true}, {1166667, false}, {1183333, true},
+ {1200000, false}, {1216667, true}, {1233333, false}, {1250000, true},
+ {1266667, false}, {1283333, true}, {1300000, false}, {1316667, true},
+};
+
+static const TestFrame testFramesVariableFps[] = {
+ // 40fps
+ {1000000, false}, {1025000, true}, {1050000, false}, {1075000, false},
+ {1100000, false}, {1125000, true}, {1150000, false}, {1175000, false},
+ {1200000, false}, {1225000, true}, {1250000, false}, {1275000, false},
+ {1300000, false}, {1325000, true}, {1350000, false}, {1375000, false},
+ {1400000, false}, {1425000, true}, {1450000, false}, {1475000, false},
+ // a timestamp jump plus switch to 20fps
+ {2000000, false}, {2050000, false}, {2100000, false}, {2150000, false},
+ {2200000, false}, {2250000, false}, {2300000, false}, {2350000, false},
+ {2400000, false}, {2450000, false}, {2500000, false}, {2550000, false},
+ {2600000, false}, {2650000, false}, {2700000, false}, {2750000, false},
+ {2800000, false}, {2850000, false}, {2900000, false}, {2950000, false},
+ // 60fps
+ {2966667, false}, {2983333, true}, {3000000, false}, {3016667, true},
+ {3033333, false}, {3050000, true}, {3066667, false}, {3083333, true},
+ {3100000, false}, {3116667, true}, {3133333, false}, {3150000, true},
+ {3166667, false}, {3183333, true}, {3200000, false}, {3216667, true},
+ {3233333, false}, {3250000, true}, {3266667, false}, {3283333, true},
+};
+
+static const int kMaxTestJitterUs = 2000;
+// return one of 1000, 0, -1000 as jitter.
+static int GetJitter(size_t i) {
+ return (1 - (i % 3)) * (kMaxTestJitterUs / 2);
+}
+
+class FrameDropperTest : public ::testing::Test {
+public:
+ FrameDropperTest() : mFrameDropper(new FrameDropper()) {
+ EXPECT_EQ(OK, mFrameDropper->setMaxFrameRate(30.0));
+ }
+
+protected:
+ void RunTest(const TestFrame* frames, size_t size) {
+ for (size_t i = 0; i < size; ++i) {
+ int jitter = GetJitter(i);
+ int64_t testTimeUs = frames[i].timeUs + jitter;
+ printf("time %lld, testTime %lld, jitter %d\n", frames[i].timeUs, testTimeUs, jitter);
+ EXPECT_EQ(frames[i].shouldDrop, mFrameDropper->shouldDrop(testTimeUs));
+ }
+ }
+
+ sp<FrameDropper> mFrameDropper;
+};
+
+TEST_F(FrameDropperTest, TestInvalidMaxFrameRate) {
+ EXPECT_NE(OK, mFrameDropper->setMaxFrameRate(-1.0));
+ EXPECT_NE(OK, mFrameDropper->setMaxFrameRate(0));
+}
+
+TEST_F(FrameDropperTest, Test20Fps) {
+ RunTest(testFrames20Fps, ARRAY_SIZE(testFrames20Fps));
+}
+
+TEST_F(FrameDropperTest, Test30Fps) {
+ RunTest(testFrames30Fps, ARRAY_SIZE(testFrames30Fps));
+}
+
+TEST_F(FrameDropperTest, Test40Fps) {
+ RunTest(testFrames40Fps, ARRAY_SIZE(testFrames40Fps));
+}
+
+TEST_F(FrameDropperTest, Test60Fps) {
+ RunTest(testFrames60Fps, ARRAY_SIZE(testFrames60Fps));
+}
+
+TEST_F(FrameDropperTest, TestVariableFps) {
+ RunTest(testFramesVariableFps, ARRAY_SIZE(testFramesVariableFps));
+}
+
+} // namespace android
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index a6bd824..a86ab74 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -82,7 +82,7 @@
size_t index,
const sp<AMessage> ¬ify,
bool injected) {
- sp<AMessage> msg = new AMessage(kWhatAddStream, id());
+ sp<AMessage> msg = new AMessage(kWhatAddStream, this);
msg->setInt32("rtp-socket", rtpSocket);
msg->setInt32("rtcp-socket", rtcpSocket);
msg->setObject("session-desc", sessionDesc);
@@ -93,7 +93,7 @@
}
void ARTPConnection::removeStream(int rtpSocket, int rtcpSocket) {
- sp<AMessage> msg = new AMessage(kWhatRemoveStream, id());
+ sp<AMessage> msg = new AMessage(kWhatRemoveStream, this);
msg->setInt32("rtp-socket", rtpSocket);
msg->setInt32("rtcp-socket", rtcpSocket);
msg->post();
@@ -233,7 +233,7 @@
return;
}
- sp<AMessage> msg = new AMessage(kWhatPollStreams, id());
+ sp<AMessage> msg = new AMessage(kWhatPollStreams, this);
msg->post();
mPollEventPending = true;
@@ -639,7 +639,7 @@
}
void ARTPConnection::injectPacket(int index, const sp<ABuffer> &buffer) {
- sp<AMessage> msg = new AMessage(kWhatInjectPacket, id());
+ sp<AMessage> msg = new AMessage(kWhatInjectPacket, this);
msg->setInt32("index", index);
msg->setBuffer("buffer", buffer);
msg->post();
diff --git a/media/libstagefright/rtsp/ARTPSession.cpp b/media/libstagefright/rtsp/ARTPSession.cpp
index ba4e33c..e5acb06 100644
--- a/media/libstagefright/rtsp/ARTPSession.cpp
+++ b/media/libstagefright/rtsp/ARTPSession.cpp
@@ -82,7 +82,7 @@
info->mRTPSocket = rtpSocket;
info->mRTCPSocket = rtcpSocket;
- sp<AMessage> notify = new AMessage(kWhatAccessUnitComplete, id());
+ sp<AMessage> notify = new AMessage(kWhatAccessUnitComplete, this);
notify->setSize("track-index", mTracks.size() - 1);
mRTPConn->addStream(
diff --git a/media/libstagefright/rtsp/ARTPWriter.cpp b/media/libstagefright/rtsp/ARTPWriter.cpp
index e1607bf..56c4aa6 100644
--- a/media/libstagefright/rtsp/ARTPWriter.cpp
+++ b/media/libstagefright/rtsp/ARTPWriter.cpp
@@ -146,7 +146,7 @@
TRESPASS();
}
- (new AMessage(kWhatStart, mReflector->id()))->post();
+ (new AMessage(kWhatStart, mReflector))->post();
while (!(mFlags & kFlagStarted)) {
mCondition.wait(mLock);
@@ -161,7 +161,7 @@
return OK;
}
- (new AMessage(kWhatStop, mReflector->id()))->post();
+ (new AMessage(kWhatStop, mReflector))->post();
while (mFlags & kFlagStarted) {
mCondition.wait(mLock);
@@ -213,8 +213,8 @@
mCondition.signal();
}
- (new AMessage(kWhatRead, mReflector->id()))->post();
- (new AMessage(kWhatSendSR, mReflector->id()))->post();
+ (new AMessage(kWhatRead, mReflector))->post();
+ (new AMessage(kWhatSendSR, mReflector))->post();
break;
}
diff --git a/media/libstagefright/rtsp/ARTSPConnection.cpp b/media/libstagefright/rtsp/ARTSPConnection.cpp
index 60b3aaf..855ffdc 100644
--- a/media/libstagefright/rtsp/ARTSPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTSPConnection.cpp
@@ -68,28 +68,28 @@
}
void ARTSPConnection::connect(const char *url, const sp<AMessage> &reply) {
- sp<AMessage> msg = new AMessage(kWhatConnect, id());
+ sp<AMessage> msg = new AMessage(kWhatConnect, this);
msg->setString("url", url);
msg->setMessage("reply", reply);
msg->post();
}
void ARTSPConnection::disconnect(const sp<AMessage> &reply) {
- sp<AMessage> msg = new AMessage(kWhatDisconnect, id());
+ sp<AMessage> msg = new AMessage(kWhatDisconnect, this);
msg->setMessage("reply", reply);
msg->post();
}
void ARTSPConnection::sendRequest(
const char *request, const sp<AMessage> &reply) {
- sp<AMessage> msg = new AMessage(kWhatSendRequest, id());
+ sp<AMessage> msg = new AMessage(kWhatSendRequest, this);
msg->setString("request", request);
msg->setMessage("reply", reply);
msg->post();
}
void ARTSPConnection::observeBinaryData(const sp<AMessage> &reply) {
- sp<AMessage> msg = new AMessage(kWhatObserveBinaryData, id());
+ sp<AMessage> msg = new AMessage(kWhatObserveBinaryData, this);
msg->setMessage("reply", reply);
msg->post();
}
@@ -286,7 +286,7 @@
if (err < 0) {
if (errno == EINPROGRESS) {
- sp<AMessage> msg = new AMessage(kWhatCompleteConnection, id());
+ sp<AMessage> msg = new AMessage(kWhatCompleteConnection, this);
msg->setMessage("reply", reply);
msg->setInt32("connection-id", mConnectionID);
msg->post();
@@ -523,7 +523,7 @@
return;
}
- sp<AMessage> msg = new AMessage(kWhatReceiveResponse, id());
+ sp<AMessage> msg = new AMessage(kWhatReceiveResponse, this);
msg->post();
mReceiveResponseEventPending = true;
@@ -746,7 +746,7 @@
AString request;
CHECK(reply->findString("original-request", &request));
- sp<AMessage> msg = new AMessage(kWhatSendRequest, id());
+ sp<AMessage> msg = new AMessage(kWhatSendRequest, this);
msg->setMessage("reply", reply);
msg->setString("request", request.c_str(), request.size());
diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index 3bf489b..0642343 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -169,10 +169,10 @@
looper()->registerHandler(mConn);
(1 ? mNetLooper : looper())->registerHandler(mRTPConn);
- sp<AMessage> notify = new AMessage('biny', id());
+ sp<AMessage> notify = new AMessage('biny', this);
mConn->observeBinaryData(notify);
- sp<AMessage> reply = new AMessage('conn', id());
+ sp<AMessage> reply = new AMessage('conn', this);
mConn->connect(mOriginalSessionURL.c_str(), reply);
}
@@ -180,10 +180,10 @@
looper()->registerHandler(mConn);
(1 ? mNetLooper : looper())->registerHandler(mRTPConn);
- sp<AMessage> notify = new AMessage('biny', id());
+ sp<AMessage> notify = new AMessage('biny', this);
mConn->observeBinaryData(notify);
- sp<AMessage> reply = new AMessage('sdpl', id());
+ sp<AMessage> reply = new AMessage('sdpl', this);
reply->setObject("description", desc);
mConn->connect(mOriginalSessionURL.c_str(), reply);
}
@@ -210,11 +210,11 @@
}
void disconnect() {
- (new AMessage('abor', id()))->post();
+ (new AMessage('abor', this))->post();
}
void seek(int64_t timeUs) {
- sp<AMessage> msg = new AMessage('seek', id());
+ sp<AMessage> msg = new AMessage('seek', this);
msg->setInt64("time", timeUs);
mPauseGeneration++;
msg->post();
@@ -225,14 +225,14 @@
}
void pause() {
- sp<AMessage> msg = new AMessage('paus', id());
+ sp<AMessage> msg = new AMessage('paus', this);
mPauseGeneration++;
msg->setInt32("pausecheck", mPauseGeneration);
msg->post(kPauseDelayUs);
}
void resume() {
- sp<AMessage> msg = new AMessage('resu', id());
+ sp<AMessage> msg = new AMessage('resu', this);
mPauseGeneration++;
msg->post();
}
@@ -454,10 +454,10 @@
request.append("Accept: application/sdp\r\n");
request.append("\r\n");
- sp<AMessage> reply = new AMessage('desc', id());
+ sp<AMessage> reply = new AMessage('desc', this);
mConn->sendRequest(request.c_str(), reply);
} else {
- (new AMessage('disc', id()))->post();
+ (new AMessage('disc', this))->post();
}
break;
}
@@ -468,10 +468,10 @@
int32_t reconnect;
if (msg->findInt32("reconnect", &reconnect) && reconnect) {
- sp<AMessage> reply = new AMessage('conn', id());
+ sp<AMessage> reply = new AMessage('conn', this);
mConn->connect(mOriginalSessionURL.c_str(), reply);
} else {
- (new AMessage('quit', id()))->post();
+ (new AMessage('quit', this))->post();
}
break;
}
@@ -514,7 +514,7 @@
ALOGI("rewritten session url: '%s'", mSessionURL.c_str());
}
- sp<AMessage> reply = new AMessage('conn', id());
+ sp<AMessage> reply = new AMessage('conn', this);
mConn->connect(mOriginalSessionURL.c_str(), reply);
break;
}
@@ -586,7 +586,7 @@
}
if (result != OK) {
- sp<AMessage> reply = new AMessage('disc', id());
+ sp<AMessage> reply = new AMessage('disc', this);
mConn->disconnect(reply);
}
break;
@@ -631,7 +631,7 @@
}
if (result != OK) {
- sp<AMessage> reply = new AMessage('disc', id());
+ sp<AMessage> reply = new AMessage('disc', this);
mConn->disconnect(reply);
}
break;
@@ -703,7 +703,7 @@
mSessionID.erase(i, mSessionID.size() - i);
}
- sp<AMessage> notify = new AMessage('accu', id());
+ sp<AMessage> notify = new AMessage('accu', this);
notify->setSize("track-index", trackIndex);
i = response->mHeaders.indexOfKey("transport");
@@ -769,10 +769,10 @@
request.append("\r\n");
- sp<AMessage> reply = new AMessage('play', id());
+ sp<AMessage> reply = new AMessage('play', this);
mConn->sendRequest(request.c_str(), reply);
} else {
- sp<AMessage> reply = new AMessage('disc', id());
+ sp<AMessage> reply = new AMessage('disc', this);
mConn->disconnect(reply);
}
break;
@@ -797,7 +797,7 @@
} else {
parsePlayResponse(response);
- sp<AMessage> timeout = new AMessage('tiou', id());
+ sp<AMessage> timeout = new AMessage('tiou', this);
mCheckTimeoutGeneration++;
timeout->setInt32("tioucheck", mCheckTimeoutGeneration);
timeout->post(kStartupTimeoutUs);
@@ -805,7 +805,7 @@
}
if (result != OK) {
- sp<AMessage> reply = new AMessage('disc', id());
+ sp<AMessage> reply = new AMessage('disc', this);
mConn->disconnect(reply);
}
@@ -831,7 +831,7 @@
request.append("\r\n");
request.append("\r\n");
- sp<AMessage> reply = new AMessage('opts', id());
+ sp<AMessage> reply = new AMessage('opts', this);
reply->setInt32("generation", mKeepAliveGeneration);
mConn->sendRequest(request.c_str(), reply);
break;
@@ -894,7 +894,7 @@
mPausing = false;
mSeekable = true;
- sp<AMessage> reply = new AMessage('tear', id());
+ sp<AMessage> reply = new AMessage('tear', this);
int32_t reconnect;
if (msg->findInt32("reconnect", &reconnect) && reconnect) {
@@ -926,7 +926,7 @@
ALOGI("TEARDOWN completed with result %d (%s)",
result, strerror(-result));
- sp<AMessage> reply = new AMessage('disc', id());
+ sp<AMessage> reply = new AMessage('disc', this);
int32_t reconnect;
if (msg->findInt32("reconnect", &reconnect) && reconnect) {
@@ -958,7 +958,7 @@
if (mNumAccessUnitsReceived == 0) {
#if 1
ALOGI("stream ended? aborting.");
- (new AMessage('abor', id()))->post();
+ (new AMessage('abor', this))->post();
break;
#else
ALOGI("haven't seen an AU in a looong time.");
@@ -1077,7 +1077,7 @@
request.append("\r\n");
- sp<AMessage> reply = new AMessage('pau2', id());
+ sp<AMessage> reply = new AMessage('pau2', this);
mConn->sendRequest(request.c_str(), reply);
break;
}
@@ -1114,7 +1114,7 @@
request.append("\r\n");
- sp<AMessage> reply = new AMessage('res2', id());
+ sp<AMessage> reply = new AMessage('res2', this);
mConn->sendRequest(request.c_str(), reply);
break;
}
@@ -1143,7 +1143,7 @@
// Post new timeout in order to make sure to use
// fake timestamps if no new Sender Reports arrive
- sp<AMessage> timeout = new AMessage('tiou', id());
+ sp<AMessage> timeout = new AMessage('tiou', this);
mCheckTimeoutGeneration++;
timeout->setInt32("tioucheck", mCheckTimeoutGeneration);
timeout->post(kStartupTimeoutUs);
@@ -1152,7 +1152,7 @@
if (result != OK) {
ALOGE("resume failed, aborting.");
- (new AMessage('abor', id()))->post();
+ (new AMessage('abor', this))->post();
}
mPausing = false;
@@ -1180,7 +1180,7 @@
mCheckPending = true;
++mCheckGeneration;
- sp<AMessage> reply = new AMessage('see1', id());
+ sp<AMessage> reply = new AMessage('see1', this);
reply->setInt64("time", timeUs);
if (mPausing) {
@@ -1221,7 +1221,7 @@
// Start new timeoutgeneration to avoid getting timeout
// before PLAY response arrive
- sp<AMessage> timeout = new AMessage('tiou', id());
+ sp<AMessage> timeout = new AMessage('tiou', this);
mCheckTimeoutGeneration++;
timeout->setInt32("tioucheck", mCheckTimeoutGeneration);
timeout->post(kStartupTimeoutUs);
@@ -1243,7 +1243,7 @@
request.append("\r\n");
- sp<AMessage> reply = new AMessage('see2', id());
+ sp<AMessage> reply = new AMessage('see2', this);
mConn->sendRequest(request.c_str(), reply);
break;
}
@@ -1277,7 +1277,7 @@
// Post new timeout in order to make sure to use
// fake timestamps if no new Sender Reports arrive
- sp<AMessage> timeout = new AMessage('tiou', id());
+ sp<AMessage> timeout = new AMessage('tiou', this);
mCheckTimeoutGeneration++;
timeout->setInt32("tioucheck", mCheckTimeoutGeneration);
timeout->post(kStartupTimeoutUs);
@@ -1293,7 +1293,7 @@
if (result != OK) {
ALOGE("seek failed, aborting.");
- (new AMessage('abor', id()))->post();
+ (new AMessage('abor', this))->post();
}
mPausing = false;
@@ -1343,12 +1343,12 @@
mTryTCPInterleaving = true;
- sp<AMessage> msg = new AMessage('abor', id());
+ sp<AMessage> msg = new AMessage('abor', this);
msg->setInt32("reconnect", true);
msg->post();
} else {
ALOGW("Never received any data, disconnecting.");
- (new AMessage('abor', id()))->post();
+ (new AMessage('abor', this))->post();
}
} else {
if (!mAllTracksHaveTime) {
@@ -1369,7 +1369,7 @@
}
void postKeepAlive() {
- sp<AMessage> msg = new AMessage('aliv', id());
+ sp<AMessage> msg = new AMessage('aliv', this);
msg->setInt32("generation", mKeepAliveGeneration);
msg->post((mKeepAliveTimeoutUs * 9) / 10);
}
@@ -1380,7 +1380,7 @@
}
mCheckPending = true;
- sp<AMessage> check = new AMessage('chek', id());
+ sp<AMessage> check = new AMessage('chek', this);
check->setInt32("generation", mCheckGeneration);
check->post(kAccessUnitTimeoutUs);
}
@@ -1566,7 +1566,7 @@
if (source->initCheck() != OK) {
ALOGW("Unsupported format. Ignoring track #%d.", index);
- sp<AMessage> reply = new AMessage('setu', id());
+ sp<AMessage> reply = new AMessage('setu', this);
reply->setSize("index", index);
reply->setInt32("result", ERROR_UNSUPPORTED);
reply->post();
@@ -1652,7 +1652,7 @@
request.append("\r\n");
- sp<AMessage> reply = new AMessage('setu', id());
+ sp<AMessage> reply = new AMessage('setu', this);
reply->setSize("index", index);
reply->setSize("track-index", mTracks.size() - 1);
mConn->sendRequest(request.c_str(), reply);
diff --git a/media/libstagefright/rtsp/MyTransmitter.h b/media/libstagefright/rtsp/MyTransmitter.h
index 009a3b1..369f276 100644
--- a/media/libstagefright/rtsp/MyTransmitter.h
+++ b/media/libstagefright/rtsp/MyTransmitter.h
@@ -100,7 +100,7 @@
mLooper->registerHandler(this);
mLooper->registerHandler(mConn);
- sp<AMessage> reply = new AMessage('conn', id());
+ sp<AMessage> reply = new AMessage('conn', this);
mConn->connect(mServerURL.c_str(), reply);
#ifdef ANDROID
@@ -229,7 +229,7 @@
request.append("\r\n");
request.append(sdp);
- sp<AMessage> reply = new AMessage('anno', id());
+ sp<AMessage> reply = new AMessage('anno', this);
mConn->sendRequest(request.c_str(), reply);
}
@@ -350,7 +350,7 @@
<< result << " (" << strerror(-result) << ")";
if (result != OK) {
- (new AMessage('quit', id()))->post();
+ (new AMessage('quit', this))->post();
break;
}
@@ -381,7 +381,7 @@
if (response->mStatusCode == 401) {
if (mAuthType != NONE) {
LOG(INFO) << "FAILED to authenticate";
- (new AMessage('quit', id()))->post();
+ (new AMessage('quit', this))->post();
break;
}
@@ -391,14 +391,14 @@
}
if (result != OK || response->mStatusCode != 200) {
- (new AMessage('quit', id()))->post();
+ (new AMessage('quit', this))->post();
break;
}
unsigned rtpPort;
ARTPConnection::MakePortPair(&mRTPSocket, &mRTCPSocket, &rtpPort);
- // (new AMessage('poll', id()))->post();
+ // (new AMessage('poll', this))->post();
AString request;
request.append("SETUP ");
@@ -414,7 +414,7 @@
request.append(";mode=record\r\n");
request.append("\r\n");
- sp<AMessage> reply = new AMessage('setu', id());
+ sp<AMessage> reply = new AMessage('setu', this);
mConn->sendRequest(request.c_str(), reply);
break;
}
@@ -468,7 +468,7 @@
}
if (result != OK || response->mStatusCode != 200) {
- (new AMessage('quit', id()))->post();
+ (new AMessage('quit', this))->post();
break;
}
@@ -535,7 +535,7 @@
request.append("\r\n");
request.append("\r\n");
- sp<AMessage> reply = new AMessage('reco', id());
+ sp<AMessage> reply = new AMessage('reco', this);
mConn->sendRequest(request.c_str(), reply);
break;
}
@@ -558,13 +558,13 @@
}
if (result != OK) {
- (new AMessage('quit', id()))->post();
+ (new AMessage('quit', this))->post();
break;
}
- (new AMessage('more', id()))->post();
- (new AMessage('sr ', id()))->post();
- (new AMessage('aliv', id()))->post(30000000ll);
+ (new AMessage('more', this))->post();
+ (new AMessage('sr ', this))->post();
+ (new AMessage('aliv', this))->post(30000000ll);
break;
}
@@ -586,7 +586,7 @@
request.append("\r\n");
request.append("\r\n");
- sp<AMessage> reply = new AMessage('opts', id());
+ sp<AMessage> reply = new AMessage('opts', this);
mConn->sendRequest(request.c_str(), reply);
break;
}
@@ -603,7 +603,7 @@
break;
}
- (new AMessage('aliv', id()))->post(30000000ll);
+ (new AMessage('aliv', this))->post(30000000ll);
break;
}
@@ -702,7 +702,7 @@
request.append("\r\n");
request.append("\r\n");
- sp<AMessage> reply = new AMessage('paus', id());
+ sp<AMessage> reply = new AMessage('paus', this);
mConn->sendRequest(request.c_str(), reply);
}
break;
@@ -753,7 +753,7 @@
request.append("\r\n");
request.append("\r\n");
- sp<AMessage> reply = new AMessage('tear', id());
+ sp<AMessage> reply = new AMessage('tear', this);
mConn->sendRequest(request.c_str(), reply);
break;
}
@@ -775,7 +775,7 @@
CHECK(response != NULL);
}
- (new AMessage('quit', id()))->post();
+ (new AMessage('quit', this))->post();
break;
}
@@ -784,14 +784,14 @@
LOG(INFO) << "disconnect completed";
mConnected = false;
- (new AMessage('quit', id()))->post();
+ (new AMessage('quit', this))->post();
break;
}
case 'quit':
{
if (mConnected) {
- mConn->disconnect(new AMessage('disc', id()));
+ mConn->disconnect(new AMessage('disc', this));
break;
}
diff --git a/media/libstagefright/rtsp/SDPLoader.cpp b/media/libstagefright/rtsp/SDPLoader.cpp
index a24eb69..0f46c83 100644
--- a/media/libstagefright/rtsp/SDPLoader.cpp
+++ b/media/libstagefright/rtsp/SDPLoader.cpp
@@ -51,7 +51,7 @@
void SDPLoader::load(const char *url, const KeyedVector<String8, String8> *headers) {
mNetLooper->registerHandler(this);
- sp<AMessage> msg = new AMessage(kWhatLoad, id());
+ sp<AMessage> msg = new AMessage(kWhatLoad, this);
msg->setString("url", url);
if (headers != NULL) {
diff --git a/media/libstagefright/rtsp/UDPPusher.cpp b/media/libstagefright/rtsp/UDPPusher.cpp
index 47ea6f1..5c685a1 100644
--- a/media/libstagefright/rtsp/UDPPusher.cpp
+++ b/media/libstagefright/rtsp/UDPPusher.cpp
@@ -65,7 +65,7 @@
mFirstTimeMs = fromlel(timeMs);
mFirstTimeUs = ALooper::GetNowUs();
- (new AMessage(kWhatPush, id()))->post();
+ (new AMessage(kWhatPush, this))->post();
}
bool UDPPusher::onPush() {
@@ -103,7 +103,7 @@
timeMs -= mFirstTimeMs;
int64_t whenUs = mFirstTimeUs + timeMs * 1000ll;
int64_t nowUs = ALooper::GetNowUs();
- (new AMessage(kWhatPush, id()))->post(whenUs - nowUs);
+ (new AMessage(kWhatPush, this))->post(whenUs - nowUs);
return true;
}
diff --git a/media/libstagefright/timedtext/TimedTextPlayer.cpp b/media/libstagefright/timedtext/TimedTextPlayer.cpp
index a070487..aecf666 100644
--- a/media/libstagefright/timedtext/TimedTextPlayer.cpp
+++ b/media/libstagefright/timedtext/TimedTextPlayer.cpp
@@ -56,25 +56,25 @@
}
void TimedTextPlayer::start() {
- (new AMessage(kWhatStart, id()))->post();
+ (new AMessage(kWhatStart, this))->post();
}
void TimedTextPlayer::pause() {
- (new AMessage(kWhatPause, id()))->post();
+ (new AMessage(kWhatPause, this))->post();
}
void TimedTextPlayer::resume() {
- (new AMessage(kWhatResume, id()))->post();
+ (new AMessage(kWhatResume, this))->post();
}
void TimedTextPlayer::seekToAsync(int64_t timeUs) {
- sp<AMessage> msg = new AMessage(kWhatSeek, id());
+ sp<AMessage> msg = new AMessage(kWhatSeek, this);
msg->setInt64("seekTimeUs", timeUs);
msg->post();
}
void TimedTextPlayer::setDataSource(sp<TimedTextSource> source) {
- sp<AMessage> msg = new AMessage(kWhatSetSource, id());
+ sp<AMessage> msg = new AMessage(kWhatSetSource, this);
msg->setObject("source", source);
msg->post();
}
@@ -231,7 +231,7 @@
status_t err = mSource->read(&startTimeUs, &endTimeUs,
&(parcelEvent->parcel), options);
if (err == WOULD_BLOCK) {
- sp<AMessage> msg = new AMessage(kWhatRetryRead, id());
+ sp<AMessage> msg = new AMessage(kWhatRetryRead, this);
if (options != NULL) {
int64_t seekTimeUs = kInvalidTimeUs;
MediaSource::ReadOptions::SeekMode seekMode =
@@ -259,7 +259,7 @@
void TimedTextPlayer::postTextEvent(const sp<ParcelEvent>& parcel, int64_t timeUs) {
int64_t delayUs = delayUsFromCurrentTime(timeUs);
- sp<AMessage> msg = new AMessage(kWhatSendSubtitle, id());
+ sp<AMessage> msg = new AMessage(kWhatSendSubtitle, this);
msg->setInt32("generation", mSendSubtitleGeneration);
if (parcel != NULL) {
msg->setObject("subtitle", parcel);
diff --git a/media/libstagefright/webm/WebmWriter.cpp b/media/libstagefright/webm/WebmWriter.cpp
index 069961b..737f144 100644
--- a/media/libstagefright/webm/WebmWriter.cpp
+++ b/media/libstagefright/webm/WebmWriter.cpp
@@ -80,38 +80,6 @@
mCuePoints);
}
-WebmWriter::WebmWriter(const char *filename)
- : mInitCheck(NO_INIT),
- mTimeCodeScale(1000000),
- mStartTimestampUs(0),
- mStartTimeOffsetMs(0),
- mSegmentOffset(0),
- mSegmentDataStart(0),
- mInfoOffset(0),
- mInfoSize(0),
- mTracksOffset(0),
- mCuesOffset(0),
- mPaused(false),
- mStarted(false),
- mIsFileSizeLimitExplicitlyRequested(false),
- mIsRealTimeRecording(false),
- mStreamableFile(true),
- mEstimatedCuesSize(0) {
- mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
- if (mFd >= 0) {
- ALOGV("fd %d; flags: %o", mFd, fcntl(mFd, F_GETFL, 0));
- mInitCheck = OK;
- }
- mStreams[kAudioIndex] = WebmStream(kAudioType, "Audio", &WebmWriter::audioTrack);
- mStreams[kVideoIndex] = WebmStream(kVideoType, "Video", &WebmWriter::videoTrack);
- mSinkThread = new WebmFrameSinkThread(
- mFd,
- mSegmentDataStart,
- mStreams[kVideoIndex].mSink,
- mStreams[kAudioIndex].mSink,
- mCuePoints);
-}
-
// static
sp<WebmElement> WebmWriter::videoTrack(const sp<MetaData>& md) {
int32_t width, height;
diff --git a/media/libstagefright/webm/WebmWriter.h b/media/libstagefright/webm/WebmWriter.h
index 36b6965..4ad770e 100644
--- a/media/libstagefright/webm/WebmWriter.h
+++ b/media/libstagefright/webm/WebmWriter.h
@@ -37,7 +37,6 @@
class WebmWriter : public MediaWriter {
public:
WebmWriter(int fd);
- WebmWriter(const char *filename);
~WebmWriter() { reset(); }
diff --git a/media/libstagefright/wifi-display/MediaSender.cpp b/media/libstagefright/wifi-display/MediaSender.cpp
index b1cdec0..6f0087f 100644
--- a/media/libstagefright/wifi-display/MediaSender.cpp
+++ b/media/libstagefright/wifi-display/MediaSender.cpp
@@ -121,7 +121,7 @@
}
if (err == OK) {
- sp<AMessage> notify = new AMessage(kWhatSenderNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatSenderNotify, this);
notify->setInt32("generation", mGeneration);
mTSSender = new RTPSender(mNetSession, notify);
looper()->registerHandler(mTSSender);
@@ -170,7 +170,7 @@
return INVALID_OPERATION;
}
- sp<AMessage> notify = new AMessage(kWhatSenderNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatSenderNotify, this);
notify->setInt32("generation", mGeneration);
notify->setSize("trackIndex", trackIndex);
diff --git a/media/libstagefright/wifi-display/rtp/RTPSender.cpp b/media/libstagefright/wifi-display/rtp/RTPSender.cpp
index e88a3bd5..4e72533 100644
--- a/media/libstagefright/wifi-display/rtp/RTPSender.cpp
+++ b/media/libstagefright/wifi-display/rtp/RTPSender.cpp
@@ -95,11 +95,11 @@
return INVALID_OPERATION;
}
- sp<AMessage> rtpNotify = new AMessage(kWhatRTPNotify, id());
+ sp<AMessage> rtpNotify = new AMessage(kWhatRTPNotify, this);
sp<AMessage> rtcpNotify;
if (remoteRTCPPort >= 0) {
- rtcpNotify = new AMessage(kWhatRTCPNotify, id());
+ rtcpNotify = new AMessage(kWhatRTCPNotify, this);
}
CHECK_EQ(mRTPSessionID, 0);
diff --git a/media/libstagefright/wifi-display/source/Converter.cpp b/media/libstagefright/wifi-display/source/Converter.cpp
index 2834a66..8368945 100644
--- a/media/libstagefright/wifi-display/source/Converter.cpp
+++ b/media/libstagefright/wifi-display/source/Converter.cpp
@@ -93,7 +93,7 @@
void Converter::shutdownAsync() {
ALOGV("shutdown");
- (new AMessage(kWhatShutdown, id()))->post();
+ (new AMessage(kWhatShutdown, this))->post();
}
status_t Converter::init() {
@@ -482,11 +482,11 @@
#if 1
if (mEncoderActivityNotify == NULL) {
- mEncoderActivityNotify = new AMessage(kWhatEncoderActivity, id());
+ mEncoderActivityNotify = new AMessage(kWhatEncoderActivity, this);
}
mEncoder->requestActivityNotification(mEncoderActivityNotify->dup());
#else
- sp<AMessage> notify = new AMessage(kWhatEncoderActivity, id());
+ sp<AMessage> notify = new AMessage(kWhatEncoderActivity, this);
notify->setInt64("whenUs", ALooper::GetNowUs());
mEncoder->requestActivityNotification(notify);
#endif
@@ -731,8 +731,7 @@
// MediaSender will post the following message when HDCP
// is done, to release the output buffer back to encoder.
- sp<AMessage> notify(new AMessage(
- kWhatReleaseOutputBuffer, id()));
+ sp<AMessage> notify(new AMessage(kWhatReleaseOutputBuffer, this));
notify->setInt32("bufferIndex", bufferIndex);
buffer = new ABuffer(
@@ -787,18 +786,18 @@
}
void Converter::requestIDRFrame() {
- (new AMessage(kWhatRequestIDRFrame, id()))->post();
+ (new AMessage(kWhatRequestIDRFrame, this))->post();
}
void Converter::dropAFrame() {
// Unsupported in surface input mode.
CHECK(!(mFlags & FLAG_USE_SURFACE_INPUT));
- (new AMessage(kWhatDropAFrame, id()))->post();
+ (new AMessage(kWhatDropAFrame, this))->post();
}
void Converter::suspendEncoding(bool suspend) {
- sp<AMessage> msg = new AMessage(kWhatSuspendEncoding, id());
+ sp<AMessage> msg = new AMessage(kWhatSuspendEncoding, this);
msg->setInt32("suspend", suspend);
msg->post();
}
diff --git a/media/libstagefright/wifi-display/source/MediaPuller.cpp b/media/libstagefright/wifi-display/source/MediaPuller.cpp
index 86b918f..ce07a4e 100644
--- a/media/libstagefright/wifi-display/source/MediaPuller.cpp
+++ b/media/libstagefright/wifi-display/source/MediaPuller.cpp
@@ -63,21 +63,21 @@
}
status_t MediaPuller::start() {
- return postSynchronouslyAndReturnError(new AMessage(kWhatStart, id()));
+ return postSynchronouslyAndReturnError(new AMessage(kWhatStart, this));
}
void MediaPuller::stopAsync(const sp<AMessage> ¬ify) {
- sp<AMessage> msg = new AMessage(kWhatStop, id());
+ sp<AMessage> msg = new AMessage(kWhatStop, this);
msg->setMessage("notify", notify);
msg->post();
}
void MediaPuller::pause() {
- (new AMessage(kWhatPause, id()))->post();
+ (new AMessage(kWhatPause, this))->post();
}
void MediaPuller::resume() {
- (new AMessage(kWhatResume, id()))->post();
+ (new AMessage(kWhatResume, this))->post();
}
void MediaPuller::onMessageReceived(const sp<AMessage> &msg) {
@@ -105,7 +105,7 @@
sp<AMessage> response = new AMessage;
response->setInt32("err", err);
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
response->postReply(replyID);
break;
@@ -215,7 +215,7 @@
}
void MediaPuller::schedulePull() {
- sp<AMessage> msg = new AMessage(kWhatPull, id());
+ sp<AMessage> msg = new AMessage(kWhatPull, this);
msg->setInt32("generation", mPullGeneration);
msg->post();
}
diff --git a/media/libstagefright/wifi-display/source/PlaybackSession.cpp b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
index 2cb4786..6080943 100644
--- a/media/libstagefright/wifi-display/source/PlaybackSession.cpp
+++ b/media/libstagefright/wifi-display/source/PlaybackSession.cpp
@@ -214,7 +214,7 @@
mConverter->shutdownAsync();
}
- sp<AMessage> msg = new AMessage(kWhatMediaPullerStopped, id());
+ sp<AMessage> msg = new AMessage(kWhatMediaPullerStopped, this);
if (mStarted && mMediaPuller != NULL) {
if (mRepeaterSource != NULL) {
@@ -382,7 +382,7 @@
size_t videoResolutionIndex,
VideoFormats::ProfileType videoProfileType,
VideoFormats::LevelType videoLevelType) {
- sp<AMessage> notify = new AMessage(kWhatMediaSenderNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatMediaSenderNotify, this);
mMediaSender = new MediaSender(mNetSession, notify);
looper()->registerHandler(mMediaSender);
@@ -440,7 +440,7 @@
status_t WifiDisplaySource::PlaybackSession::play() {
updateLiveness();
- (new AMessage(kWhatResume, id()))->post();
+ (new AMessage(kWhatResume, this))->post();
return OK;
}
@@ -460,7 +460,7 @@
status_t WifiDisplaySource::PlaybackSession::pause() {
updateLiveness();
- (new AMessage(kWhatPause, id()))->post();
+ (new AMessage(kWhatPause, this))->post();
return OK;
}
@@ -786,7 +786,7 @@
size_t trackIndex = mTracks.size();
- sp<AMessage> notify = new AMessage(kWhatTrackNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatTrackNotify, this);
notify->setSize("trackIndex", trackIndex);
sp<Track> track = new Track(notify, format);
@@ -833,7 +833,7 @@
int64_t whenUs = sampleTimeUs - mFirstSampleTimeUs + mFirstSampleTimeRealUs;
- sp<AMessage> msg = new AMessage(kWhatPullExtractorSample, id());
+ sp<AMessage> msg = new AMessage(kWhatPullExtractorSample, this);
msg->setInt32("generation", mPullExtractorGeneration);
msg->post(whenUs - nowUs);
@@ -857,7 +857,7 @@
size_t trackIndex;
CHECK_EQ((status_t)OK, mExtractor->getSampleTrackIndex(&trackIndex));
- sp<AMessage> msg = new AMessage(kWhatConverterNotify, id());
+ sp<AMessage> msg = new AMessage(kWhatConverterNotify, this);
msg->setSize(
"trackIndex", mExtractorTrackToInternalTrack.valueFor(trackIndex));
@@ -955,7 +955,7 @@
? MEDIA_MIMETYPE_AUDIO_RAW : MEDIA_MIMETYPE_AUDIO_AAC);
}
- notify = new AMessage(kWhatConverterNotify, id());
+ notify = new AMessage(kWhatConverterNotify, this);
notify->setSize("trackIndex", trackIndex);
sp<Converter> converter = new Converter(notify, codecLooper, format);
@@ -970,7 +970,7 @@
return err;
}
- notify = new AMessage(Converter::kWhatMediaPullerNotify, converter->id());
+ notify = new AMessage(Converter::kWhatMediaPullerNotify, converter);
notify->setSize("trackIndex", trackIndex);
sp<MediaPuller> puller = new MediaPuller(source, notify);
@@ -980,7 +980,7 @@
*numInputBuffers = converter->getInputBufferCount();
}
- notify = new AMessage(kWhatTrackNotify, id());
+ notify = new AMessage(kWhatTrackNotify, this);
notify->setSize("trackIndex", trackIndex);
sp<Track> track = new Track(
diff --git a/media/libstagefright/wifi-display/source/RepeaterSource.cpp b/media/libstagefright/wifi-display/source/RepeaterSource.cpp
index 59d7e6e..af6b663 100644
--- a/media/libstagefright/wifi-display/source/RepeaterSource.cpp
+++ b/media/libstagefright/wifi-display/source/RepeaterSource.cpp
@@ -173,7 +173,7 @@
}
void RepeaterSource::postRead() {
- (new AMessage(kWhatRead, mReflector->id()))->post();
+ (new AMessage(kWhatRead, mReflector))->post();
}
void RepeaterSource::onMessageReceived(const sp<AMessage> &msg) {
diff --git a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
index 7eb8b73..14d0951 100644
--- a/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
+++ b/media/libstagefright/wifi-display/source/WifiDisplaySource.cpp
@@ -57,7 +57,7 @@
mNetSession(netSession),
mClient(client),
mSessionID(0),
- mStopReplyID(0),
+ mStopReplyID(NULL),
mChosenRTPPort(-1),
mUsingPCMAudio(false),
mClientSessionID(0),
@@ -106,7 +106,7 @@
status_t WifiDisplaySource::start(const char *iface) {
CHECK_EQ(mState, INITIALIZED);
- sp<AMessage> msg = new AMessage(kWhatStart, id());
+ sp<AMessage> msg = new AMessage(kWhatStart, this);
msg->setString("iface", iface);
sp<AMessage> response;
@@ -114,21 +114,21 @@
}
status_t WifiDisplaySource::stop() {
- sp<AMessage> msg = new AMessage(kWhatStop, id());
+ sp<AMessage> msg = new AMessage(kWhatStop, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
}
status_t WifiDisplaySource::pause() {
- sp<AMessage> msg = new AMessage(kWhatPause, id());
+ sp<AMessage> msg = new AMessage(kWhatPause, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
}
status_t WifiDisplaySource::resume() {
- sp<AMessage> msg = new AMessage(kWhatResume, id());
+ sp<AMessage> msg = new AMessage(kWhatResume, this);
sp<AMessage> response;
return PostAndAwaitResponse(msg, &response);
@@ -138,7 +138,7 @@
switch (msg->what()) {
case kWhatStart:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
AString iface;
@@ -167,7 +167,7 @@
if (err == OK) {
if (inet_aton(iface.c_str(), &mInterfaceAddr) != 0) {
- sp<AMessage> notify = new AMessage(kWhatRTSPNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatRTSPNotify, this);
err = mNetSession->createRTSPServer(
mInterfaceAddr, port, notify, &mSessionID);
@@ -310,7 +310,7 @@
if (err == OK) {
mState = AWAITING_CLIENT_TEARDOWN;
- (new AMessage(kWhatTeardownTriggerTimedOut, id()))->post(
+ (new AMessage(kWhatTeardownTriggerTimedOut, this))->post(
kTeardownTriggerTimeouSecs * 1000000ll);
break;
@@ -325,7 +325,7 @@
case kWhatPause:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
status_t err = OK;
@@ -345,7 +345,7 @@
case kWhatResume:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
CHECK(msg->senderAwaitsResponse(&replyID));
status_t err = OK;
@@ -492,7 +492,7 @@
if (mState == AWAITING_CLIENT_TEARDOWN) {
ALOGI("TEARDOWN trigger timed out, forcing disconnection.");
- CHECK_NE(mStopReplyID, 0);
+ CHECK(mStopReplyID != NULL);
finishStop();
break;
}
@@ -529,7 +529,7 @@
// HDCPObserver::notify is completely handled before
// we clear the HDCP instance and unload the shared
// library :(
- (new AMessage(kWhatFinishStop2, id()))->post(300000ll);
+ (new AMessage(kWhatFinishStop2, this))->post(300000ll);
break;
}
@@ -1027,7 +1027,7 @@
}
mReaperPending = true;
- (new AMessage(kWhatReapDeadClients, id()))->post(kReaperIntervalUs);
+ (new AMessage(kWhatReapDeadClients, this))->post(kReaperIntervalUs);
}
void WifiDisplaySource::scheduleKeepAlive(int32_t sessionID) {
@@ -1035,7 +1035,7 @@
// expire, make sure the timeout is greater than 5 secs to begin with.
CHECK_GT(kPlaybackSessionTimeoutUs, 5000000ll);
- sp<AMessage> msg = new AMessage(kWhatKeepAlive, id());
+ sp<AMessage> msg = new AMessage(kWhatKeepAlive, this);
msg->setInt32("sessionID", sessionID);
msg->post(kPlaybackSessionTimeoutUs - 5000000ll);
}
@@ -1239,7 +1239,7 @@
int32_t playbackSessionID = makeUniquePlaybackSessionID();
- sp<AMessage> notify = new AMessage(kWhatPlaybackSessionNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatPlaybackSessionNotify, this);
notify->setInt32("playbackSessionID", playbackSessionID);
notify->setInt32("sessionID", sessionID);
@@ -1470,7 +1470,7 @@
mNetSession->sendRequest(sessionID, response.c_str());
if (mState == AWAITING_CLIENT_TEARDOWN) {
- CHECK_NE(mStopReplyID, 0);
+ CHECK(mStopReplyID != NULL);
finishStop();
} else {
mClient->onDisplayError(IRemoteDisplayClient::kDisplayErrorUnknown);
@@ -1707,7 +1707,7 @@
return ERROR_UNSUPPORTED;
}
- sp<AMessage> notify = new AMessage(kWhatHDCPNotify, id());
+ sp<AMessage> notify = new AMessage(kWhatHDCPNotify, this);
mHDCPObserver = new HDCPObserver(notify);
status_t err = mHDCP->setObserver(mHDCPObserver);
diff --git a/media/libstagefright/wifi-display/source/WifiDisplaySource.h b/media/libstagefright/wifi-display/source/WifiDisplaySource.h
index 750265f..0f779e4 100644
--- a/media/libstagefright/wifi-display/source/WifiDisplaySource.h
+++ b/media/libstagefright/wifi-display/source/WifiDisplaySource.h
@@ -27,6 +27,7 @@
namespace android {
+struct AReplyToken;
struct IHDCP;
struct IRemoteDisplayClient;
struct ParsedMessage;
@@ -121,7 +122,7 @@
struct in_addr mInterfaceAddr;
int32_t mSessionID;
- uint32_t mStopReplyID;
+ sp<AReplyToken> mStopReplyID;
AString mWfdClientRtpPorts;
int32_t mChosenRTPPort; // extracted from "wfd_client_rtp_ports"
diff --git a/media/libstagefright/yuv/YUVImage.cpp b/media/libstagefright/yuv/YUVImage.cpp
index bb3e2fd..c098135 100644
--- a/media/libstagefright/yuv/YUVImage.cpp
+++ b/media/libstagefright/yuv/YUVImage.cpp
@@ -374,13 +374,13 @@
void YUVImage::yuv2rgb(uint8_t yValue, uint8_t uValue, uint8_t vValue,
uint8_t *r, uint8_t *g, uint8_t *b) const {
- *r = yValue + (1.370705 * (vValue-128));
- *g = yValue - (0.698001 * (vValue-128)) - (0.337633 * (uValue-128));
- *b = yValue + (1.732446 * (uValue-128));
+ int rTmp = yValue + (1.370705 * (vValue-128));
+ int gTmp = yValue - (0.698001 * (vValue-128)) - (0.337633 * (uValue-128));
+ int bTmp = yValue + (1.732446 * (uValue-128));
- *r = clamp(*r, 0, 255);
- *g = clamp(*g, 0, 255);
- *b = clamp(*b, 0, 255);
+ *r = clamp(rTmp, 0, 255);
+ *g = clamp(gTmp, 0, 255);
+ *b = clamp(bTmp, 0, 255);
}
bool YUVImage::writeToPPM(const char *filename) const {
diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk
index 3a280f0..0e2e48c 100644
--- a/media/mediaserver/Android.mk
+++ b/media/mediaserver/Android.mk
@@ -11,7 +11,7 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- main_mediaserver.cpp
+ main_mediaserver.cpp
LOCAL_SHARED_LIBRARIES := \
libaudioflinger \
@@ -19,6 +19,7 @@
libcamera_metadata\
libcameraservice \
libmedialogservice \
+ libresourcemanagerservice \
libcutils \
libnbaio \
libmedia \
@@ -26,19 +27,25 @@
libutils \
liblog \
libbinder \
- libsoundtriggerservice
+ libsoundtriggerservice \
+ libradioservice
LOCAL_STATIC_LIBRARIES := \
- libregistermsext
+ libregistermsext
LOCAL_C_INCLUDES := \
frameworks/av/media/libmediaplayerservice \
frameworks/av/services/medialog \
frameworks/av/services/audioflinger \
frameworks/av/services/audiopolicy \
+ frameworks/av/services/audiopolicy/common/managerdefinitions/include \
+ frameworks/av/services/audiopolicy/common/include \
+ frameworks/av/services/audiopolicy/engine/interface \
frameworks/av/services/camera/libcameraservice \
+ frameworks/av/services/mediaresourcemanager \
$(call include-path-for, audio-utils) \
- frameworks/av/services/soundtrigger
+ frameworks/av/services/soundtrigger \
+ frameworks/av/services/radio
LOCAL_MODULE:= mediaserver
LOCAL_32_BIT_ONLY := true
diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index af1c9e6..99572f8 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -33,8 +33,9 @@
#include "CameraService.h"
#include "MediaLogService.h"
#include "MediaPlayerService.h"
-#include "AudioPolicyService.h"
+#include "service/AudioPolicyService.h"
#include "SoundTriggerHwService.h"
+#include "RadioService.h"
using namespace android;
@@ -130,6 +131,7 @@
CameraService::instantiate();
AudioPolicyService::instantiate();
SoundTriggerHwService::instantiate();
+ RadioService::instantiate();
registerExtensions();
ProcessState::self()->startThreadPool();
IPCThreadState::self()->joinThreadPool();
diff --git a/media/ndk/NdkMediaCodec.cpp b/media/ndk/NdkMediaCodec.cpp
index ed00b72..80c1c2f 100644
--- a/media/ndk/NdkMediaCodec.cpp
+++ b/media/ndk/NdkMediaCodec.cpp
@@ -116,7 +116,7 @@
case kWhatStopActivityNotifications:
{
- uint32_t replyID;
+ sp<AReplyToken> replyID;
msg->senderAwaitsResponse(&replyID);
mCodec->mGeneration++;
@@ -136,7 +136,7 @@
static void requestActivityNotification(AMediaCodec *codec) {
- (new AMessage(kWhatRequestActivityNotifications, codec->mHandler->id()))->post();
+ (new AMessage(kWhatRequestActivityNotifications, codec->mHandler))->post();
}
extern "C" {
@@ -219,7 +219,7 @@
if (ret != OK) {
return translate_error(ret);
}
- mData->mActivityNotification = new AMessage(kWhatActivityNotify, mData->mHandler->id());
+ mData->mActivityNotification = new AMessage(kWhatActivityNotify, mData->mHandler);
mData->mActivityNotification->setInt32("generation", mData->mGeneration);
requestActivityNotification(mData);
return AMEDIA_OK;
@@ -229,7 +229,7 @@
media_status_t AMediaCodec_stop(AMediaCodec *mData) {
media_status_t ret = translate_error(mData->mCodec->stop());
- sp<AMessage> msg = new AMessage(kWhatStopActivityNotifications, mData->mHandler->id());
+ sp<AMessage> msg = new AMessage(kWhatStopActivityNotifications, mData->mHandler);
sp<AMessage> response;
msg->postAndAwaitResponse(&response);
mData->mActivityNotification.clear();
@@ -352,7 +352,8 @@
}
//EXPORT
-media_status_t AMediaCodec_setNotificationCallback(AMediaCodec *mData, OnCodecEvent callback, void *userdata) {
+media_status_t AMediaCodec_setNotificationCallback(AMediaCodec *mData, OnCodecEvent callback,
+ void *userdata) {
mData->mCallback = callback;
mData->mCallbackUserData = userdata;
return AMEDIA_OK;
diff --git a/media/ndk/NdkMediaDrm.cpp b/media/ndk/NdkMediaDrm.cpp
index 7a1048c..83a5ba1 100644
--- a/media/ndk/NdkMediaDrm.cpp
+++ b/media/ndk/NdkMediaDrm.cpp
@@ -312,8 +312,10 @@
String8(optionalParameters[i].mValue));
}
String8 defaultUrl;
+ DrmPlugin::KeyRequestType keyRequestType;
status_t status = mObj->mDrm->getKeyRequest(*iter, mdInit, String8(mimeType),
- mdKeyType, mdOptionalParameters, mObj->mKeyRequest, defaultUrl);
+ mdKeyType, mdOptionalParameters, mObj->mKeyRequest, defaultUrl,
+ &keyRequestType);
if (status != OK) {
return translateStatus(status);
} else {
@@ -725,4 +727,3 @@
}
} // extern "C"
-
diff --git a/media/ndk/NdkMediaExtractor.cpp b/media/ndk/NdkMediaExtractor.cpp
index db57d0b..0ecd64f 100644
--- a/media/ndk/NdkMediaExtractor.cpp
+++ b/media/ndk/NdkMediaExtractor.cpp
@@ -70,7 +70,8 @@
}
EXPORT
-media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor *mData, int fd, off64_t offset, off64_t length) {
+media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor *mData, int fd, off64_t offset,
+ off64_t length) {
ALOGV("setDataSource(%d, %lld, %lld)", fd, offset, length);
return translate_error(mData->mImpl->setDataSource(fd, offset, length));
}
diff --git a/radio/Android.mk b/radio/Android.mk
new file mode 100644
index 0000000..ecbb8fd
--- /dev/null
+++ b/radio/Android.mk
@@ -0,0 +1,39 @@
+# Copyright 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.
+
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ Radio.cpp \
+ IRadio.cpp \
+ IRadioClient.cpp \
+ IRadioService.cpp
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libutils \
+ liblog \
+ libbinder \
+ libhardware \
+ libradio_metadata
+
+#LOCAL_C_INCLUDES += \
+ system/media/camera/include \
+ system/media/private/camera/include
+
+LOCAL_MODULE:= libradio
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/radio/IRadio.cpp b/radio/IRadio.cpp
new file mode 100644
index 0000000..242df77
--- /dev/null
+++ b/radio/IRadio.cpp
@@ -0,0 +1,344 @@
+/*
+**
+** Copyright 2015, 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 "IRadio"
+#include <utils/Log.h>
+#include <utils/Errors.h>
+#include <binder/IMemory.h>
+#include <radio/IRadio.h>
+#include <radio/IRadioService.h>
+#include <radio/IRadioClient.h>
+#include <system/radio.h>
+#include <system/radio_metadata.h>
+
+namespace android {
+
+enum {
+ DETACH = IBinder::FIRST_CALL_TRANSACTION,
+ SET_CONFIGURATION,
+ GET_CONFIGURATION,
+ SET_MUTE,
+ GET_MUTE,
+ SCAN,
+ STEP,
+ TUNE,
+ CANCEL,
+ GET_PROGRAM_INFORMATION,
+ HAS_CONTROL
+};
+
+class BpRadio: public BpInterface<IRadio>
+{
+public:
+ BpRadio(const sp<IBinder>& impl)
+ : BpInterface<IRadio>(impl)
+ {
+ }
+
+ void detach()
+ {
+ ALOGV("detach");
+ Parcel data, reply;
+ data.writeInterfaceToken(IRadio::getInterfaceDescriptor());
+ remote()->transact(DETACH, data, &reply);
+ }
+
+ virtual status_t setConfiguration(const struct radio_band_config *config)
+ {
+ Parcel data, reply;
+ if (config == NULL) {
+ return BAD_VALUE;
+ }
+ data.writeInterfaceToken(IRadio::getInterfaceDescriptor());
+ data.write(config, sizeof(struct radio_band_config));
+ status_t status = remote()->transact(SET_CONFIGURATION, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ }
+ return status;
+ }
+
+ virtual status_t getConfiguration(struct radio_band_config *config)
+ {
+ Parcel data, reply;
+ if (config == NULL) {
+ return BAD_VALUE;
+ }
+ data.writeInterfaceToken(IRadio::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_CONFIGURATION, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ if (status == NO_ERROR) {
+ reply.read(config, sizeof(struct radio_band_config));
+ }
+ }
+ return status;
+ }
+
+ virtual status_t setMute(bool mute)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IRadio::getInterfaceDescriptor());
+ data.writeInt32(mute ? 1 : 0);
+ status_t status = remote()->transact(SET_MUTE, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ }
+ return status;
+ }
+
+ virtual status_t getMute(bool *mute)
+ {
+ Parcel data, reply;
+ if (mute == NULL) {
+ return BAD_VALUE;
+ }
+ data.writeInterfaceToken(IRadio::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_MUTE, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ if (status == NO_ERROR) {
+ int muteread = reply.readInt32();
+ *mute = muteread != 0;
+ }
+ }
+ return status;
+ }
+
+ virtual status_t scan(radio_direction_t direction, bool skipSubChannel)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IRadio::getInterfaceDescriptor());
+ data.writeInt32(direction);
+ data.writeInt32(skipSubChannel ? 1 : 0);
+ status_t status = remote()->transact(SCAN, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ }
+ return status;
+ }
+
+ virtual status_t step(radio_direction_t direction, bool skipSubChannel)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IRadio::getInterfaceDescriptor());
+ data.writeInt32(direction);
+ data.writeInt32(skipSubChannel ? 1 : 0);
+ status_t status = remote()->transact(STEP, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ }
+ return status;
+ }
+
+ virtual status_t tune(unsigned int channel, unsigned int subChannel)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IRadio::getInterfaceDescriptor());
+ data.writeInt32(channel);
+ data.writeInt32(subChannel);
+ status_t status = remote()->transact(TUNE, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ }
+ return status;
+ }
+
+ virtual status_t cancel()
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IRadio::getInterfaceDescriptor());
+ status_t status = remote()->transact(CANCEL, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ }
+ return status;
+ }
+
+ virtual status_t getProgramInformation(struct radio_program_info *info)
+ {
+ Parcel data, reply;
+ if (info == NULL) {
+ return BAD_VALUE;
+ }
+ radio_metadata_t *metadata = info->metadata;
+ data.writeInterfaceToken(IRadio::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_PROGRAM_INFORMATION, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ if (status == NO_ERROR) {
+ reply.read(info, sizeof(struct radio_program_info));
+ info->metadata = metadata;
+ if (metadata == NULL) {
+ return status;
+ }
+ size_t size = (size_t)reply.readInt32();
+ if (size == 0) {
+ return status;
+ }
+ metadata =
+ (radio_metadata_t *)calloc(size / sizeof(unsigned int), sizeof(unsigned int));
+ if (metadata == NULL) {
+ return NO_MEMORY;
+ }
+ reply.read(metadata, size);
+ status = radio_metadata_add_metadata(&info->metadata, metadata);
+ free(metadata);
+ }
+ }
+ return status;
+ }
+
+ virtual status_t hasControl(bool *hasControl)
+ {
+ Parcel data, reply;
+ if (hasControl == NULL) {
+ return BAD_VALUE;
+ }
+ data.writeInterfaceToken(IRadio::getInterfaceDescriptor());
+ status_t status = remote()->transact(HAS_CONTROL, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ if (status == NO_ERROR) {
+ *hasControl = reply.readInt32() != 0;
+ }
+ }
+ return status;
+ }
+};
+
+IMPLEMENT_META_INTERFACE(Radio, "android.hardware.IRadio");
+
+// ----------------------------------------------------------------------
+
+status_t BnRadio::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ switch(code) {
+ case DETACH: {
+ ALOGV("DETACH");
+ CHECK_INTERFACE(IRadio, data, reply);
+ detach();
+ return NO_ERROR;
+ } break;
+ case SET_CONFIGURATION: {
+ CHECK_INTERFACE(IRadio, data, reply);
+ struct radio_band_config config;
+ data.read(&config, sizeof(struct radio_band_config));
+ status_t status = setConfiguration(&config);
+ reply->writeInt32(status);
+ return NO_ERROR;
+ }
+ case GET_CONFIGURATION: {
+ CHECK_INTERFACE(IRadio, data, reply);
+ struct radio_band_config config;
+ status_t status = getConfiguration(&config);
+ reply->writeInt32(status);
+ if (status == NO_ERROR) {
+ reply->write(&config, sizeof(struct radio_band_config));
+ }
+ return NO_ERROR;
+ }
+ case SET_MUTE: {
+ CHECK_INTERFACE(IRadio, data, reply);
+ bool mute = data.readInt32() != 0;
+ status_t status = setMute(mute);
+ reply->writeInt32(status);
+ return NO_ERROR;
+ }
+ case GET_MUTE: {
+ CHECK_INTERFACE(IRadio, data, reply);
+ bool mute;
+ status_t status = getMute(&mute);
+ reply->writeInt32(status);
+ if (status == NO_ERROR) {
+ reply->writeInt32(mute ? 1 : 0);
+ }
+ return NO_ERROR;
+ }
+ case SCAN: {
+ CHECK_INTERFACE(IRadio, data, reply);
+ radio_direction_t direction = (radio_direction_t)data.readInt32();
+ bool skipSubChannel = data.readInt32() == 1;
+ status_t status = scan(direction, skipSubChannel);
+ reply->writeInt32(status);
+ return NO_ERROR;
+ }
+ case STEP: {
+ CHECK_INTERFACE(IRadio, data, reply);
+ radio_direction_t direction = (radio_direction_t)data.readInt32();
+ bool skipSubChannel = data.readInt32() == 1;
+ status_t status = step(direction, skipSubChannel);
+ reply->writeInt32(status);
+ return NO_ERROR;
+ }
+ case TUNE: {
+ CHECK_INTERFACE(IRadio, data, reply);
+ unsigned int channel = (unsigned int)data.readInt32();
+ unsigned int subChannel = (unsigned int)data.readInt32();
+ status_t status = tune(channel, subChannel);
+ reply->writeInt32(status);
+ return NO_ERROR;
+ }
+ case CANCEL: {
+ CHECK_INTERFACE(IRadio, data, reply);
+ status_t status = cancel();
+ reply->writeInt32(status);
+ return NO_ERROR;
+ }
+ case GET_PROGRAM_INFORMATION: {
+ CHECK_INTERFACE(IRadio, data, reply);
+ struct radio_program_info info;
+
+ status_t status = radio_metadata_allocate(&info.metadata, 0, 0);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ status = getProgramInformation(&info);
+ reply->writeInt32(status);
+ if (status == NO_ERROR) {
+ reply->write(&info, sizeof(struct radio_program_info));
+ int count = radio_metadata_get_count(info.metadata);
+ if (count > 0) {
+ size_t size = radio_metadata_get_size(info.metadata);
+ reply->writeInt32(size);
+ reply->write(info.metadata, size);
+ } else {
+ reply->writeInt32(0);
+ }
+ }
+ radio_metadata_deallocate(info.metadata);
+ return NO_ERROR;
+ }
+ case HAS_CONTROL: {
+ CHECK_INTERFACE(IRadio, data, reply);
+ bool control;
+ status_t status = hasControl(&control);
+ reply->writeInt32(status);
+ if (status == NO_ERROR) {
+ reply->writeInt32(control ? 1 : 0);
+ }
+ return NO_ERROR;
+ }
+ default:
+ return BBinder::onTransact(code, data, reply, flags);
+ }
+}
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
diff --git a/radio/IRadioClient.cpp b/radio/IRadioClient.cpp
new file mode 100644
index 0000000..033ca49
--- /dev/null
+++ b/radio/IRadioClient.cpp
@@ -0,0 +1,75 @@
+/*
+**
+** Copyright 2015, 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 <stdint.h>
+#include <sys/types.h>
+#include <binder/IMemory.h>
+#include <binder/Parcel.h>
+#include <binder/IPCThreadState.h>
+#include <binder/IServiceManager.h>
+#include <radio/IRadioClient.h>
+
+namespace android {
+
+enum {
+ ON_EVENT = IBinder::FIRST_CALL_TRANSACTION,
+};
+
+class BpRadioClient: public BpInterface<IRadioClient>
+{
+
+public:
+ BpRadioClient(const sp<IBinder>& impl)
+ : BpInterface<IRadioClient>(impl)
+ {
+ }
+
+ virtual void onEvent(const sp<IMemory>& eventMemory)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IRadioClient::getInterfaceDescriptor());
+ data.writeStrongBinder(IInterface::asBinder(eventMemory));
+ remote()->transact(ON_EVENT,
+ data,
+ &reply);
+ }
+};
+
+IMPLEMENT_META_INTERFACE(RadioClient,
+ "android.hardware.IRadioClient");
+
+// ----------------------------------------------------------------------
+
+status_t BnRadioClient::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ switch(code) {
+ case ON_EVENT: {
+ CHECK_INTERFACE(IRadioClient, data, reply);
+ sp<IMemory> eventMemory = interface_cast<IMemory>(
+ data.readStrongBinder());
+ onEvent(eventMemory);
+ return NO_ERROR;
+ } break;
+ default:
+ return BBinder::onTransact(code, data, reply, flags);
+ } return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
diff --git a/radio/IRadioService.cpp b/radio/IRadioService.cpp
new file mode 100644
index 0000000..8c2b3ef
--- /dev/null
+++ b/radio/IRadioService.cpp
@@ -0,0 +1,181 @@
+/*
+**
+** Copyright 2015, 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 "BpRadioService"
+//
+#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+#include <utils/Errors.h>
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <binder/IMemory.h>
+#include <binder/Parcel.h>
+#include <binder/IPCThreadState.h>
+#include <binder/IServiceManager.h>
+
+#include <radio/IRadioService.h>
+#include <radio/IRadio.h>
+#include <radio/IRadioClient.h>
+
+namespace android {
+
+enum {
+ LIST_MODULES = IBinder::FIRST_CALL_TRANSACTION,
+ ATTACH,
+};
+
+#define MAX_ITEMS_PER_LIST 1024
+
+class BpRadioService: public BpInterface<IRadioService>
+{
+public:
+ BpRadioService(const sp<IBinder>& impl)
+ : BpInterface<IRadioService>(impl)
+ {
+ }
+
+ virtual status_t listModules(struct radio_properties *properties,
+ uint32_t *numModules)
+ {
+ if (numModules == NULL || (*numModules != 0 && properties == NULL)) {
+ return BAD_VALUE;
+ }
+ Parcel data, reply;
+ data.writeInterfaceToken(IRadioService::getInterfaceDescriptor());
+ unsigned int numModulesReq = (properties == NULL) ? 0 : *numModules;
+ data.writeInt32(numModulesReq);
+ status_t status = remote()->transact(LIST_MODULES, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ *numModules = (unsigned int)reply.readInt32();
+ }
+ ALOGV("listModules() status %d got *numModules %d", status, *numModules);
+ if (status == NO_ERROR) {
+ if (numModulesReq > *numModules) {
+ numModulesReq = *numModules;
+ }
+ if (numModulesReq > 0) {
+ reply.read(properties, numModulesReq * sizeof(struct radio_properties));
+ }
+ }
+ return status;
+ }
+
+ virtual status_t attach(radio_handle_t handle,
+ const sp<IRadioClient>& client,
+ const struct radio_band_config *config,
+ bool withAudio,
+ sp<IRadio>& radio)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IRadioService::getInterfaceDescriptor());
+ data.writeInt32(handle);
+ data.writeStrongBinder(IInterface::asBinder(client));
+ ALOGV("attach() config %p withAudio %d region %d type %d", config, withAudio, config->region, config->band.type);
+ if (config == NULL) {
+ data.writeInt32(0);
+ } else {
+ data.writeInt32(1);
+ data.write(config, sizeof(struct radio_band_config));
+ }
+ data.writeInt32(withAudio ? 1 : 0);
+ status_t status = remote()->transact(ATTACH, data, &reply);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ status = reply.readInt32();
+ if (reply.readInt32() != 0) {
+ radio = interface_cast<IRadio>(reply.readStrongBinder());
+ }
+ return status;
+ }
+};
+
+IMPLEMENT_META_INTERFACE(RadioService, "android.hardware.IRadioService");
+
+// ----------------------------------------------------------------------
+
+status_t BnRadioService::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ switch(code) {
+ case LIST_MODULES: {
+ CHECK_INTERFACE(IRadioService, data, reply);
+ unsigned int numModulesReq = data.readInt32();
+ if (numModulesReq > MAX_ITEMS_PER_LIST) {
+ numModulesReq = MAX_ITEMS_PER_LIST;
+ }
+ unsigned int numModules = numModulesReq;
+ struct radio_properties *properties =
+ (struct radio_properties *)calloc(numModulesReq,
+ sizeof(struct radio_properties));
+ if (properties == NULL) {
+ reply->writeInt32(NO_MEMORY);
+ reply->writeInt32(0);
+ return NO_ERROR;
+ }
+
+ status_t status = listModules(properties, &numModules);
+ reply->writeInt32(status);
+ reply->writeInt32(numModules);
+ ALOGV("LIST_MODULES status %d got numModules %d", status, numModules);
+
+ if (status == NO_ERROR) {
+ if (numModulesReq > numModules) {
+ numModulesReq = numModules;
+ }
+ reply->write(properties,
+ numModulesReq * sizeof(struct radio_properties));
+ }
+ free(properties);
+ return NO_ERROR;
+ } break;
+
+ case ATTACH: {
+ CHECK_INTERFACE(IRadioService, data, reply);
+ radio_handle_t handle = data.readInt32();
+ sp<IRadioClient> client =
+ interface_cast<IRadioClient>(data.readStrongBinder());
+ struct radio_band_config config;
+ struct radio_band_config *configPtr = NULL;
+ if (data.readInt32() != 0) {
+ data.read(&config, sizeof(struct radio_band_config));
+ configPtr = &config;
+ }
+ bool withAudio = data.readInt32() != 0;
+ ALOGV("ATTACH configPtr %p withAudio %d", configPtr, withAudio);
+ sp<IRadio> radio;
+ status_t status = attach(handle, client, configPtr, withAudio, radio);
+ reply->writeInt32(status);
+ if (radio != 0) {
+ reply->writeInt32(1);
+ reply->writeStrongBinder(IInterface::asBinder(radio));
+ } else {
+ reply->writeInt32(0);
+ }
+ return NO_ERROR;
+ } break;
+ default:
+ return BBinder::onTransact(code, data, reply, flags);
+ }
+}
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
diff --git a/radio/Radio.cpp b/radio/Radio.cpp
new file mode 100644
index 0000000..e3554c2
--- /dev/null
+++ b/radio/Radio.cpp
@@ -0,0 +1,283 @@
+/*
+**
+** Copyright (C) 2015, 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 "Radio"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+#include <utils/threads.h>
+#include <binder/IPCThreadState.h>
+#include <binder/IServiceManager.h>
+#include <binder/IMemory.h>
+
+#include <radio/Radio.h>
+#include <radio/IRadio.h>
+#include <radio/IRadioService.h>
+#include <radio/IRadioClient.h>
+#include <radio/RadioCallback.h>
+
+namespace android {
+
+namespace {
+ sp<IRadioService> gRadioService;
+ const int kRadioServicePollDelay = 500000; // 0.5s
+ const char* kRadioServiceName = "media.radio";
+ Mutex gLock;
+
+ class DeathNotifier : public IBinder::DeathRecipient
+ {
+ public:
+ DeathNotifier() {
+ }
+
+ virtual void binderDied(const wp<IBinder>& who __unused) {
+ ALOGV("binderDied");
+ Mutex::Autolock _l(gLock);
+ gRadioService.clear();
+ ALOGW("Radio service died!");
+ }
+ };
+
+ sp<DeathNotifier> gDeathNotifier;
+}; // namespace anonymous
+
+const sp<IRadioService>& Radio::getRadioService()
+{
+ Mutex::Autolock _l(gLock);
+ if (gRadioService.get() == 0) {
+ sp<IServiceManager> sm = defaultServiceManager();
+ sp<IBinder> binder;
+ do {
+ binder = sm->getService(String16(kRadioServiceName));
+ if (binder != 0) {
+ break;
+ }
+ ALOGW("RadioService not published, waiting...");
+ usleep(kRadioServicePollDelay);
+ } while(true);
+ if (gDeathNotifier == NULL) {
+ gDeathNotifier = new DeathNotifier();
+ }
+ binder->linkToDeath(gDeathNotifier);
+ gRadioService = interface_cast<IRadioService>(binder);
+ }
+ ALOGE_IF(gRadioService == 0, "no RadioService!?");
+ return gRadioService;
+}
+
+// Static methods
+status_t Radio::listModules(struct radio_properties *properties,
+ uint32_t *numModules)
+{
+ ALOGV("listModules()");
+ const sp<IRadioService>& service = getRadioService();
+ if (service == 0) {
+ return NO_INIT;
+ }
+ return service->listModules(properties, numModules);
+}
+
+sp<Radio> Radio::attach(radio_handle_t handle,
+ const struct radio_band_config *config,
+ bool withAudio,
+ const sp<RadioCallback>& callback)
+{
+ ALOGV("attach()");
+ sp<Radio> radio;
+ const sp<IRadioService>& service = getRadioService();
+ if (service == 0) {
+ return radio;
+ }
+ radio = new Radio(handle, callback);
+ status_t status = service->attach(handle, radio, config, withAudio, radio->mIRadio);
+
+ if (status == NO_ERROR && radio->mIRadio != 0) {
+ IInterface::asBinder(radio->mIRadio)->linkToDeath(radio);
+ } else {
+ ALOGW("Error %d connecting to radio service", status);
+ radio.clear();
+ }
+ return radio;
+}
+
+
+
+// Radio
+Radio::Radio(radio_handle_t handle, const sp<RadioCallback>& callback)
+ : mHandle(handle), mCallback(callback)
+{
+}
+
+Radio::~Radio()
+{
+ if (mIRadio != 0) {
+ mIRadio->detach();
+ }
+}
+
+
+void Radio::detach() {
+ ALOGV("detach()");
+ Mutex::Autolock _l(mLock);
+ mCallback.clear();
+ if (mIRadio != 0) {
+ mIRadio->detach();
+ IInterface::asBinder(mIRadio)->unlinkToDeath(this);
+ mIRadio = 0;
+ }
+}
+
+status_t Radio::setConfiguration(const struct radio_band_config *config)
+{
+ Mutex::Autolock _l(mLock);
+ if (mIRadio == 0) {
+ return NO_INIT;
+ }
+ return mIRadio->setConfiguration(config);
+}
+
+status_t Radio::getConfiguration(struct radio_band_config *config)
+{
+ Mutex::Autolock _l(mLock);
+ if (mIRadio == 0) {
+ return NO_INIT;
+ }
+ return mIRadio->getConfiguration(config);
+}
+
+status_t Radio::setMute(bool mute)
+{
+ Mutex::Autolock _l(mLock);
+ if (mIRadio == 0) {
+ return NO_INIT;
+ }
+ return mIRadio->setMute(mute);
+}
+
+status_t Radio::getMute(bool *mute)
+{
+ Mutex::Autolock _l(mLock);
+ if (mIRadio == 0) {
+ return NO_INIT;
+ }
+ return mIRadio->getMute(mute);
+}
+
+status_t Radio::scan(radio_direction_t direction, bool skipSubchannel)
+{
+ Mutex::Autolock _l(mLock);
+ if (mIRadio == 0) {
+ return NO_INIT;
+ }
+ return mIRadio->scan(direction, skipSubchannel);
+}
+
+status_t Radio::step(radio_direction_t direction, bool skipSubchannel)
+{
+ Mutex::Autolock _l(mLock);
+ if (mIRadio == 0) {
+ return NO_INIT;
+ }
+ return mIRadio->step(direction, skipSubchannel);
+}
+
+status_t Radio::tune(unsigned int channel, unsigned int subChannel)
+{
+ Mutex::Autolock _l(mLock);
+ if (mIRadio == 0) {
+ return NO_INIT;
+ }
+ return mIRadio->tune(channel, subChannel);
+}
+
+status_t Radio::cancel()
+{
+ Mutex::Autolock _l(mLock);
+ if (mIRadio == 0) {
+ return NO_INIT;
+ }
+ return mIRadio->cancel();
+}
+
+status_t Radio::getProgramInformation(struct radio_program_info *info)
+{
+ Mutex::Autolock _l(mLock);
+ if (mIRadio == 0) {
+ return NO_INIT;
+ }
+ return mIRadio->getProgramInformation(info);
+}
+
+status_t Radio::hasControl(bool *hasControl)
+{
+ Mutex::Autolock _l(mLock);
+ if (mIRadio == 0) {
+ return NO_INIT;
+ }
+ return mIRadio->hasControl(hasControl);
+}
+
+
+// BpRadioClient
+void Radio::onEvent(const sp<IMemory>& eventMemory)
+{
+ Mutex::Autolock _l(mLock);
+ if (eventMemory == 0 || eventMemory->pointer() == NULL) {
+ return;
+ }
+
+ struct radio_event *event = (struct radio_event *)eventMemory->pointer();
+ // restore local metadata pointer from offset
+ switch (event->type) {
+ case RADIO_EVENT_TUNED:
+ case RADIO_EVENT_AF_SWITCH:
+ if (event->info.metadata != NULL) {
+ event->info.metadata =
+ (radio_metadata_t *)((char *)event + (size_t)event->info.metadata);
+ }
+ break;
+ case RADIO_EVENT_METADATA:
+ if (event->metadata != NULL) {
+ event->metadata =
+ (radio_metadata_t *)((char *)event + (size_t)event->metadata);
+ }
+ break;
+ default:
+ break;
+ }
+
+ if (mCallback != 0) {
+ mCallback->onEvent(event);
+ }
+}
+
+
+//IBinder::DeathRecipient
+void Radio::binderDied(const wp<IBinder>& who __unused) {
+ Mutex::Autolock _l(mLock);
+ ALOGW("Radio server binder Died ");
+ mIRadio = 0;
+ struct radio_event event;
+ memset(&event, 0, sizeof(struct radio_event));
+ event.type = RADIO_EVENT_SERVER_DIED;
+ event.status = DEAD_OBJECT;
+ if (mCallback != 0) {
+ mCallback->onEvent(&event);
+ }
+}
+
+}; // namespace android
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 44d2553..fee2347 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -39,6 +39,9 @@
AudioFlinger.cpp \
Threads.cpp \
Tracks.cpp \
+ AudioHwDevice.cpp \
+ AudioStreamOut.cpp \
+ SpdifStreamOut.cpp \
Effects.cpp \
AudioMixer.cpp.arm \
PatchPanel.cpp
@@ -52,6 +55,7 @@
LOCAL_SHARED_LIBRARIES := \
libaudioresampler \
+ libaudiospdif \
libaudioutils \
libcommon_time_client \
libcutils \
@@ -74,9 +78,17 @@
LOCAL_MODULE:= libaudioflinger
LOCAL_32_BIT_ONLY := true
-LOCAL_SRC_FILES += FastMixer.cpp FastMixerState.cpp AudioWatchdog.cpp
-LOCAL_SRC_FILES += FastThread.cpp FastThreadState.cpp
-LOCAL_SRC_FILES += FastCapture.cpp FastCaptureState.cpp
+LOCAL_SRC_FILES += \
+ AudioWatchdog.cpp \
+ FastCapture.cpp \
+ FastCaptureDumpState.cpp \
+ FastCaptureState.cpp \
+ FastMixer.cpp \
+ FastMixerDumpState.cpp \
+ FastMixerState.cpp \
+ FastThread.cpp \
+ FastThreadDumpState.cpp \
+ FastThreadState.cpp
LOCAL_CFLAGS += -DSTATE_QUEUE_INSTANTIATIONS='"StateQueueInstantiations.cpp"'
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 993db73..f3206cb 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -185,7 +185,8 @@
char value[PROPERTY_VALUE_MAX];
bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
if (doLog) {
- mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY);
+ mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
+ MemoryHeapBase::READ_ONLY);
}
#ifdef TEE_SINK
@@ -271,7 +272,7 @@
};
#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
-AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
+AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
audio_module_handle_t module,
audio_devices_t devices)
{
@@ -401,6 +402,9 @@
String8 result(kClientLockedString);
write(fd, result.string(), result.size());
}
+
+ EffectDumpEffects(fd);
+
dumpClients(fd, args);
if (clientLocked) {
mClientLock.unlock();
@@ -822,14 +826,20 @@
if (ret != NO_ERROR) {
return false;
}
-
+ bool mute = true;
bool state = AUDIO_MODE_INVALID;
AutoMutex lock(mHardwareLock);
- audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
- dev->get_mic_mute(dev, &state);
+ for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
+ audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
+ status_t result = dev->get_mic_mute(dev, &state);
+ if (result == NO_ERROR) {
+ mute = mute && state;
+ }
+ }
mHardwareStatus = AUDIO_HW_IDLE;
- return state;
+
+ return mute;
}
status_t AudioFlinger::setMasterMute(bool muted)
@@ -1706,8 +1716,6 @@
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
- audio_stream_out_t *outStream = NULL;
-
// FOR TESTING ONLY:
// This if statement allows overriding the audio policy settings
// and forcing a specific format or channel mask to the HAL/Sink device for testing.
@@ -1729,25 +1737,18 @@
}
}
- status_t status = hwDevHal->open_output_stream(hwDevHal,
- *output,
- devices,
- flags,
- config,
- &outStream,
- address.string());
+ AudioStreamOut *outputStream = NULL;
+ status_t status = outHwDev->openOutputStream(
+ &outputStream,
+ *output,
+ devices,
+ flags,
+ config,
+ address.string());
mHardwareStatus = AUDIO_HW_IDLE;
- ALOGV("openOutput_l() openOutputStream returned output %p, sampleRate %d, Format %#x, "
- "channelMask %#x, status %d",
- outStream,
- config->sample_rate,
- config->format,
- config->channel_mask,
- status);
- if (status == NO_ERROR && outStream != NULL) {
- AudioStreamOut *outputStream = new AudioStreamOut(outHwDev, outStream, flags);
+ if (status == NO_ERROR) {
PlaybackThread *thread;
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
@@ -1777,7 +1778,7 @@
uint32_t *latencyMs,
audio_output_flags_t flags)
{
- ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
+ ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
module,
(devices != NULL) ? *devices : 0,
config->sample_rate,
@@ -1947,18 +1948,18 @@
status_t AudioFlinger::openInput(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
- audio_devices_t *device,
+ audio_devices_t *devices,
const String8& address,
audio_source_t source,
audio_input_flags_t flags)
{
Mutex::Autolock _l(mLock);
- if (*device == AUDIO_DEVICE_NONE) {
+ if (*devices == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
- sp<RecordThread> thread = openInput_l(module, input, config, *device, address, source, flags);
+ sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags);
if (thread != 0) {
// notify client processes of the new input creation
@@ -1971,12 +1972,12 @@
sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
- audio_devices_t device,
+ audio_devices_t devices,
const String8& address,
audio_source_t source,
audio_input_flags_t flags)
{
- AudioHwDevice *inHwDev = findSuitableHwDev_l(module, device);
+ AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices);
if (inHwDev == NULL) {
*input = AUDIO_IO_HANDLE_NONE;
return 0;
@@ -1989,7 +1990,7 @@
audio_config_t halconfig = *config;
audio_hw_device_t *inHwHal = inHwDev->hwDevice();
audio_stream_in_t *inStream = NULL;
- status_t status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig,
+ status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
&inStream, flags, address.string(), source);
ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d"
", Format %#x, Channels %x, flags %#x, status %d addr %s",
@@ -2011,7 +2012,7 @@
// FIXME describe the change proposed by HAL (save old values so we can log them here)
ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
inStream = NULL;
- status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig,
+ status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
&inStream, flags, address.string(), source);
// FIXME log this new status; HAL should not propose any further changes
}
@@ -2076,7 +2077,7 @@
inputStream,
*input,
primaryOutputDevice_l(),
- device
+ devices
#ifdef TEE_SINK
, teeSink
#endif
@@ -2799,13 +2800,13 @@
struct Entry {
-#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav
- char mName[MAX_NAME];
+#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21
+ char mFileName[TEE_MAX_FILENAME];
};
int comparEntry(const void *p1, const void *p2)
{
- return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
+ return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName);
}
#ifdef TEE_SINK
@@ -2824,11 +2825,11 @@
DIR *dir = opendir(teePath);
teePath[teePathLen++] = '/';
if (dir != NULL) {
-#define MAX_SORT 20 // number of entries to sort
-#define MAX_KEEP 10 // number of entries to keep
- struct Entry entries[MAX_SORT];
+#define TEE_MAX_SORT 20 // number of entries to sort
+#define TEE_MAX_KEEP 10 // number of entries to keep
+ struct Entry entries[TEE_MAX_SORT];
size_t entryCount = 0;
- while (entryCount < MAX_SORT) {
+ while (entryCount < TEE_MAX_SORT) {
struct dirent de;
struct dirent *result = NULL;
int rc = readdir_r(dir, &de, &result);
@@ -2845,17 +2846,17 @@
}
// ignore non .wav file entries
size_t nameLen = strlen(de.d_name);
- if (nameLen <= 4 || nameLen >= MAX_NAME ||
+ if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME ||
strcmp(&de.d_name[nameLen - 4], ".wav")) {
continue;
}
- strcpy(entries[entryCount++].mName, de.d_name);
+ strcpy(entries[entryCount++].mFileName, de.d_name);
}
(void) closedir(dir);
- if (entryCount > MAX_KEEP) {
+ if (entryCount > TEE_MAX_KEEP) {
qsort(entries, entryCount, sizeof(Entry), comparEntry);
- for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
- strcpy(&teePath[teePathLen], entries[i].mName);
+ for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) {
+ strcpy(&teePath[teePathLen], entries[i].mFileName);
(void) unlink(teePath);
}
}
@@ -2939,4 +2940,4 @@
return BnAudioFlinger::onTransact(code, data, reply, flags);
}
-}; // namespace android
+} // namespace android
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index aa0af1f..c7d9161 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -56,6 +56,9 @@
#include <media/nbaio/NBAIO.h>
#include "AudioWatchdog.h"
#include "AudioMixer.h"
+#include "AudioStreamOut.h"
+#include "SpdifStreamOut.h"
+#include "AudioHwDevice.h"
#include <powermanager/IPowerManager.h>
@@ -311,7 +314,6 @@
wp<RefBase> cookie);
private:
- class AudioHwDevice; // fwd declaration for findSuitableHwDev_l
audio_mode_t getMode() const { return mMode; }
@@ -449,7 +451,7 @@
class EffectModule;
class EffectHandle;
class EffectChain;
- struct AudioStreamOut;
+
struct AudioStreamIn;
struct stream_type_t {
@@ -586,57 +588,11 @@
// Return true if the effect was found in mOrphanEffectChains, false otherwise.
bool updateOrphanEffectChains(const sp<EffectModule>& effect);
- class AudioHwDevice {
- public:
- enum Flags {
- AHWD_CAN_SET_MASTER_VOLUME = 0x1,
- AHWD_CAN_SET_MASTER_MUTE = 0x2,
- };
- AudioHwDevice(audio_module_handle_t handle,
- const char *moduleName,
- audio_hw_device_t *hwDevice,
- Flags flags)
- : mHandle(handle), mModuleName(strdup(moduleName))
- , mHwDevice(hwDevice)
- , mFlags(flags) { }
- /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); }
-
- bool canSetMasterVolume() const {
- return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME));
- }
-
- bool canSetMasterMute() const {
- return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE));
- }
-
- audio_module_handle_t handle() const { return mHandle; }
- const char *moduleName() const { return mModuleName; }
- audio_hw_device_t *hwDevice() const { return mHwDevice; }
- uint32_t version() const { return mHwDevice->common.version; }
-
- private:
- const audio_module_handle_t mHandle;
- const char * const mModuleName;
- audio_hw_device_t * const mHwDevice;
- const Flags mFlags;
- };
-
- // AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
+ // AudioStreamIn is immutable, so their fields are const.
// For emphasis, we could also make all pointers to them be "const *",
// but that would clutter the code unnecessarily.
- struct AudioStreamOut {
- AudioHwDevice* const audioHwDev;
- audio_stream_out_t* const stream;
- const audio_output_flags_t flags;
-
- audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
-
- AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) :
- audioHwDev(dev), stream(out), flags(flags) {}
- };
-
struct AudioStreamIn {
AudioHwDevice* const audioHwDev;
audio_stream_in_t* const stream;
@@ -796,9 +752,13 @@
#undef INCLUDING_FROM_AUDIOFLINGER_H
const char *formatToString(audio_format_t format);
+String8 inputFlagsToString(audio_input_flags_t flags);
+String8 outputFlagsToString(audio_output_flags_t flags);
+String8 devicesToString(audio_devices_t devices);
+const char *sourceToString(audio_source_t source);
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
#endif // ANDROID_AUDIO_FLINGER_H
diff --git a/services/audioflinger/AudioHwDevice.cpp b/services/audioflinger/AudioHwDevice.cpp
new file mode 100644
index 0000000..09d86ea
--- /dev/null
+++ b/services/audioflinger/AudioHwDevice.cpp
@@ -0,0 +1,94 @@
+/*
+**
+** 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 "AudioHwDevice"
+//#define LOG_NDEBUG 0
+
+#include <hardware/audio.h>
+#include <utils/Log.h>
+
+#include <audio_utils/spdif/SPDIFEncoder.h>
+
+#include "AudioHwDevice.h"
+#include "AudioStreamOut.h"
+#include "SpdifStreamOut.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+status_t AudioHwDevice::openOutputStream(
+ AudioStreamOut **ppStreamOut,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ const char *address)
+{
+
+ struct audio_config originalConfig = *config;
+ AudioStreamOut *outputStream = new AudioStreamOut(this, flags);
+
+ // Try to open the HAL first using the current format.
+ ALOGV("AudioHwDevice::openOutputStream(), try "
+ " sampleRate %d, Format %#x, "
+ "channelMask %#x",
+ config->sample_rate,
+ config->format,
+ config->channel_mask);
+ status_t status = outputStream->open(handle, devices, config, address);
+
+ if (status != NO_ERROR) {
+ delete outputStream;
+ outputStream = NULL;
+
+ // FIXME Look at any modification to the config.
+ // The HAL might modify the config to suggest a wrapped format.
+ // Log this so we can see what the HALs are doing.
+ ALOGI("AudioHwDevice::openOutputStream(), HAL returned"
+ " sampleRate %d, Format %#x, "
+ "channelMask %#x, status %d",
+ config->sample_rate,
+ config->format,
+ config->channel_mask,
+ status);
+
+ // If the data is encoded then try again using wrapped PCM.
+ bool wrapperNeeded = !audio_is_linear_pcm(originalConfig.format)
+ && ((flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)
+ && ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0);
+
+ // FIXME - Add isEncodingSupported() query to SPDIF wrapper then
+ // call it from here.
+ if (wrapperNeeded) {
+ outputStream = new SpdifStreamOut(this, flags);
+ status = outputStream->open(handle, devices, &originalConfig, address);
+ if (status != NO_ERROR) {
+ ALOGE("ERROR - AudioHwDevice::openOutputStream(), SPDIF open returned %d",
+ status);
+ delete outputStream;
+ outputStream = NULL;
+ }
+ }
+ }
+
+ *ppStreamOut = outputStream;
+ return status;
+}
+
+
+}; // namespace android
diff --git a/services/audioflinger/AudioHwDevice.h b/services/audioflinger/AudioHwDevice.h
new file mode 100644
index 0000000..b9f65c1
--- /dev/null
+++ b/services/audioflinger/AudioHwDevice.h
@@ -0,0 +1,88 @@
+/*
+**
+** 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.
+*/
+
+#ifndef ANDROID_AUDIO_HW_DEVICE_H
+#define ANDROID_AUDIO_HW_DEVICE_H
+
+#include <stdint.h>
+#include <stdlib.h>
+#include <sys/types.h>
+
+#include <hardware/audio.h>
+#include <utils/Errors.h>
+#include <system/audio.h>
+
+
+namespace android {
+
+class AudioStreamOut;
+
+class AudioHwDevice {
+public:
+ enum Flags {
+ AHWD_CAN_SET_MASTER_VOLUME = 0x1,
+ AHWD_CAN_SET_MASTER_MUTE = 0x2,
+ };
+
+ AudioHwDevice(audio_module_handle_t handle,
+ const char *moduleName,
+ audio_hw_device_t *hwDevice,
+ Flags flags)
+ : mHandle(handle)
+ , mModuleName(strdup(moduleName))
+ , mHwDevice(hwDevice)
+ , mFlags(flags) { }
+ virtual ~AudioHwDevice() { free((void *)mModuleName); }
+
+ bool canSetMasterVolume() const {
+ return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME));
+ }
+
+ bool canSetMasterMute() const {
+ return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE));
+ }
+
+ audio_module_handle_t handle() const { return mHandle; }
+ const char *moduleName() const { return mModuleName; }
+ audio_hw_device_t *hwDevice() const { return mHwDevice; }
+ uint32_t version() const { return mHwDevice->common.version; }
+
+ /** This method creates and opens the audio hardware output stream.
+ * The "address" parameter qualifies the "devices" audio device type if needed.
+ * The format format depends on the device type:
+ * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC"
+ * - USB devices use the ALSA card and device numbers in the form "card=X;device=Y"
+ * - Other devices may use a number or any other string.
+ */
+ status_t openOutputStream(
+ AudioStreamOut **ppStreamOut,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ const char *address);
+
+private:
+ const audio_module_handle_t mHandle;
+ const char * const mModuleName;
+ audio_hw_device_t * const mHwDevice;
+ const Flags mFlags;
+};
+
+} // namespace android
+
+#endif // ANDROID_AUDIO_HW_DEVICE_H
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index fd28ea1..dddca02 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -69,9 +69,9 @@
#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
#endif
-// Set kUseNewMixer to true to use the new mixer engine. Otherwise the
-// original code will be used. This is false for now.
-static const bool kUseNewMixer = false;
+// 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;
// Set kUseFloat to true to allow floating input into the mixer engine.
// If kUseNewMixer is false, this is ignored or may be overridden internally
@@ -341,11 +341,46 @@
ALOGV("RemixBufferProvider(%p)(%#x, %#x, %#x) %zu %zu",
this, format, inputChannelMask, outputChannelMask,
mInputChannels, mOutputChannels);
- // TODO: consider channel representation in index array formulation
- // We ignore channel representation, and just use the bits.
- memcpy_by_index_array_initialization(mIdxAry, ARRAY_SIZE(mIdxAry),
- audio_channel_mask_get_bits(outputChannelMask),
- audio_channel_mask_get_bits(inputChannelMask));
+
+ const audio_channel_representation_t inputRepresentation =
+ audio_channel_mask_get_representation(inputChannelMask);
+ const audio_channel_representation_t outputRepresentation =
+ audio_channel_mask_get_representation(outputChannelMask);
+ const uint32_t inputBits = audio_channel_mask_get_bits(inputChannelMask);
+ const uint32_t outputBits = audio_channel_mask_get_bits(outputChannelMask);
+
+ switch (inputRepresentation) {
+ case AUDIO_CHANNEL_REPRESENTATION_POSITION:
+ switch (outputRepresentation) {
+ case AUDIO_CHANNEL_REPRESENTATION_POSITION:
+ memcpy_by_index_array_initialization(mIdxAry, ARRAY_SIZE(mIdxAry),
+ outputBits, inputBits);
+ return;
+ case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+ // TODO: output channel index mask not currently allowed
+ // fall through
+ default:
+ break;
+ }
+ break;
+ case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+ switch (outputRepresentation) {
+ case AUDIO_CHANNEL_REPRESENTATION_POSITION:
+ memcpy_by_index_array_initialization_src_index(mIdxAry, ARRAY_SIZE(mIdxAry),
+ outputBits, inputBits);
+ return;
+ case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+ // TODO: output channel index mask not currently allowed
+ // fall through
+ default:
+ break;
+ }
+ break;
+ default:
+ break;
+ }
+ LOG_ALWAYS_FATAL("invalid channel mask conversion from %#x to %#x",
+ inputChannelMask, outputChannelMask);
}
void AudioMixer::RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
@@ -430,6 +465,10 @@
mState.mLog = log;
}
+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)
{
@@ -492,24 +531,23 @@
t->mInputBufferProvider = NULL;
t->mReformatBufferProvider = NULL;
t->downmixerBufferProvider = NULL;
+ t->mPostDownmixReformatBufferProvider = NULL;
t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
t->mFormat = format;
- t->mMixerInFormat = kUseFloat && kUseNewMixer
- ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ t->mMixerInFormat = selectMixerInFormat(format);
+ t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
// Check the downmixing (or upmixing) requirements.
- status_t status = initTrackDownmix(t, n);
+ status_t status = t->prepareForDownmix();
if (status != OK) {
ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
return -1;
}
- // initTrackDownmix() may change the input format requirement.
- // If you desire floating point input to the mixer, it may change
- // to integer because the downmixer requires integer to process.
+ // prepareForDownmix() may change mDownmixRequiresFormat
ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
- prepareTrackForReformat(t, n);
+ t->prepareForReformat();
mTrackNames |= 1 << n;
return TRACK0 + n;
}
@@ -526,7 +564,7 @@
}
// Called when channel masks have changed for a track name
-// TODO: Fix Downmixbufferprofider not to (possibly) change mixer input format,
+// 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) {
@@ -551,21 +589,18 @@
// 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 prevMixerInFormat = track.mMixerInFormat;
- track.mMixerInFormat = kUseFloat && kUseNewMixer
- ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
- const status_t status = initTrackDownmix(&mState.tracks[name], name);
+ const audio_format_t prevDownmixerFormat = track.mDownmixRequiresFormat;
+ const status_t status = mState.tracks[name].prepareForDownmix();
ALOGE_IF(status != OK,
- "initTrackDownmix error %d, track channel mask %#x, mixer channel mask %#x",
+ "prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x",
status, track.channelMask, track.mMixerChannelMask);
- const bool mixerInFormatChanged = prevMixerInFormat != track.mMixerInFormat;
- if (mixerInFormatChanged) {
- prepareTrackForReformat(&track, name); // because of downmixer, track format may change!
+ if (prevDownmixerFormat != track.mDownmixRequiresFormat) {
+ track.prepareForReformat(); // because of downmixer, track format may change!
}
- if (track.resampler && (mixerInFormatChanged || mixerChannelCountChanged)) {
- // resampler input format or channels may have changed.
+ if (track.resampler && mixerChannelCountChanged) {
+ // resampler channels may have changed.
const uint32_t resetToSampleRate = track.sampleRate;
delete track.resampler;
track.resampler = NULL;
@@ -576,99 +611,125 @@
return true;
}
-status_t AudioMixer::initTrackDownmix(track_t* pTrack, int trackName)
-{
- // Only remix (upmix or downmix) if the track and mixer/device channel masks
- // are not the same and not handled internally, as mono -> stereo currently is.
- if (pTrack->channelMask != pTrack->mMixerChannelMask
- && !(pTrack->channelMask == AUDIO_CHANNEL_OUT_MONO
- && pTrack->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) {
- return prepareTrackForDownmix(pTrack, trackName);
- }
- // no remix necessary
- unprepareTrackForDownmix(pTrack, trackName);
- return NO_ERROR;
-}
+void AudioMixer::track_t::unprepareForDownmix() {
+ ALOGV("AudioMixer::unprepareForDownmix(%p)", this);
-void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName __unused) {
- ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName);
-
- if (pTrack->downmixerBufferProvider != NULL) {
+ mDownmixRequiresFormat = AUDIO_FORMAT_INVALID;
+ if (downmixerBufferProvider != NULL) {
// this track had previously been configured with a downmixer, delete it
ALOGV(" deleting old downmixer");
- delete pTrack->downmixerBufferProvider;
- pTrack->downmixerBufferProvider = NULL;
- reconfigureBufferProviders(pTrack);
+ delete downmixerBufferProvider;
+ downmixerBufferProvider = NULL;
+ reconfigureBufferProviders();
} else {
ALOGV(" nothing to do, no downmixer to delete");
}
}
-status_t AudioMixer::prepareTrackForDownmix(track_t* pTrack, int trackName)
+status_t AudioMixer::track_t::prepareForDownmix()
{
- ALOGV("AudioMixer::prepareTrackForDownmix(%d) with mask 0x%x", trackName, pTrack->channelMask);
+ ALOGV("AudioMixer::prepareForDownmix(%p) with mask 0x%x",
+ this, channelMask);
// discard the previous downmixer if there was one
- unprepareTrackForDownmix(pTrack, trackName);
- if (DownmixerBufferProvider::isMultichannelCapable()) {
- DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(pTrack->channelMask,
- pTrack->mMixerChannelMask,
- AUDIO_FORMAT_PCM_16_BIT /* TODO: use pTrack->mMixerInFormat, now only PCM 16 */,
- pTrack->sampleRate, pTrack->sessionId, kCopyBufferFrameCount);
+ unprepareForDownmix();
+ // Only remix (upmix or downmix) if the track and mixer/device channel masks
+ // are not the same and not handled internally, as mono -> stereo currently is.
+ if (channelMask == mMixerChannelMask
+ || (channelMask == AUDIO_CHANNEL_OUT_MONO
+ && mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) {
+ return NO_ERROR;
+ }
+ // DownmixerBufferProvider is only used for position masks.
+ if (audio_channel_mask_get_representation(channelMask)
+ == AUDIO_CHANNEL_REPRESENTATION_POSITION
+ && DownmixerBufferProvider::isMultichannelCapable()) {
+ DownmixerBufferProvider* pDbp = 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
- pTrack->mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
- pTrack->downmixerBufferProvider = pDbp;
- reconfigureBufferProviders(pTrack);
+ mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
+ downmixerBufferProvider = pDbp;
+ reconfigureBufferProviders();
return NO_ERROR;
}
delete pDbp;
}
// Effect downmixer does not accept the channel conversion. Let's use our remixer.
- RemixBufferProvider* pRbp = new RemixBufferProvider(pTrack->channelMask,
- pTrack->mMixerChannelMask, pTrack->mMixerInFormat, kCopyBufferFrameCount);
+ RemixBufferProvider* pRbp = new RemixBufferProvider(channelMask,
+ mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount);
// Remix always finds a conversion whereas Downmixer effect above may fail.
- pTrack->downmixerBufferProvider = pRbp;
- reconfigureBufferProviders(pTrack);
+ downmixerBufferProvider = pRbp;
+ reconfigureBufferProviders();
return NO_ERROR;
}
-void AudioMixer::unprepareTrackForReformat(track_t* pTrack, int trackName __unused) {
- ALOGV("AudioMixer::unprepareTrackForReformat(%d)", trackName);
- if (pTrack->mReformatBufferProvider != NULL) {
- delete pTrack->mReformatBufferProvider;
- pTrack->mReformatBufferProvider = NULL;
- reconfigureBufferProviders(pTrack);
+void AudioMixer::track_t::unprepareForReformat() {
+ ALOGV("AudioMixer::unprepareForReformat(%p)", this);
+ bool requiresReconfigure = false;
+ if (mReformatBufferProvider != NULL) {
+ delete mReformatBufferProvider;
+ mReformatBufferProvider = NULL;
+ requiresReconfigure = true;
+ }
+ if (mPostDownmixReformatBufferProvider != NULL) {
+ delete mPostDownmixReformatBufferProvider;
+ mPostDownmixReformatBufferProvider = NULL;
+ requiresReconfigure = true;
+ }
+ if (requiresReconfigure) {
+ reconfigureBufferProviders();
}
}
-status_t AudioMixer::prepareTrackForReformat(track_t* pTrack, int trackName)
+status_t AudioMixer::track_t::prepareForReformat()
{
- ALOGV("AudioMixer::prepareTrackForReformat(%d) with format %#x", trackName, pTrack->mFormat);
- // discard the previous reformatter if there was one
- unprepareTrackForReformat(pTrack, trackName);
- // only configure reformatter if needed
- if (pTrack->mFormat != pTrack->mMixerInFormat) {
- pTrack->mReformatBufferProvider = new ReformatBufferProvider(
- audio_channel_count_from_out_mask(pTrack->channelMask),
- pTrack->mFormat, pTrack->mMixerInFormat,
+ ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat);
+ // discard previous reformatters
+ unprepareForReformat();
+ // only configure reformatters as needed
+ const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID
+ ? mDownmixRequiresFormat : mMixerInFormat;
+ bool requiresReconfigure = false;
+ if (mFormat != targetFormat) {
+ mReformatBufferProvider = new ReformatBufferProvider(
+ audio_channel_count_from_out_mask(channelMask),
+ mFormat,
+ targetFormat,
kCopyBufferFrameCount);
- reconfigureBufferProviders(pTrack);
+ requiresReconfigure = true;
+ }
+ if (targetFormat != mMixerInFormat) {
+ mPostDownmixReformatBufferProvider = new ReformatBufferProvider(
+ audio_channel_count_from_out_mask(mMixerChannelMask),
+ targetFormat,
+ mMixerInFormat,
+ kCopyBufferFrameCount);
+ requiresReconfigure = true;
+ }
+ if (requiresReconfigure) {
+ reconfigureBufferProviders();
}
return NO_ERROR;
}
-void AudioMixer::reconfigureBufferProviders(track_t* pTrack)
+void AudioMixer::track_t::reconfigureBufferProviders()
{
- pTrack->bufferProvider = pTrack->mInputBufferProvider;
- if (pTrack->mReformatBufferProvider) {
- pTrack->mReformatBufferProvider->setBufferProvider(pTrack->bufferProvider);
- pTrack->bufferProvider = pTrack->mReformatBufferProvider;
+ bufferProvider = mInputBufferProvider;
+ if (mReformatBufferProvider) {
+ mReformatBufferProvider->setBufferProvider(bufferProvider);
+ bufferProvider = mReformatBufferProvider;
}
- if (pTrack->downmixerBufferProvider) {
- pTrack->downmixerBufferProvider->setBufferProvider(pTrack->bufferProvider);
- pTrack->bufferProvider = pTrack->downmixerBufferProvider;
+ if (downmixerBufferProvider) {
+ downmixerBufferProvider->setBufferProvider(bufferProvider);
+ bufferProvider = downmixerBufferProvider;
+ }
+ if (mPostDownmixReformatBufferProvider) {
+ mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider);
+ bufferProvider = mPostDownmixReformatBufferProvider;
}
}
@@ -687,9 +748,9 @@
delete track.resampler;
track.resampler = NULL;
// delete the downmixer
- unprepareTrackForDownmix(&mState.tracks[name], name);
+ mState.tracks[name].unprepareForDownmix();
// delete the reformatter
- unprepareTrackForReformat(&mState.tracks[name], name);
+ mState.tracks[name].unprepareForReformat();
mTrackNames &= ~(1<<name);
}
@@ -828,7 +889,7 @@
ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
track.mFormat = format;
ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
- prepareTrackForReformat(&track, name);
+ track.prepareForReformat();
invalidateState(1 << name);
}
} break;
@@ -1032,10 +1093,13 @@
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();
}
mState.tracks[name].mInputBufferProvider = bufferProvider;
- reconfigureBufferProviders(&mState.tracks[name]);
+ mState.tracks[name].reconfigureBufferProviders();
}
@@ -2236,4 +2300,4 @@
}
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index f4f142b..381036b 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -127,10 +127,16 @@
size_t getUnreleasedFrames(int name) const;
static inline bool isValidPcmTrackFormat(audio_format_t format) {
- return format == AUDIO_FORMAT_PCM_16_BIT ||
- format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
- format == AUDIO_FORMAT_PCM_32_BIT ||
- format == AUDIO_FORMAT_PCM_FLOAT;
+ switch (format) {
+ case AUDIO_FORMAT_PCM_8_BIT:
+ case AUDIO_FORMAT_PCM_16_BIT:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_32_BIT:
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return true;
+ default:
+ return false;
+ }
}
private:
@@ -205,17 +211,34 @@
int32_t* auxBuffer;
// 16-byte boundary
+
+ /* Buffer providers are constructed to translate the track input data as needed.
+ *
+ * 1) mInputBufferProvider: The AudioTrack buffer provider.
+ * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to
+ * 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
+ * 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.
+ */
AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
CopyBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting.
CopyBufferProvider* downmixerBufferProvider; // wrapper for channel conversion.
-
- int32_t sessionId;
+ CopyBufferProvider* mPostDownmixReformatBufferProvider;
// 16-byte boundary
+ int32_t sessionId;
+
audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
audio_format_t mFormat; // input track format
audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
// each track must be converted to this format.
+ audio_format_t mDownmixRequiresFormat; // required downmixer format
+ // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
+ // AUDIO_FORMAT_INVALID if no required format
float mVolume[MAX_NUM_VOLUMES]; // floating point set volume
float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
@@ -225,7 +248,6 @@
float mPrevAuxLevel; // floating point prev aux level
float mAuxInc; // floating point aux increment
- // 16-byte boundary
audio_channel_mask_t mMixerChannelMask;
uint32_t mMixerChannelCount;
@@ -236,6 +258,12 @@
void adjustVolumeRamp(bool aux, bool useFloat = false);
size_t getUnreleasedFrames() const { return resampler != NULL ?
resampler->getUnreleasedFrames() : 0; };
+
+ status_t prepareForDownmix();
+ void unprepareForDownmix();
+ status_t prepareForReformat();
+ void unprepareForReformat();
+ void reconfigureBufferProviders();
};
typedef void (*process_hook_t)(state_t* state, int64_t pts);
@@ -382,14 +410,6 @@
bool setChannelMasks(int name,
audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
- // TODO: remove unused trackName/trackNum from functions below.
- static status_t initTrackDownmix(track_t* pTrack, int trackName);
- static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
- static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
- static status_t prepareTrackForReformat(track_t* pTrack, int trackNum);
- static void unprepareTrackForReformat(track_t* pTrack, int trackName);
- static void reconfigureBufferProviders(track_t* pTrack);
-
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);
@@ -465,6 +485,6 @@
};
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
#endif // ANDROID_AUDIO_MIXER_H
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 069d946..863614a 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -170,7 +170,6 @@
};
// ----------------------------------------------------------------------------
-}
-; // namespace android
+} // namespace android
#endif // ANDROID_AUDIO_RESAMPLER_H
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 8f14ff9..d3cbd1c 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -185,5 +185,4 @@
}
// ----------------------------------------------------------------------------
-}
-; // namespace android
+} // namespace android
diff --git a/services/audioflinger/AudioResamplerCubic.h b/services/audioflinger/AudioResamplerCubic.h
index b315da5..1ddc5f9 100644
--- a/services/audioflinger/AudioResamplerCubic.h
+++ b/services/audioflinger/AudioResamplerCubic.h
@@ -63,6 +63,6 @@
};
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
#endif /*ANDROID_AUDIO_RESAMPLER_CUBIC_H*/
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 0eeb201..c21d4ca 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -618,4 +618,4 @@
template class AudioResamplerDyn<int32_t, int16_t, int32_t>;
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
index e886a68..238b163 100644
--- a/services/audioflinger/AudioResamplerDyn.h
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -127,6 +127,6 @@
void* mCoefBuffer; // if a filter is created, this is not null
};
-}; // namespace android
+} // namespace android
#endif /*ANDROID_AUDIO_RESAMPLER_DYN_H*/
diff --git a/services/audioflinger/AudioResamplerFirGen.h b/services/audioflinger/AudioResamplerFirGen.h
index f3718b6..ad18965 100644
--- a/services/audioflinger/AudioResamplerFirGen.h
+++ b/services/audioflinger/AudioResamplerFirGen.h
@@ -204,7 +204,8 @@
template <>
struct I0ATerm<0> { // 1/sqrt(2*PI);
- static const CONSTEXPR double value = 0.398942280401432677939946059934381868475858631164934657665925;
+ static const CONSTEXPR double value =
+ 0.398942280401432677939946059934381868475858631164934657665925;
};
#if USE_HORNERS_METHOD
@@ -706,6 +707,6 @@
}
}
-}; // namespace android
+} // namespace android
#endif /*ANDROID_AUDIO_RESAMPLER_FIR_GEN_H*/
diff --git a/services/audioflinger/AudioResamplerFirOps.h b/services/audioflinger/AudioResamplerFirOps.h
index bf2163f..658285d 100644
--- a/services/audioflinger/AudioResamplerFirOps.h
+++ b/services/audioflinger/AudioResamplerFirOps.h
@@ -25,7 +25,7 @@
#define USE_INLINE_ASSEMBLY (false)
#endif
-#if USE_INLINE_ASSEMBLY && defined(__ARM_NEON__)
+#if defined(__aarch64__) || defined(__ARM_NEON__)
#define USE_NEON (true)
#include <arm_neon.h>
#else
@@ -158,6 +158,6 @@
#endif
}
-}; // namespace android
+} // namespace android
#endif /*ANDROID_AUDIO_RESAMPLER_FIR_OPS_H*/
diff --git a/services/audioflinger/AudioResamplerFirProcess.h b/services/audioflinger/AudioResamplerFirProcess.h
index efc8055..176202e 100644
--- a/services/audioflinger/AudioResamplerFirProcess.h
+++ b/services/audioflinger/AudioResamplerFirProcess.h
@@ -174,7 +174,8 @@
* Process() calls ProcessBase() with TFUNC = InterpCompute, for interpolated phase.
*/
-template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO, typename TINTERP>
+template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO,
+ typename TINTERP>
static inline
void ProcessBase(TO* const out,
size_t count,
@@ -242,6 +243,9 @@
}
}
+/* Calculates a single output frame from a polyphase resampling filter.
+ * See Process() for parameter details.
+ */
template <int CHANNELS, int STRIDE, typename TC, typename TI, typename TO>
static inline
void ProcessL(TO* const out,
@@ -255,6 +259,39 @@
ProcessBase<CHANNELS, STRIDE, InterpNull>(out, count, coefsP, coefsN, sP, sN, 0, volumeLR);
}
+/*
+ * Calculates a single output frame from a polyphase resampling filter,
+ * with filter phase interpolation.
+ *
+ * @param out should point to the output buffer with space for at least one output frame.
+ *
+ * @param count should be half the size of the total filter length (halfNumCoefs), as we
+ * use symmetry in filter coefficients to evaluate two dot products.
+ *
+ * @param coefsP is one phase of the polyphase filter bank of size halfNumCoefs, corresponding
+ * to the positive sP.
+ *
+ * @param coefsN is one phase of the polyphase filter bank of size halfNumCoefs, corresponding
+ * to the negative sN.
+ *
+ * @param coefsP1 is the next phase of coefsP (used for interpolation).
+ *
+ * @param coefsN1 is the next phase of coefsN (used for interpolation).
+ *
+ * @param sP is the positive half of the coefficients (as viewed by a convolution),
+ * starting at the original samples pointer and decrementing (by CHANNELS).
+ *
+ * @param sN is the negative half of the samples (as viewed by a convolution),
+ * starting at the original samples pointer + CHANNELS and incrementing (by CHANNELS).
+ *
+ * @param lerpP The fractional siting between the polyphase indices is given by the bits
+ * below coefShift. See fir() for details.
+ *
+ * @param volumeLR is a pointer to an array of two 32 bit volume values, one per stereo channel,
+ * expressed as a S32 integer or float. A negative value inverts the channel 180 degrees.
+ * The pointer volumeLR should be aligned to a minimum of 8 bytes.
+ * A typical value for volume is 0x1000 to align to a unity gain output of 20.12.
+ */
template <int CHANNELS, int STRIDE, typename TC, typename TI, typename TO, typename TINTERP>
static inline
void Process(TO* const out,
@@ -268,11 +305,12 @@
TINTERP lerpP,
const TO* const volumeLR)
{
- ProcessBase<CHANNELS, STRIDE, InterpCompute>(out, count, coefsP, coefsN, sP, sN, lerpP, volumeLR);
+ ProcessBase<CHANNELS, STRIDE, InterpCompute>(out, count, coefsP, coefsN, sP, sN, lerpP,
+ volumeLR);
}
/*
- * Calculates a single output frame (two samples) from input sample pointer.
+ * Calculates a single output frame from input sample pointer.
*
* This sets up the params for the accelerated Process() and ProcessL()
* functions to do the appropriate dot products.
@@ -307,7 +345,7 @@
* the positive half of the filter is dot product from samples to samples-halfNumCoefs+1.
*
* @param volumeLR is a pointer to an array of two 32 bit volume values, one per stereo channel,
- * expressed as a S32 integer. A negative value inverts the channel 180 degrees.
+ * expressed as a S32 integer or float. A negative value inverts the channel 180 degrees.
* The pointer volumeLR should be aligned to a minimum of 8 bytes.
* A typical value for volume is 0x1000 to align to a unity gain output of 20.12.
*
@@ -396,6 +434,6 @@
}
}
-}; // namespace android
+} // namespace android
#endif /*ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_H*/
diff --git a/services/audioflinger/AudioResamplerFirProcessNeon.h b/services/audioflinger/AudioResamplerFirProcessNeon.h
index f311cef..3de9edd 100644
--- a/services/audioflinger/AudioResamplerFirProcessNeon.h
+++ b/services/audioflinger/AudioResamplerFirProcessNeon.h
@@ -22,14 +22,35 @@
// depends on AudioResamplerFirOps.h, AudioResamplerFirProcess.h
#if USE_NEON
+
+// use intrinsics if inline arm32 assembly is not possible
+#if !USE_INLINE_ASSEMBLY
+#define USE_INTRINSIC
+#endif
+
+// following intrinsics available only on ARM 64 bit ACLE
+#ifndef __aarch64__
+#undef vld1q_f32_x2
+#undef vld1q_s32_x2
+#endif
+
+#define TO_STRING2(x) #x
+#define TO_STRING(x) TO_STRING2(x)
+// uncomment to print GCC version, may be relevant for intrinsic optimizations
+/* #pragma message ("GCC version: " TO_STRING(__GNUC__) \
+ "." TO_STRING(__GNUC_MINOR__) \
+ "." TO_STRING(__GNUC_PATCHLEVEL__)) */
+
//
-// NEON specializations are enabled for Process() and ProcessL()
+// NEON specializations are enabled for Process() and ProcessL() in AudioResamplerFirProcess.h
//
-// TODO: Stride 16 and Stride 8 can be combined with one pass stride 8 (if necessary)
-// and looping stride 16 (or vice versa). This has some polyphase coef data alignment
-// issues with S16 coefs. Consider this later.
+// Two variants are presented here:
+// ARM NEON inline assembly which appears up to 10-15% faster than intrinsics (gcc 4.9) for arm32.
+// ARM NEON intrinsics which can also be used by arm64 and x86/64 with NEON header.
+//
// Macros to save a mono/stereo accumulator sample in q0 (and q4) as stereo out.
+// These are only used for inline assembly.
#define ASSEMBLY_ACCUMULATE_MONO \
"vld1.s32 {d2}, [%[vLR]:64] \n"/* (1) load volumes */\
"vld1.s32 {d3}, %[out] \n"/* (2) unaligned load the output */\
@@ -49,6 +70,458 @@
"vqadd.s32 d3, d3, d0 \n"/* (1+4d) accumulate result (saturating)*/\
"vst1.s32 {d3}, %[out] \n"/* (2+2d)store result*/
+template <int CHANNELS, int STRIDE, bool FIXED>
+static inline void ProcessNeonIntrinsic(int32_t* out,
+ int count,
+ const int16_t* coefsP,
+ const int16_t* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* volumeLR,
+ uint32_t lerpP,
+ const int16_t* coefsP1,
+ const int16_t* coefsN1)
+{
+ ALOG_ASSERT(count > 0 && (count & 7) == 0); // multiple of 8
+ COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS == 1 || CHANNELS == 2);
+
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ coefsP = (const int16_t*)__builtin_assume_aligned(coefsP, 16);
+ coefsN = (const int16_t*)__builtin_assume_aligned(coefsN, 16);
+
+ int16x4_t interp;
+ if (!FIXED) {
+ interp = vdup_n_s16(lerpP);
+ //interp = (int16x4_t)vset_lane_s32 ((int32x2_t)lerpP, interp, 0);
+ coefsP1 = (const int16_t*)__builtin_assume_aligned(coefsP1, 16);
+ coefsN1 = (const int16_t*)__builtin_assume_aligned(coefsN1, 16);
+ }
+ int32x4_t accum, accum2;
+ // warning uninitialized if we use veorq_s32
+ // (alternative to below) accum = veorq_s32(accum, accum);
+ accum = vdupq_n_s32(0);
+ if (CHANNELS == 2) {
+ // (alternative to below) accum2 = veorq_s32(accum2, accum2);
+ accum2 = vdupq_n_s32(0);
+ }
+ do {
+ int16x8_t posCoef = vld1q_s16(coefsP);
+ coefsP += 8;
+ int16x8_t negCoef = vld1q_s16(coefsN);
+ coefsN += 8;
+ if (!FIXED) { // interpolate
+ int16x8_t posCoef1 = vld1q_s16(coefsP1);
+ coefsP1 += 8;
+ int16x8_t negCoef1 = vld1q_s16(coefsN1);
+ coefsN1 += 8;
+
+ posCoef1 = vsubq_s16(posCoef1, posCoef);
+ negCoef = vsubq_s16(negCoef, negCoef1);
+
+ posCoef1 = vqrdmulhq_lane_s16(posCoef1, interp, 0);
+ negCoef = vqrdmulhq_lane_s16(negCoef, interp, 0);
+
+ posCoef = vaddq_s16(posCoef, posCoef1);
+ negCoef = vaddq_s16(negCoef, negCoef1);
+ }
+ switch (CHANNELS) {
+ case 1: {
+ int16x8_t posSamp = vld1q_s16(sP);
+ int16x8_t negSamp = vld1q_s16(sN);
+ sN += 8;
+ posSamp = vrev64q_s16(posSamp);
+
+ // dot product
+ accum = vmlal_s16(accum, vget_low_s16(posSamp), vget_high_s16(posCoef)); // reversed
+ accum = vmlal_s16(accum, vget_high_s16(posSamp), vget_low_s16(posCoef)); // reversed
+ accum = vmlal_s16(accum, vget_low_s16(negSamp), vget_low_s16(negCoef));
+ accum = vmlal_s16(accum, vget_high_s16(negSamp), vget_high_s16(negCoef));
+ sP -= 8;
+ } break;
+ case 2: {
+ int16x8x2_t posSamp = vld2q_s16(sP);
+ int16x8x2_t negSamp = vld2q_s16(sN);
+ sN += 16;
+ posSamp.val[0] = vrev64q_s16(posSamp.val[0]);
+ posSamp.val[1] = vrev64q_s16(posSamp.val[1]);
+
+ // dot product
+ accum = vmlal_s16(accum, vget_low_s16(posSamp.val[0]), vget_high_s16(posCoef)); // r
+ accum = vmlal_s16(accum, vget_high_s16(posSamp.val[0]), vget_low_s16(posCoef)); // r
+ accum2 = vmlal_s16(accum2, vget_low_s16(posSamp.val[1]), vget_high_s16(posCoef)); // r
+ accum2 = vmlal_s16(accum2, vget_high_s16(posSamp.val[1]), vget_low_s16(posCoef)); // r
+ accum = vmlal_s16(accum, vget_low_s16(negSamp.val[0]), vget_low_s16(negCoef));
+ accum = vmlal_s16(accum, vget_high_s16(negSamp.val[0]), vget_high_s16(negCoef));
+ accum2 = vmlal_s16(accum2, vget_low_s16(negSamp.val[1]), vget_low_s16(negCoef));
+ accum2 = vmlal_s16(accum2, vget_high_s16(negSamp.val[1]), vget_high_s16(negCoef));
+ sP -= 16;
+ }
+ } break;
+ } while (count -= 8);
+
+ // multiply by volume and save
+ volumeLR = (const int32_t*)__builtin_assume_aligned(volumeLR, 8);
+ int32x2_t vLR = vld1_s32(volumeLR);
+ int32x2_t outSamp = vld1_s32(out);
+ // combine and funnel down accumulator
+ int32x2_t outAccum = vpadd_s32(vget_low_s32(accum), vget_high_s32(accum));
+ if (CHANNELS == 1) {
+ // duplicate accum to both L and R
+ outAccum = vpadd_s32(outAccum, outAccum);
+ } else if (CHANNELS == 2) {
+ // accum2 contains R, fold in
+ int32x2_t outAccum2 = vpadd_s32(vget_low_s32(accum2), vget_high_s32(accum2));
+ outAccum = vpadd_s32(outAccum, outAccum2);
+ }
+ outAccum = vqrdmulh_s32(outAccum, vLR);
+ outSamp = vqadd_s32(outSamp, outAccum);
+ vst1_s32(out, outSamp);
+}
+
+template <int CHANNELS, int STRIDE, bool FIXED>
+static inline void ProcessNeonIntrinsic(int32_t* out,
+ int count,
+ const int32_t* coefsP,
+ const int32_t* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* volumeLR,
+ uint32_t lerpP,
+ const int32_t* coefsP1,
+ const int32_t* coefsN1)
+{
+ ALOG_ASSERT(count > 0 && (count & 7) == 0); // multiple of 8
+ COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS == 1 || CHANNELS == 2);
+
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ coefsP = (const int32_t*)__builtin_assume_aligned(coefsP, 16);
+ coefsN = (const int32_t*)__builtin_assume_aligned(coefsN, 16);
+
+ int32x2_t interp;
+ if (!FIXED) {
+ interp = vdup_n_s32(lerpP);
+ coefsP1 = (const int32_t*)__builtin_assume_aligned(coefsP1, 16);
+ coefsN1 = (const int32_t*)__builtin_assume_aligned(coefsN1, 16);
+ }
+ int32x4_t accum, accum2;
+ // warning uninitialized if we use veorq_s32
+ // (alternative to below) accum = veorq_s32(accum, accum);
+ accum = vdupq_n_s32(0);
+ if (CHANNELS == 2) {
+ // (alternative to below) accum2 = veorq_s32(accum2, accum2);
+ accum2 = vdupq_n_s32(0);
+ }
+ do {
+#ifdef vld1q_s32_x2
+ int32x4x2_t posCoef = vld1q_s32_x2(coefsP);
+ coefsP += 8;
+ int32x4x2_t negCoef = vld1q_s32_x2(coefsN);
+ coefsN += 8;
+#else
+ int32x4x2_t posCoef;
+ posCoef.val[0] = vld1q_s32(coefsP);
+ coefsP += 4;
+ posCoef.val[1] = vld1q_s32(coefsP);
+ coefsP += 4;
+ int32x4x2_t negCoef;
+ negCoef.val[0] = vld1q_s32(coefsN);
+ coefsN += 4;
+ negCoef.val[1] = vld1q_s32(coefsN);
+ coefsN += 4;
+#endif
+ if (!FIXED) { // interpolate
+#ifdef vld1q_s32_x2
+ int32x4x2_t posCoef1 = vld1q_s32_x2(coefsP1);
+ coefsP1 += 8;
+ int32x4x2_t negCoef1 = vld1q_s32_x2(coefsN1);
+ coefsN1 += 8;
+#else
+ int32x4x2_t posCoef1;
+ posCoef1.val[0] = vld1q_s32(coefsP1);
+ coefsP1 += 4;
+ posCoef1.val[1] = vld1q_s32(coefsP1);
+ coefsP1 += 4;
+ int32x4x2_t negCoef1;
+ negCoef1.val[0] = vld1q_s32(coefsN1);
+ coefsN1 += 4;
+ negCoef1.val[1] = vld1q_s32(coefsN1);
+ coefsN1 += 4;
+#endif
+
+ posCoef1.val[0] = vsubq_s32(posCoef1.val[0], posCoef.val[0]);
+ posCoef1.val[1] = vsubq_s32(posCoef1.val[1], posCoef.val[1]);
+ negCoef.val[0] = vsubq_s32(negCoef.val[0], negCoef1.val[0]);
+ negCoef.val[1] = vsubq_s32(negCoef.val[1], negCoef1.val[1]);
+
+ posCoef1.val[0] = vqrdmulhq_lane_s32(posCoef1.val[0], interp, 0);
+ posCoef1.val[1] = vqrdmulhq_lane_s32(posCoef1.val[1], interp, 0);
+ negCoef.val[0] = vqrdmulhq_lane_s32(negCoef.val[0], interp, 0);
+ negCoef.val[1] = vqrdmulhq_lane_s32(negCoef.val[1], interp, 0);
+
+ posCoef.val[0] = vaddq_s32(posCoef.val[0], posCoef1.val[0]);
+ posCoef.val[1] = vaddq_s32(posCoef.val[1], posCoef1.val[1]);
+ negCoef.val[0] = vaddq_s32(negCoef.val[0], negCoef1.val[0]);
+ negCoef.val[1] = vaddq_s32(negCoef.val[1], negCoef1.val[1]);
+ }
+ switch (CHANNELS) {
+ case 1: {
+ int16x8_t posSamp = vld1q_s16(sP);
+ int16x8_t negSamp = vld1q_s16(sN);
+ sN += 8;
+ posSamp = vrev64q_s16(posSamp);
+
+ int32x4_t posSamp0 = vshll_n_s16(vget_low_s16(posSamp), 15);
+ int32x4_t posSamp1 = vshll_n_s16(vget_high_s16(posSamp), 15);
+ int32x4_t negSamp0 = vshll_n_s16(vget_low_s16(negSamp), 15);
+ int32x4_t negSamp1 = vshll_n_s16(vget_high_s16(negSamp), 15);
+
+ // dot product
+ posSamp0 = vqrdmulhq_s32(posSamp0, posCoef.val[1]); // reversed
+ posSamp1 = vqrdmulhq_s32(posSamp1, posCoef.val[0]); // reversed
+ negSamp0 = vqrdmulhq_s32(negSamp0, negCoef.val[0]);
+ negSamp1 = vqrdmulhq_s32(negSamp1, negCoef.val[1]);
+
+ accum = vaddq_s32(accum, posSamp0);
+ negSamp0 = vaddq_s32(negSamp0, negSamp1);
+ accum = vaddq_s32(accum, posSamp1);
+ accum = vaddq_s32(accum, negSamp0);
+
+ sP -= 8;
+ } break;
+ case 2: {
+ int16x8x2_t posSamp = vld2q_s16(sP);
+ int16x8x2_t negSamp = vld2q_s16(sN);
+ sN += 16;
+ posSamp.val[0] = vrev64q_s16(posSamp.val[0]);
+ posSamp.val[1] = vrev64q_s16(posSamp.val[1]);
+
+ // left
+ int32x4_t posSamp0 = vshll_n_s16(vget_low_s16(posSamp.val[0]), 15);
+ int32x4_t posSamp1 = vshll_n_s16(vget_high_s16(posSamp.val[0]), 15);
+ int32x4_t negSamp0 = vshll_n_s16(vget_low_s16(negSamp.val[0]), 15);
+ int32x4_t negSamp1 = vshll_n_s16(vget_high_s16(negSamp.val[0]), 15);
+
+ // dot product
+ posSamp0 = vqrdmulhq_s32(posSamp0, posCoef.val[1]); // reversed
+ posSamp1 = vqrdmulhq_s32(posSamp1, posCoef.val[0]); // reversed
+ negSamp0 = vqrdmulhq_s32(negSamp0, negCoef.val[0]);
+ negSamp1 = vqrdmulhq_s32(negSamp1, negCoef.val[1]);
+
+ accum = vaddq_s32(accum, posSamp0);
+ negSamp0 = vaddq_s32(negSamp0, negSamp1);
+ accum = vaddq_s32(accum, posSamp1);
+ accum = vaddq_s32(accum, negSamp0);
+
+ // right
+ posSamp0 = vshll_n_s16(vget_low_s16(posSamp.val[1]), 15);
+ posSamp1 = vshll_n_s16(vget_high_s16(posSamp.val[1]), 15);
+ negSamp0 = vshll_n_s16(vget_low_s16(negSamp.val[1]), 15);
+ negSamp1 = vshll_n_s16(vget_high_s16(negSamp.val[1]), 15);
+
+ // dot product
+ posSamp0 = vqrdmulhq_s32(posSamp0, posCoef.val[1]); // reversed
+ posSamp1 = vqrdmulhq_s32(posSamp1, posCoef.val[0]); // reversed
+ negSamp0 = vqrdmulhq_s32(negSamp0, negCoef.val[0]);
+ negSamp1 = vqrdmulhq_s32(negSamp1, negCoef.val[1]);
+
+ accum2 = vaddq_s32(accum2, posSamp0);
+ negSamp0 = vaddq_s32(negSamp0, negSamp1);
+ accum2 = vaddq_s32(accum2, posSamp1);
+ accum2 = vaddq_s32(accum2, negSamp0);
+
+ sP -= 16;
+ } break;
+ }
+ } while (count -= 8);
+
+ // multiply by volume and save
+ volumeLR = (const int32_t*)__builtin_assume_aligned(volumeLR, 8);
+ int32x2_t vLR = vld1_s32(volumeLR);
+ int32x2_t outSamp = vld1_s32(out);
+ // combine and funnel down accumulator
+ int32x2_t outAccum = vpadd_s32(vget_low_s32(accum), vget_high_s32(accum));
+ if (CHANNELS == 1) {
+ // duplicate accum to both L and R
+ outAccum = vpadd_s32(outAccum, outAccum);
+ } else if (CHANNELS == 2) {
+ // accum2 contains R, fold in
+ int32x2_t outAccum2 = vpadd_s32(vget_low_s32(accum2), vget_high_s32(accum2));
+ outAccum = vpadd_s32(outAccum, outAccum2);
+ }
+ outAccum = vqrdmulh_s32(outAccum, vLR);
+ outSamp = vqadd_s32(outSamp, outAccum);
+ vst1_s32(out, outSamp);
+}
+
+template <int CHANNELS, int STRIDE, bool FIXED>
+static inline void ProcessNeonIntrinsic(float* out,
+ int count,
+ const float* coefsP,
+ const float* coefsN,
+ const float* sP,
+ const float* sN,
+ const float* volumeLR,
+ float lerpP,
+ const float* coefsP1,
+ const float* coefsN1)
+{
+ ALOG_ASSERT(count > 0 && (count & 7) == 0); // multiple of 8
+ COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS == 1 || CHANNELS == 2);
+
+ sP -= CHANNELS*((STRIDE>>1)-1);
+ coefsP = (const float*)__builtin_assume_aligned(coefsP, 16);
+ coefsN = (const float*)__builtin_assume_aligned(coefsN, 16);
+
+ float32x2_t interp;
+ if (!FIXED) {
+ interp = vdup_n_f32(lerpP);
+ coefsP1 = (const float*)__builtin_assume_aligned(coefsP1, 16);
+ coefsN1 = (const float*)__builtin_assume_aligned(coefsN1, 16);
+ }
+ float32x4_t accum, accum2;
+ // warning uninitialized if we use veorq_s32
+ // (alternative to below) accum = veorq_s32(accum, accum);
+ accum = vdupq_n_f32(0);
+ if (CHANNELS == 2) {
+ // (alternative to below) accum2 = veorq_s32(accum2, accum2);
+ accum2 = vdupq_n_f32(0);
+ }
+ do {
+#ifdef vld1q_f32_x2
+ float32x4x2_t posCoef = vld1q_f32_x2(coefsP);
+ coefsP += 8;
+ float32x4x2_t negCoef = vld1q_f32_x2(coefsN);
+ coefsN += 8;
+#else
+ float32x4x2_t posCoef;
+ posCoef.val[0] = vld1q_f32(coefsP);
+ coefsP += 4;
+ posCoef.val[1] = vld1q_f32(coefsP);
+ coefsP += 4;
+ float32x4x2_t negCoef;
+ negCoef.val[0] = vld1q_f32(coefsN);
+ coefsN += 4;
+ negCoef.val[1] = vld1q_f32(coefsN);
+ coefsN += 4;
+#endif
+ if (!FIXED) { // interpolate
+#ifdef vld1q_f32_x2
+ float32x4x2_t posCoef1 = vld1q_f32_x2(coefsP1);
+ coefsP1 += 8;
+ float32x4x2_t negCoef1 = vld1q_f32_x2(coefsN1);
+ coefsN1 += 8;
+#else
+ float32x4x2_t posCoef1;
+ posCoef1.val[0] = vld1q_f32(coefsP1);
+ coefsP1 += 4;
+ posCoef1.val[1] = vld1q_f32(coefsP1);
+ coefsP1 += 4;
+ float32x4x2_t negCoef1;
+ negCoef1.val[0] = vld1q_f32(coefsN1);
+ coefsN1 += 4;
+ negCoef1.val[1] = vld1q_f32(coefsN1);
+ coefsN1 += 4;
+#endif
+ posCoef1.val[0] = vsubq_f32(posCoef1.val[0], posCoef.val[0]);
+ posCoef1.val[1] = vsubq_f32(posCoef1.val[1], posCoef.val[1]);
+ negCoef.val[0] = vsubq_f32(negCoef.val[0], negCoef1.val[0]);
+ negCoef.val[1] = vsubq_f32(negCoef.val[1], negCoef1.val[1]);
+
+ posCoef.val[0] = vmlaq_lane_f32(posCoef.val[0], posCoef1.val[0], interp, 0);
+ posCoef.val[1] = vmlaq_lane_f32(posCoef.val[1], posCoef1.val[1], interp, 0);
+ negCoef.val[0] = vmlaq_lane_f32(negCoef1.val[0], negCoef.val[0], interp, 0); // rev
+ negCoef.val[1] = vmlaq_lane_f32(negCoef1.val[1], negCoef.val[1], interp, 0); // rev
+ }
+ switch (CHANNELS) {
+ case 1: {
+#ifdef vld1q_f32_x2
+ float32x4x2_t posSamp = vld1q_f32_x2(sP);
+ float32x4x2_t negSamp = vld1q_f32_x2(sN);
+ sN += 8;
+ sP -= 8;
+#else
+ float32x4x2_t posSamp;
+ posSamp.val[0] = vld1q_f32(sP);
+ sP += 4;
+ posSamp.val[1] = vld1q_f32(sP);
+ sP -= 12;
+ float32x4x2_t negSamp;
+ negSamp.val[0] = vld1q_f32(sN);
+ sN += 4;
+ negSamp.val[1] = vld1q_f32(sN);
+ sN += 4;
+#endif
+ // effectively we want a vrev128q_f32()
+ posSamp.val[0] = vrev64q_f32(posSamp.val[0]);
+ posSamp.val[1] = vrev64q_f32(posSamp.val[1]);
+ posSamp.val[0] = vcombine_f32(
+ vget_high_f32(posSamp.val[0]), vget_low_f32(posSamp.val[0]));
+ posSamp.val[1] = vcombine_f32(
+ vget_high_f32(posSamp.val[1]), vget_low_f32(posSamp.val[1]));
+
+ accum = vmlaq_f32(accum, posSamp.val[0], posCoef.val[1]);
+ accum = vmlaq_f32(accum, posSamp.val[1], posCoef.val[0]);
+ accum = vmlaq_f32(accum, negSamp.val[0], negCoef.val[0]);
+ accum = vmlaq_f32(accum, negSamp.val[1], negCoef.val[1]);
+ } break;
+ case 2: {
+ float32x4x2_t posSamp0 = vld2q_f32(sP);
+ sP += 8;
+ float32x4x2_t negSamp0 = vld2q_f32(sN);
+ sN += 8;
+ posSamp0.val[0] = vrev64q_f32(posSamp0.val[0]);
+ posSamp0.val[1] = vrev64q_f32(posSamp0.val[1]);
+ posSamp0.val[0] = vcombine_f32(
+ vget_high_f32(posSamp0.val[0]), vget_low_f32(posSamp0.val[0]));
+ posSamp0.val[1] = vcombine_f32(
+ vget_high_f32(posSamp0.val[1]), vget_low_f32(posSamp0.val[1]));
+
+ float32x4x2_t posSamp1 = vld2q_f32(sP);
+ sP -= 24;
+ float32x4x2_t negSamp1 = vld2q_f32(sN);
+ sN += 8;
+ posSamp1.val[0] = vrev64q_f32(posSamp1.val[0]);
+ posSamp1.val[1] = vrev64q_f32(posSamp1.val[1]);
+ posSamp1.val[0] = vcombine_f32(
+ vget_high_f32(posSamp1.val[0]), vget_low_f32(posSamp1.val[0]));
+ posSamp1.val[1] = vcombine_f32(
+ vget_high_f32(posSamp1.val[1]), vget_low_f32(posSamp1.val[1]));
+
+ // Note: speed is affected by accumulation order.
+ // Also, speed appears slower using vmul/vadd instead of vmla for
+ // stereo case, comparable for mono.
+
+ accum = vmlaq_f32(accum, negSamp0.val[0], negCoef.val[0]);
+ accum = vmlaq_f32(accum, negSamp1.val[0], negCoef.val[1]);
+ accum2 = vmlaq_f32(accum2, negSamp0.val[1], negCoef.val[0]);
+ accum2 = vmlaq_f32(accum2, negSamp1.val[1], negCoef.val[1]);
+
+ accum = vmlaq_f32(accum, posSamp0.val[0], posCoef.val[1]); // reversed
+ accum = vmlaq_f32(accum, posSamp1.val[0], posCoef.val[0]); // reversed
+ accum2 = vmlaq_f32(accum2, posSamp0.val[1], posCoef.val[1]); // reversed
+ accum2 = vmlaq_f32(accum2, posSamp1.val[1], posCoef.val[0]); // reversed
+ } break;
+ }
+ } while (count -= 8);
+
+ // multiply by volume and save
+ volumeLR = (const float*)__builtin_assume_aligned(volumeLR, 8);
+ float32x2_t vLR = vld1_f32(volumeLR);
+ float32x2_t outSamp = vld1_f32(out);
+ // combine and funnel down accumulator
+ float32x2_t outAccum = vpadd_f32(vget_low_f32(accum), vget_high_f32(accum));
+ if (CHANNELS == 1) {
+ // duplicate accum to both L and R
+ outAccum = vpadd_f32(outAccum, outAccum);
+ } else if (CHANNELS == 2) {
+ // accum2 contains R, fold in
+ float32x2_t outAccum2 = vpadd_f32(vget_low_f32(accum2), vget_high_f32(accum2));
+ outAccum = vpadd_f32(outAccum, outAccum2);
+ }
+ outSamp = vmla_f32(outSamp, outAccum, vLR);
+ vst1_f32(out, outSamp);
+}
+
template <>
inline void ProcessL<1, 16>(int32_t* const out,
int count,
@@ -58,6 +531,10 @@
const int16_t* sN,
const int32_t* const volumeLR)
{
+#ifdef USE_INTRINSIC
+ ProcessNeonIntrinsic<1, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ 0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
+#else
const int CHANNELS = 1; // template specialization does not preserve params
const int STRIDE = 16;
sP -= CHANNELS*((STRIDE>>1)-1);
@@ -99,6 +576,7 @@
"q0", "q1", "q2", "q3",
"q8", "q10"
);
+#endif
}
template <>
@@ -110,6 +588,10 @@
const int16_t* sN,
const int32_t* const volumeLR)
{
+#ifdef USE_INTRINSIC
+ ProcessNeonIntrinsic<2, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ 0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
+#else
const int CHANNELS = 2; // template specialization does not preserve params
const int STRIDE = 16;
sP -= CHANNELS*((STRIDE>>1)-1);
@@ -119,13 +601,13 @@
"1: \n"
- "vld2.16 {q2, q3}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo samples
- "vld2.16 {q5, q6}, [%[sN]]! \n"// (3) load 8 16-bits stereo samples
+ "vld2.16 {q2, q3}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo frames
+ "vld2.16 {q5, q6}, [%[sN]]! \n"// (3) load 8 16-bits stereo frames
"vld1.16 {q8}, [%[coefsP0]:128]! \n"// (1) load 8 16-bits coefs
"vld1.16 {q10}, [%[coefsN0]:128]! \n"// (1) load 8 16-bits coefs
- "vrev64.16 q2, q2 \n"// (1) reverse 8 frames of the left positive
- "vrev64.16 q3, q3 \n"// (0 combines+) reverse right positive
+ "vrev64.16 q2, q2 \n"// (1) reverse 8 samples of positive left
+ "vrev64.16 q3, q3 \n"// (0 combines+) reverse positive right
"vmlal.s16 q0, d4, d17 \n"// (1) multiply (reversed) samples left
"vmlal.s16 q0, d5, d16 \n"// (1) multiply (reversed) samples left
@@ -157,6 +639,7 @@
"q4", "q5", "q6",
"q8", "q10"
);
+#endif
}
template <>
@@ -171,6 +654,11 @@
uint32_t lerpP,
const int32_t* const volumeLR)
{
+#ifdef USE_INTRINSIC
+ ProcessNeonIntrinsic<1, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ lerpP, coefsP1, coefsN1);
+#else
+
const int CHANNELS = 1; // template specialization does not preserve params
const int STRIDE = 16;
sP -= CHANNELS*((STRIDE>>1)-1);
@@ -227,6 +715,7 @@
"q0", "q1", "q2", "q3",
"q8", "q9", "q10", "q11"
);
+#endif
}
template <>
@@ -241,6 +730,10 @@
uint32_t lerpP,
const int32_t* const volumeLR)
{
+#ifdef USE_INTRINSIC
+ ProcessNeonIntrinsic<2, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ lerpP, coefsP1, coefsN1);
+#else
const int CHANNELS = 2; // template specialization does not preserve params
const int STRIDE = 16;
sP -= CHANNELS*((STRIDE>>1)-1);
@@ -251,8 +744,8 @@
"1: \n"
- "vld2.16 {q2, q3}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo samples
- "vld2.16 {q5, q6}, [%[sN]]! \n"// (3) load 8 16-bits stereo samples
+ "vld2.16 {q2, q3}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo frames
+ "vld2.16 {q5, q6}, [%[sN]]! \n"// (3) load 8 16-bits stereo frames
"vld1.16 {q8}, [%[coefsP0]:128]! \n"// (1) load 8 16-bits coefs
"vld1.16 {q9}, [%[coefsP1]:128]! \n"// (1) load 8 16-bits coefs for interpolation
"vld1.16 {q10}, [%[coefsN1]:128]! \n"// (1) load 8 16-bits coefs
@@ -264,8 +757,8 @@
"vqrdmulh.s16 q9, q9, d2[0] \n"// (2) interpolate (step2) 1st set of coefs
"vqrdmulh.s16 q11, q11, d2[0] \n"// (2) interpolate (step2) 2nd set of coefs
- "vrev64.16 q2, q2 \n"// (1) reverse 8 frames of the left positive
- "vrev64.16 q3, q3 \n"// (1) reverse 8 frames of the right positive
+ "vrev64.16 q2, q2 \n"// (1) reverse 8 samples of positive left
+ "vrev64.16 q3, q3 \n"// (1) reverse 8 samples of positive right
"vadd.s16 q8, q8, q9 \n"// (1+1d) interpolate (step3) 1st set
"vadd.s16 q10, q10, q11 \n"// (1+1d) interpolate (step3) 2nd set
@@ -303,6 +796,7 @@
"q4", "q5", "q6",
"q8", "q9", "q10", "q11"
);
+#endif
}
template <>
@@ -314,6 +808,10 @@
const int16_t* sN,
const int32_t* const volumeLR)
{
+#ifdef USE_INTRINSIC
+ ProcessNeonIntrinsic<1, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ 0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
+#else
const int CHANNELS = 1; // template specialization does not preserve params
const int STRIDE = 16;
sP -= CHANNELS*((STRIDE>>1)-1);
@@ -327,7 +825,7 @@
"vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 8 32-bits coefs
"vld1.32 {q10, q11}, [%[coefsN0]:128]! \n"// load 8 32-bits coefs
- "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side
+ "vrev64.16 q2, q2 \n"// reverse 8 samples of the positive side
"vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
"vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
@@ -335,10 +833,10 @@
"vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits
"vshll.s16 q15, d7, #15 \n"// extend samples to 31 bits
- "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples
+ "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples
+ "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples
"vadd.s32 q0, q0, q12 \n"// accumulate result
"vadd.s32 q13, q13, q14 \n"// accumulate result
@@ -364,6 +862,7 @@
"q8", "q9", "q10", "q11",
"q12", "q13", "q14", "q15"
);
+#endif
}
template <>
@@ -375,6 +874,10 @@
const int16_t* sN,
const int32_t* const volumeLR)
{
+#ifdef USE_INTRINSIC
+ ProcessNeonIntrinsic<2, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ 0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
+#else
const int CHANNELS = 2; // template specialization does not preserve params
const int STRIDE = 16;
sP -= CHANNELS*((STRIDE>>1)-1);
@@ -384,13 +887,13 @@
"1: \n"
- "vld2.16 {q2, q3}, [%[sP]] \n"// load 4 16-bits stereo samples
- "vld2.16 {q5, q6}, [%[sN]]! \n"// load 4 16-bits stereo samples
- "vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 4 32-bits coefs
- "vld1.32 {q10, q11}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs
+ "vld2.16 {q2, q3}, [%[sP]] \n"// load 8 16-bits stereo frames
+ "vld2.16 {q5, q6}, [%[sN]]! \n"// load 8 16-bits stereo frames
+ "vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 8 32-bits coefs
+ "vld1.32 {q10, q11}, [%[coefsN0]:128]! \n"// load 8 32-bits coefs
- "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side
- "vrev64.16 q3, q3 \n"// reverse 8 frames of the positive side
+ "vrev64.16 q2, q2 \n"// reverse 8 samples of positive left
+ "vrev64.16 q3, q3 \n"// reverse 8 samples of positive right
"vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
"vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
@@ -398,15 +901,15 @@
"vshll.s16 q14, d10, #15 \n"// extend samples to 31 bits
"vshll.s16 q15, d11, #15 \n"// extend samples to 31 bits
- "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by coef
+ "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by coef
+ "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by coef
"vadd.s32 q0, q0, q12 \n"// accumulate result
"vadd.s32 q13, q13, q14 \n"// accumulate result
- "vadd.s32 q0, q0, q15 \n"// (+1) accumulate result
- "vadd.s32 q0, q0, q13 \n"// (+1) accumulate result
+ "vadd.s32 q0, q0, q15 \n"// accumulate result
+ "vadd.s32 q0, q0, q13 \n"// accumulate result
"vshll.s16 q12, d6, #15 \n"// extend samples to 31 bits
"vshll.s16 q13, d7, #15 \n"// extend samples to 31 bits
@@ -414,15 +917,15 @@
"vshll.s16 q14, d12, #15 \n"// extend samples to 31 bits
"vshll.s16 q15, d13, #15 \n"// extend samples to 31 bits
- "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef
+ "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by coef
+ "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by coef
+ "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by coef
+ "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by coef
"vadd.s32 q4, q4, q12 \n"// accumulate result
"vadd.s32 q13, q13, q14 \n"// accumulate result
- "vadd.s32 q4, q4, q15 \n"// (+1) accumulate result
- "vadd.s32 q4, q4, q13 \n"// (+1) accumulate result
+ "vadd.s32 q4, q4, q15 \n"// accumulate result
+ "vadd.s32 q4, q4, q13 \n"// accumulate result
"subs %[count], %[count], #8 \n"// update loop counter
"sub %[sP], %[sP], #32 \n"// move pointer to next set of samples
@@ -444,6 +947,7 @@
"q8", "q9", "q10", "q11",
"q12", "q13", "q14", "q15"
);
+#endif
}
template <>
@@ -458,6 +962,10 @@
uint32_t lerpP,
const int32_t* const volumeLR)
{
+#ifdef USE_INTRINSIC
+ ProcessNeonIntrinsic<1, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ lerpP, coefsP1, coefsN1);
+#else
const int CHANNELS = 1; // template specialization does not preserve params
const int STRIDE = 16;
sP -= CHANNELS*((STRIDE>>1)-1);
@@ -489,7 +997,7 @@
"vadd.s32 q10, q10, q14 \n"// interpolate (step3)
"vadd.s32 q11, q11, q15 \n"// interpolate (step3)
- "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side
+ "vrev64.16 q2, q2 \n"// reverse 8 samples of the positive side
"vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
"vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
@@ -529,6 +1037,7 @@
"q8", "q9", "q10", "q11",
"q12", "q13", "q14", "q15"
);
+#endif
}
template <>
@@ -543,6 +1052,10 @@
uint32_t lerpP,
const int32_t* const volumeLR)
{
+#ifdef USE_INTRINSIC
+ ProcessNeonIntrinsic<2, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ lerpP, coefsP1, coefsN1);
+#else
const int CHANNELS = 2; // template specialization does not preserve params
const int STRIDE = 16;
sP -= CHANNELS*((STRIDE>>1)-1);
@@ -553,8 +1066,8 @@
"1: \n"
- "vld2.16 {q2, q3}, [%[sP]] \n"// load 4 16-bits stereo samples
- "vld2.16 {q5, q6}, [%[sN]]! \n"// load 4 16-bits stereo samples
+ "vld2.16 {q2, q3}, [%[sP]] \n"// load 8 16-bits stereo frames
+ "vld2.16 {q5, q6}, [%[sN]]! \n"// load 8 16-bits stereo frames
"vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 8 32-bits coefs
"vld1.32 {q12, q13}, [%[coefsP1]:128]! \n"// load 8 32-bits coefs
"vld1.32 {q10, q11}, [%[coefsN1]:128]! \n"// load 8 32-bits coefs
@@ -575,8 +1088,8 @@
"vadd.s32 q10, q10, q14 \n"// interpolate (step3)
"vadd.s32 q11, q11, q15 \n"// interpolate (step3)
- "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side
- "vrev64.16 q3, q3 \n"// reverse 8 frames of the positive side
+ "vrev64.16 q2, q2 \n"// reverse 8 samples of positive left
+ "vrev64.16 q3, q3 \n"// reverse 8 samples of positive right
"vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
"vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
@@ -591,8 +1104,8 @@
"vadd.s32 q0, q0, q12 \n"// accumulate result
"vadd.s32 q13, q13, q14 \n"// accumulate result
- "vadd.s32 q0, q0, q15 \n"// (+1) accumulate result
- "vadd.s32 q0, q0, q13 \n"// (+1) accumulate result
+ "vadd.s32 q0, q0, q15 \n"// accumulate result
+ "vadd.s32 q0, q0, q13 \n"// accumulate result
"vshll.s16 q12, d6, #15 \n"// extend samples to 31 bits
"vshll.s16 q13, d7, #15 \n"// extend samples to 31 bits
@@ -607,8 +1120,8 @@
"vadd.s32 q4, q4, q12 \n"// accumulate result
"vadd.s32 q13, q13, q14 \n"// accumulate result
- "vadd.s32 q4, q4, q15 \n"// (+1) accumulate result
- "vadd.s32 q4, q4, q13 \n"// (+1) accumulate result
+ "vadd.s32 q4, q4, q15 \n"// accumulate result
+ "vadd.s32 q4, q4, q13 \n"// accumulate result
"subs %[count], %[count], #8 \n"// update loop counter
"sub %[sP], %[sP], #32 \n"// move pointer to next set of samples
@@ -633,517 +1146,69 @@
"q8", "q9", "q10", "q11",
"q12", "q13", "q14", "q15"
);
+#endif
}
-template <>
-inline void ProcessL<1, 8>(int32_t* const out,
+template<>
+inline void ProcessL<1, 16>(float* const out,
int count,
- const int16_t* coefsP,
- const int16_t* coefsN,
- const int16_t* sP,
- const int16_t* sN,
- const int32_t* const volumeLR)
+ const float* coefsP,
+ const float* coefsN,
+ const float* sP,
+ const float* sN,
+ const float* const volumeLR)
{
- const int CHANNELS = 1; // template specialization does not preserve params
- const int STRIDE = 8;
- sP -= CHANNELS*((STRIDE>>1)-1);
- asm (
- "veor q0, q0, q0 \n"// (0 - combines+) accumulator = 0
-
- "1: \n"
-
- "vld1.16 {d4}, [%[sP]] \n"// (2+0d) load 4 16-bits mono samples
- "vld1.16 {d6}, [%[sN]]! \n"// (2) load 4 16-bits mono samples
- "vld1.16 {d16}, [%[coefsP0]:64]! \n"// (1) load 4 16-bits coefs
- "vld1.16 {d20}, [%[coefsN0]:64]! \n"// (1) load 4 16-bits coefs
-
- "vrev64.16 d4, d4 \n"// (1) reversed s3, s2, s1, s0, s7, s6, s5, s4
-
- // reordering the vmal to do d6, d7 before d4, d5 is slower(?)
- "vmlal.s16 q0, d4, d16 \n"// (1) multiply (reversed)samples by coef
- "vmlal.s16 q0, d6, d20 \n"// (1) multiply neg samples
-
- // moving these ARM instructions before neon above seems to be slower
- "subs %[count], %[count], #4 \n"// (1) update loop counter
- "sub %[sP], %[sP], #8 \n"// (0) move pointer to next set of samples
-
- // sP used after branch (warning)
- "bne 1b \n"// loop
-
- ASSEMBLY_ACCUMULATE_MONO
-
- : [out] "=Uv" (out[0]),
- [count] "+r" (count),
- [coefsP0] "+r" (coefsP),
- [coefsN0] "+r" (coefsN),
- [sP] "+r" (sP),
- [sN] "+r" (sN)
- : [vLR] "r" (volumeLR)
- : "cc", "memory",
- "q0", "q1", "q2", "q3",
- "q8", "q10"
- );
+ ProcessNeonIntrinsic<1, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ 0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
}
-template <>
-inline void ProcessL<2, 8>(int32_t* const out,
+template<>
+inline void ProcessL<2, 16>(float* const out,
int count,
- const int16_t* coefsP,
- const int16_t* coefsN,
- const int16_t* sP,
- const int16_t* sN,
- const int32_t* const volumeLR)
+ const float* coefsP,
+ const float* coefsN,
+ const float* sP,
+ const float* sN,
+ const float* const volumeLR)
{
- const int CHANNELS = 2; // template specialization does not preserve params
- const int STRIDE = 8;
- sP -= CHANNELS*((STRIDE>>1)-1);
- asm (
- "veor q0, q0, q0 \n"// (1) acc_L = 0
- "veor q4, q4, q4 \n"// (0 combines+) acc_R = 0
-
- "1: \n"
-
- "vld2.16 {d4, d5}, [%[sP]] \n"// (2+0d) load 8 16-bits stereo samples
- "vld2.16 {d6, d7}, [%[sN]]! \n"// (2) load 8 16-bits stereo samples
- "vld1.16 {d16}, [%[coefsP0]:64]! \n"// (1) load 8 16-bits coefs
- "vld1.16 {d20}, [%[coefsN0]:64]! \n"// (1) load 8 16-bits coefs
-
- "vrev64.16 q2, q2 \n"// (1) reverse 8 frames of the left positive
-
- "vmlal.s16 q0, d4, d16 \n"// (1) multiply (reversed) samples left
- "vmlal.s16 q4, d5, d16 \n"// (1) multiply (reversed) samples right
- "vmlal.s16 q0, d6, d20 \n"// (1) multiply samples left
- "vmlal.s16 q4, d7, d20 \n"// (1) multiply samples right
-
- // moving these ARM before neon seems to be slower
- "subs %[count], %[count], #4 \n"// (1) update loop counter
- "sub %[sP], %[sP], #16 \n"// (0) move pointer to next set of samples
-
- // sP used after branch (warning)
- "bne 1b \n"// loop
-
- ASSEMBLY_ACCUMULATE_STEREO
-
- : [out] "=Uv" (out[0]),
- [count] "+r" (count),
- [coefsP0] "+r" (coefsP),
- [coefsN0] "+r" (coefsN),
- [sP] "+r" (sP),
- [sN] "+r" (sN)
- : [vLR] "r" (volumeLR)
- : "cc", "memory",
- "q0", "q1", "q2", "q3",
- "q4", "q5", "q6",
- "q8", "q10"
- );
+ ProcessNeonIntrinsic<2, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ 0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
}
-template <>
-inline void Process<1, 8>(int32_t* const out,
+template<>
+inline void Process<1, 16>(float* const out,
int count,
- const int16_t* coefsP,
- const int16_t* coefsN,
- const int16_t* coefsP1,
- const int16_t* coefsN1,
- const int16_t* sP,
- const int16_t* sN,
- uint32_t lerpP,
- const int32_t* const volumeLR)
+ const float* coefsP,
+ const float* coefsN,
+ const float* coefsP1,
+ const float* coefsN1,
+ const float* sP,
+ const float* sN,
+ float lerpP,
+ const float* const volumeLR)
{
- const int CHANNELS = 1; // template specialization does not preserve params
- const int STRIDE = 8;
- sP -= CHANNELS*((STRIDE>>1)-1);
- asm (
- "vmov.32 d2[0], %[lerpP] \n"// load the positive phase S32 Q15
- "veor q0, q0, q0 \n"// (0 - combines+) accumulator = 0
-
- "1: \n"
-
- "vld1.16 {d4}, [%[sP]] \n"// (2+0d) load 4 16-bits mono samples
- "vld1.16 {d6}, [%[sN]]! \n"// (2) load 4 16-bits mono samples
- "vld1.16 {d16}, [%[coefsP0]:64]! \n"// (1) load 4 16-bits coefs
- "vld1.16 {d17}, [%[coefsP1]:64]! \n"// (1) load 4 16-bits coefs for interpolation
- "vld1.16 {d20}, [%[coefsN1]:64]! \n"// (1) load 4 16-bits coefs
- "vld1.16 {d21}, [%[coefsN0]:64]! \n"// (1) load 4 16-bits coefs for interpolation
-
- "vsub.s16 d17, d17, d16 \n"// (1) interpolate (step1) 1st set of coefs
- "vsub.s16 d21, d21, d20 \n"// (1) interpolate (step1) 2nd set of coets
-
- "vqrdmulh.s16 d17, d17, d2[0] \n"// (2) interpolate (step2) 1st set of coefs
- "vqrdmulh.s16 d21, d21, d2[0] \n"// (2) interpolate (step2) 2nd set of coefs
-
- "vrev64.16 d4, d4 \n"// (1) reverse s3, s2, s1, s0, s7, s6, s5, s4
-
- "vadd.s16 d16, d16, d17 \n"// (1+2d) interpolate (step3) 1st set
- "vadd.s16 d20, d20, d21 \n"// (1+1d) interpolate (step3) 2nd set
-
- // reordering the vmal to do d6, d7 before d4, d5 is slower(?)
- "vmlal.s16 q0, d4, d16 \n"// (1+0d) multiply (reversed)by coef
- "vmlal.s16 q0, d6, d20 \n"// (1) multiply neg samples
-
- // moving these ARM instructions before neon above seems to be slower
- "subs %[count], %[count], #4 \n"// (1) update loop counter
- "sub %[sP], %[sP], #8 \n"// move pointer to next set of samples
-
- // sP used after branch (warning)
- "bne 1b \n"// loop
-
- ASSEMBLY_ACCUMULATE_MONO
-
- : [out] "=Uv" (out[0]),
- [count] "+r" (count),
- [coefsP0] "+r" (coefsP),
- [coefsN0] "+r" (coefsN),
- [coefsP1] "+r" (coefsP1),
- [coefsN1] "+r" (coefsN1),
- [sP] "+r" (sP),
- [sN] "+r" (sN)
- : [lerpP] "r" (lerpP),
- [vLR] "r" (volumeLR)
- : "cc", "memory",
- "q0", "q1", "q2", "q3",
- "q8", "q9", "q10", "q11"
- );
+ ProcessNeonIntrinsic<1, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ lerpP, coefsP1, coefsN1);
}
-template <>
-inline void Process<2, 8>(int32_t* const out,
+template<>
+inline void Process<2, 16>(float* const out,
int count,
- const int16_t* coefsP,
- const int16_t* coefsN,
- const int16_t* coefsP1,
- const int16_t* coefsN1,
- const int16_t* sP,
- const int16_t* sN,
- uint32_t lerpP,
- const int32_t* const volumeLR)
+ const float* coefsP,
+ const float* coefsN,
+ const float* coefsP1,
+ const float* coefsN1,
+ const float* sP,
+ const float* sN,
+ float lerpP,
+ const float* const volumeLR)
{
- const int CHANNELS = 2; // template specialization does not preserve params
- const int STRIDE = 8;
- sP -= CHANNELS*((STRIDE>>1)-1);
- asm (
- "vmov.32 d2[0], %[lerpP] \n"// load the positive phase
- "veor q0, q0, q0 \n"// (1) acc_L = 0
- "veor q4, q4, q4 \n"// (0 combines+) acc_R = 0
-
- "1: \n"
-
- "vld2.16 {d4, d5}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo samples
- "vld2.16 {d6, d7}, [%[sN]]! \n"// (3) load 8 16-bits stereo samples
- "vld1.16 {d16}, [%[coefsP0]:64]! \n"// (1) load 8 16-bits coefs
- "vld1.16 {d17}, [%[coefsP1]:64]! \n"// (1) load 8 16-bits coefs for interpolation
- "vld1.16 {d20}, [%[coefsN1]:64]! \n"// (1) load 8 16-bits coefs
- "vld1.16 {d21}, [%[coefsN0]:64]! \n"// (1) load 8 16-bits coefs for interpolation
-
- "vsub.s16 d17, d17, d16 \n"// (1) interpolate (step1) 1st set of coefs
- "vsub.s16 d21, d21, d20 \n"// (1) interpolate (step1) 2nd set of coets
-
- "vqrdmulh.s16 d17, d17, d2[0] \n"// (2) interpolate (step2) 1st set of coefs
- "vqrdmulh.s16 d21, d21, d2[0] \n"// (2) interpolate (step2) 2nd set of coefs
-
- "vrev64.16 q2, q2 \n"// (1) reverse 8 frames of the left positive
-
- "vadd.s16 d16, d16, d17 \n"// (1+1d) interpolate (step3) 1st set
- "vadd.s16 d20, d20, d21 \n"// (1+1d) interpolate (step3) 2nd set
-
- "vmlal.s16 q0, d4, d16 \n"// (1) multiply (reversed) samples left
- "vmlal.s16 q4, d5, d16 \n"// (1) multiply (reversed) samples right
- "vmlal.s16 q0, d6, d20 \n"// (1) multiply samples left
- "vmlal.s16 q4, d7, d20 \n"// (1) multiply samples right
-
- // moving these ARM before neon seems to be slower
- "subs %[count], %[count], #4 \n"// (1) update loop counter
- "sub %[sP], %[sP], #16 \n"// move pointer to next set of samples
-
- // sP used after branch (warning)
- "bne 1b \n"// loop
-
- ASSEMBLY_ACCUMULATE_STEREO
-
- : [out] "=Uv" (out[0]),
- [count] "+r" (count),
- [coefsP0] "+r" (coefsP),
- [coefsN0] "+r" (coefsN),
- [coefsP1] "+r" (coefsP1),
- [coefsN1] "+r" (coefsN1),
- [sP] "+r" (sP),
- [sN] "+r" (sN)
- : [lerpP] "r" (lerpP),
- [vLR] "r" (volumeLR)
- : "cc", "memory",
- "q0", "q1", "q2", "q3",
- "q4", "q5", "q6",
- "q8", "q9", "q10", "q11"
- );
-}
-
-template <>
-inline void ProcessL<1, 8>(int32_t* const out,
- int count,
- const int32_t* coefsP,
- const int32_t* coefsN,
- const int16_t* sP,
- const int16_t* sN,
- const int32_t* const volumeLR)
-{
- const int CHANNELS = 1; // template specialization does not preserve params
- const int STRIDE = 8;
- sP -= CHANNELS*((STRIDE>>1)-1);
- asm (
- "veor q0, q0, q0 \n"// result, initialize to 0
-
- "1: \n"
-
- "vld1.16 {d4}, [%[sP]] \n"// load 4 16-bits mono samples
- "vld1.16 {d6}, [%[sN]]! \n"// load 4 16-bits mono samples
- "vld1.32 {q8}, [%[coefsP0]:128]! \n"// load 4 32-bits coefs
- "vld1.32 {q10}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs
-
- "vrev64.16 d4, d4 \n"// reverse 2 frames of the positive side
-
- "vshll.s16 q12, d4, #15 \n"// (stall) extend samples to 31 bits
- "vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits
-
- "vqrdmulh.s32 q12, q12, q8 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
-
- "vadd.s32 q0, q0, q12 \n"// accumulate result
- "vadd.s32 q0, q0, q14 \n"// (stall) accumulate result
-
- "subs %[count], %[count], #4 \n"// update loop counter
- "sub %[sP], %[sP], #8 \n"// move pointer to next set of samples
-
- "bne 1b \n"// loop
-
- ASSEMBLY_ACCUMULATE_MONO
-
- : [out] "=Uv" (out[0]),
- [count] "+r" (count),
- [coefsP0] "+r" (coefsP),
- [coefsN0] "+r" (coefsN),
- [sP] "+r" (sP),
- [sN] "+r" (sN)
- : [vLR] "r" (volumeLR)
- : "cc", "memory",
- "q0", "q1", "q2", "q3",
- "q8", "q9", "q10", "q11",
- "q12", "q14"
- );
-}
-
-template <>
-inline void ProcessL<2, 8>(int32_t* const out,
- int count,
- const int32_t* coefsP,
- const int32_t* coefsN,
- const int16_t* sP,
- const int16_t* sN,
- const int32_t* const volumeLR)
-{
- const int CHANNELS = 2; // template specialization does not preserve params
- const int STRIDE = 8;
- sP -= CHANNELS*((STRIDE>>1)-1);
- asm (
- "veor q0, q0, q0 \n"// result, initialize to 0
- "veor q4, q4, q4 \n"// result, initialize to 0
-
- "1: \n"
-
- "vld2.16 {d4, d5}, [%[sP]] \n"// load 4 16-bits stereo samples
- "vld2.16 {d6, d7}, [%[sN]]! \n"// load 4 16-bits stereo samples
- "vld1.32 {q8}, [%[coefsP0]:128]! \n"// load 4 32-bits coefs
- "vld1.32 {q10}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs
-
- "vrev64.16 q2, q2 \n"// reverse 2 frames of the positive side
-
- "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
- "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
-
- "vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits
- "vshll.s16 q15, d7, #15 \n"// extend samples to 31 bits
-
- "vqrdmulh.s32 q12, q12, q8 \n"// multiply samples by coef
- "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by coef
- "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by coef
- "vqrdmulh.s32 q15, q15, q10 \n"// multiply samples by coef
-
- "vadd.s32 q0, q0, q12 \n"// accumulate result
- "vadd.s32 q4, q4, q13 \n"// accumulate result
- "vadd.s32 q0, q0, q14 \n"// accumulate result
- "vadd.s32 q4, q4, q15 \n"// accumulate result
-
- "subs %[count], %[count], #4 \n"// update loop counter
- "sub %[sP], %[sP], #16 \n"// move pointer to next set of samples
-
- "bne 1b \n"// loop
-
- ASSEMBLY_ACCUMULATE_STEREO
-
- : [out] "=Uv" (out[0]),
- [count] "+r" (count),
- [coefsP0] "+r" (coefsP),
- [coefsN0] "+r" (coefsN),
- [sP] "+r" (sP),
- [sN] "+r" (sN)
- : [vLR] "r" (volumeLR)
- : "cc", "memory",
- "q0", "q1", "q2", "q3", "q4",
- "q8", "q9", "q10", "q11",
- "q12", "q13", "q14", "q15"
- );
-}
-
-template <>
-inline void Process<1, 8>(int32_t* const out,
- int count,
- const int32_t* coefsP,
- const int32_t* coefsN,
- const int32_t* coefsP1,
- const int32_t* coefsN1,
- const int16_t* sP,
- const int16_t* sN,
- uint32_t lerpP,
- const int32_t* const volumeLR)
-{
- const int CHANNELS = 1; // template specialization does not preserve params
- const int STRIDE = 8;
- sP -= CHANNELS*((STRIDE>>1)-1);
- asm (
- "vmov.32 d2[0], %[lerpP] \n"// load the positive phase
- "veor q0, q0, q0 \n"// result, initialize to 0
-
- "1: \n"
-
- "vld1.16 {d4}, [%[sP]] \n"// load 4 16-bits mono samples
- "vld1.16 {d6}, [%[sN]]! \n"// load 4 16-bits mono samples
- "vld1.32 {q8}, [%[coefsP0]:128]! \n"// load 4 32-bits coefs
- "vld1.32 {q9}, [%[coefsP1]:128]! \n"// load 4 32-bits coefs for interpolation
- "vld1.32 {q10}, [%[coefsN1]:128]! \n"// load 4 32-bits coefs
- "vld1.32 {q11}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs for interpolation
-
- "vrev64.16 d4, d4 \n"// reverse 2 frames of the positive side
-
- "vsub.s32 q9, q9, q8 \n"// interpolate (step1) 1st set of coefs
- "vsub.s32 q11, q11, q10 \n"// interpolate (step1) 2nd set of coets
- "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
-
- "vqrdmulh.s32 q9, q9, d2[0] \n"// interpolate (step2) 1st set of coefs
- "vqrdmulh.s32 q11, q11, d2[0] \n"// interpolate (step2) 2nd set of coefs
- "vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits
-
- "vadd.s32 q8, q8, q9 \n"// interpolate (step3) 1st set
- "vadd.s32 q10, q10, q11 \n"// interpolate (step4) 2nd set
-
- "vqrdmulh.s32 q12, q12, q8 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
-
- "vadd.s32 q0, q0, q12 \n"// accumulate result
- "vadd.s32 q0, q0, q14 \n"// accumulate result
-
- "subs %[count], %[count], #4 \n"// update loop counter
- "sub %[sP], %[sP], #8 \n"// move pointer to next set of samples
-
- "bne 1b \n"// loop
-
- ASSEMBLY_ACCUMULATE_MONO
-
- : [out] "=Uv" (out[0]),
- [count] "+r" (count),
- [coefsP0] "+r" (coefsP),
- [coefsP1] "+r" (coefsP1),
- [coefsN0] "+r" (coefsN),
- [coefsN1] "+r" (coefsN1),
- [sP] "+r" (sP),
- [sN] "+r" (sN)
- : [lerpP] "r" (lerpP),
- [vLR] "r" (volumeLR)
- : "cc", "memory",
- "q0", "q1", "q2", "q3",
- "q8", "q9", "q10", "q11",
- "q12", "q14"
- );
-}
-
-template <>
-inline
-void Process<2, 8>(int32_t* const out,
- int count,
- const int32_t* coefsP,
- const int32_t* coefsN,
- const int32_t* coefsP1,
- const int32_t* coefsN1,
- const int16_t* sP,
- const int16_t* sN,
- uint32_t lerpP,
- const int32_t* const volumeLR)
-{
- const int CHANNELS = 2; // template specialization does not preserve params
- const int STRIDE = 8;
- sP -= CHANNELS*((STRIDE>>1)-1);
- asm (
- "vmov.32 d2[0], %[lerpP] \n"// load the positive phase
- "veor q0, q0, q0 \n"// result, initialize to 0
- "veor q4, q4, q4 \n"// result, initialize to 0
-
- "1: \n"
- "vld2.16 {d4, d5}, [%[sP]] \n"// load 4 16-bits stereo samples
- "vld2.16 {d6, d7}, [%[sN]]! \n"// load 4 16-bits stereo samples
- "vld1.32 {q8}, [%[coefsP0]:128]! \n"// load 4 32-bits coefs
- "vld1.32 {q9}, [%[coefsP1]:128]! \n"// load 4 32-bits coefs for interpolation
- "vld1.32 {q10}, [%[coefsN1]:128]! \n"// load 4 32-bits coefs
- "vld1.32 {q11}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs for interpolation
-
- "vrev64.16 q2, q2 \n"// (reversed) 2 frames of the positive side
-
- "vsub.s32 q9, q9, q8 \n"// interpolate (step1) 1st set of coefs
- "vsub.s32 q11, q11, q10 \n"// interpolate (step1) 2nd set of coets
- "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits
- "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits
-
- "vqrdmulh.s32 q9, q9, d2[0] \n"// interpolate (step2) 1st set of coefs
- "vqrdmulh.s32 q11, q11, d2[1] \n"// interpolate (step3) 2nd set of coefs
- "vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits
- "vshll.s16 q15, d7, #15 \n"// extend samples to 31 bits
-
- "vadd.s32 q8, q8, q9 \n"// interpolate (step3) 1st set
- "vadd.s32 q10, q10, q11 \n"// interpolate (step4) 2nd set
-
- "vqrdmulh.s32 q12, q12, q8 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef
- "vqrdmulh.s32 q15, q15, q10 \n"// multiply samples by interpolated coef
-
- "vadd.s32 q0, q0, q12 \n"// accumulate result
- "vadd.s32 q4, q4, q13 \n"// accumulate result
- "vadd.s32 q0, q0, q14 \n"// accumulate result
- "vadd.s32 q4, q4, q15 \n"// accumulate result
-
- "subs %[count], %[count], #4 \n"// update loop counter
- "sub %[sP], %[sP], #16 \n"// move pointer to next set of samples
-
- "bne 1b \n"// loop
-
- ASSEMBLY_ACCUMULATE_STEREO
-
- : [out] "=Uv" (out[0]),
- [count] "+r" (count),
- [coefsP0] "+r" (coefsP),
- [coefsP1] "+r" (coefsP1),
- [coefsN0] "+r" (coefsN),
- [coefsN1] "+r" (coefsN1),
- [sP] "+r" (sP),
- [sN] "+r" (sN)
- : [lerpP] "r" (lerpP),
- [vLR] "r" (volumeLR)
- : "cc", "memory",
- "q0", "q1", "q2", "q3", "q4",
- "q8", "q9", "q10", "q11",
- "q12", "q13", "q14", "q15"
- );
+ ProcessNeonIntrinsic<2, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+ lerpP, coefsP1, coefsN1);
}
#endif //USE_NEON
-}; // namespace android
+} // namespace android
#endif /*ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_NEON_H*/
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index e6fb76c..ba9a356 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -61,135 +61,7 @@
* cmd-line: fir -l 7 -s 48000 -c 20478
*/
const uint32_t AudioResamplerSinc::mFirCoefsUp[] __attribute__ ((aligned (32))) = {
- 0x6d374bc7, 0x111c6ba0, 0xf3240e61, 0x07d14a38, 0xfc509e64, 0x0139cee9, 0xffc8c866, 0xfffcc300,
- 0x6d35278a, 0x103e8192, 0xf36b9dfd, 0x07bdfaa5, 0xfc5102d0, 0x013d618d, 0xffc663b9, 0xfffd9592,
- 0x6d2ebafe, 0x0f62811a, 0xf3b3d8ac, 0x07a9f399, 0xfc51d9a6, 0x0140bea5, 0xffc41212, 0xfffe631e,
- 0x6d24069d, 0x0e8875ad, 0xf3fcb43e, 0x07953976, 0xfc53216f, 0x0143e67c, 0xffc1d373, 0xffff2b9f,
- 0x6d150b35, 0x0db06a89, 0xf4462690, 0x077fd0ac, 0xfc54d8ae, 0x0146d965, 0xffbfa7d9, 0xffffef10,
- 0x6d01c9e3, 0x0cda6ab5, 0xf4902587, 0x0769bdaf, 0xfc56fdda, 0x014997bb, 0xffbd8f40, 0x0000ad6e,
- 0x6cea4418, 0x0c0680fe, 0xf4daa718, 0x07530501, 0xfc598f60, 0x014c21db, 0xffbb89a1, 0x000166b6,
- 0x6cce7b97, 0x0b34b7f5, 0xf525a143, 0x073bab28, 0xfc5c8ba5, 0x014e782a, 0xffb996f3, 0x00021ae5,
- 0x6cae7272, 0x0a6519f4, 0xf5710a17, 0x0723b4b4, 0xfc5ff105, 0x01509b14, 0xffb7b728, 0x0002c9fd,
- 0x6c8a2b0f, 0x0997b116, 0xf5bcd7b1, 0x070b2639, 0xfc63bdd3, 0x01528b08, 0xffb5ea31, 0x000373fb,
- 0x6c61a823, 0x08cc873c, 0xf609003f, 0x06f20453, 0xfc67f05a, 0x0154487b, 0xffb42ffc, 0x000418e2,
- 0x6c34ecb5, 0x0803a60a, 0xf6557a00, 0x06d853a2, 0xfc6c86dd, 0x0155d3e8, 0xffb28876, 0x0004b8b3,
- 0x6c03fc1c, 0x073d16e7, 0xf6a23b44, 0x06be18cd, 0xfc717f97, 0x01572dcf, 0xffb0f388, 0x00055371,
- 0x6bced9ff, 0x0678e2fc, 0xf6ef3a6e, 0x06a3587e, 0xfc76d8bc, 0x015856b6, 0xffaf7118, 0x0005e921,
- 0x6b958a54, 0x05b71332, 0xf73c6df4, 0x06881761, 0xfc7c9079, 0x01594f25, 0xffae010b, 0x000679c5,
- 0x6b581163, 0x04f7b037, 0xf789cc61, 0x066c5a27, 0xfc82a4f4, 0x015a17ab, 0xffaca344, 0x00070564,
- 0x6b1673c1, 0x043ac276, 0xf7d74c53, 0x06502583, 0xfc89144d, 0x015ab0db, 0xffab57a1, 0x00078c04,
- 0x6ad0b652, 0x0380521c, 0xf824e480, 0x06337e2a, 0xfc8fdc9f, 0x015b1b4e, 0xffaa1e02, 0x00080dab,
- 0x6a86de48, 0x02c86715, 0xf8728bb3, 0x061668d2, 0xfc96fbfc, 0x015b579e, 0xffa8f641, 0x00088a62,
- 0x6a38f123, 0x0213090c, 0xf8c038d0, 0x05f8ea30, 0xfc9e7074, 0x015b666c, 0xffa7e039, 0x00090230,
- 0x69e6f4b1, 0x01603f6e, 0xf90de2d1, 0x05db06fc, 0xfca63810, 0x015b485b, 0xffa6dbc0, 0x0009751e,
- 0x6990ef0b, 0x00b01162, 0xf95b80cb, 0x05bcc3ed, 0xfcae50d6, 0x015afe14, 0xffa5e8ad, 0x0009e337,
- 0x6936e697, 0x000285d0, 0xf9a909ea, 0x059e25b5, 0xfcb6b8c4, 0x015a8843, 0xffa506d2, 0x000a4c85,
- 0x68d8e206, 0xff57a35e, 0xf9f67577, 0x057f310a, 0xfcbf6dd8, 0x0159e796, 0xffa43603, 0x000ab112,
- 0x6876e855, 0xfeaf706f, 0xfa43bad2, 0x055fea9d, 0xfcc86e09, 0x01591cc0, 0xffa3760e, 0x000b10ec,
- 0x681100c9, 0xfe09f323, 0xfa90d17b, 0x0540571a, 0xfcd1b74c, 0x01582878, 0xffa2c6c2, 0x000b6c1d,
- 0x67a732f4, 0xfd673159, 0xfaddb10c, 0x05207b2f, 0xfcdb4793, 0x01570b77, 0xffa227ec, 0x000bc2b3,
- 0x673986ac, 0xfcc730aa, 0xfb2a513b, 0x05005b82, 0xfce51ccb, 0x0155c678, 0xffa19957, 0x000c14bb,
- 0x66c80413, 0xfc29f670, 0xfb76a9dd, 0x04dffcb6, 0xfcef34e1, 0x01545a3c, 0xffa11acb, 0x000c6244,
- 0x6652b392, 0xfb8f87bd, 0xfbc2b2e4, 0x04bf6369, 0xfcf98dbe, 0x0152c783, 0xffa0ac11, 0x000cab5c,
- 0x65d99dd5, 0xfaf7e963, 0xfc0e6461, 0x049e9433, 0xfd04254a, 0x01510f13, 0xffa04cf0, 0x000cf012,
- 0x655ccbd3, 0xfa631fef, 0xfc59b685, 0x047d93a8, 0xfd0ef969, 0x014f31b2, 0xff9ffd2c, 0x000d3075,
- 0x64dc46c3, 0xf9d12fab, 0xfca4a19f, 0x045c6654, 0xfd1a0801, 0x014d3029, 0xff9fbc89, 0x000d6c97,
- 0x64581823, 0xf9421c9d, 0xfcef1e20, 0x043b10bd, 0xfd254ef4, 0x014b0b45, 0xff9f8ac9, 0x000da486,
- 0x63d049b4, 0xf8b5ea87, 0xfd392498, 0x04199760, 0xfd30cc24, 0x0148c3d2, 0xff9f67ae, 0x000dd854,
- 0x6344e578, 0xf82c9ce7, 0xfd82adba, 0x03f7feb4, 0xfd3c7d73, 0x01465a9f, 0xff9f52f7, 0x000e0812,
- 0x62b5f5b2, 0xf7a636fa, 0xfdcbb25a, 0x03d64b27, 0xfd4860c2, 0x0143d07f, 0xff9f4c65, 0x000e33d3,
- 0x622384e8, 0xf722bbb5, 0xfe142b6e, 0x03b4811d, 0xfd5473f3, 0x01412643, 0xff9f53b4, 0x000e5ba7,
- 0x618d9ddc, 0xf6a22dcf, 0xfe5c120f, 0x0392a4f4, 0xfd60b4e7, 0x013e5cc0, 0xff9f68a1, 0x000e7fa1,
- 0x60f44b91, 0xf6248fb6, 0xfea35f79, 0x0370bafc, 0xfd6d2180, 0x013b74ca, 0xff9f8ae9, 0x000e9fd5,
- 0x60579947, 0xf5a9e398, 0xfeea0d0c, 0x034ec77f, 0xfd79b7a1, 0x01386f3a, 0xff9fba47, 0x000ebc54,
- 0x5fb79278, 0xf5322b61, 0xff30144a, 0x032ccebb, 0xfd86752e, 0x01354ce7, 0xff9ff674, 0x000ed533,
- 0x5f1442dc, 0xf4bd68b6, 0xff756edc, 0x030ad4e1, 0xfd93580d, 0x01320ea9, 0xffa03f2b, 0x000eea84,
- 0x5e6db665, 0xf44b9cfe, 0xffba168d, 0x02e8de19, 0xfda05e23, 0x012eb55a, 0xffa09425, 0x000efc5c,
- 0x5dc3f93c, 0xf3dcc959, 0xfffe054e, 0x02c6ee7f, 0xfdad855b, 0x012b41d3, 0xffa0f519, 0x000f0ace,
- 0x5d1717c4, 0xf370eea9, 0x00413536, 0x02a50a22, 0xfdbacb9e, 0x0127b4f1, 0xffa161bf, 0x000f15ef,
- 0x5c671e96, 0xf3080d8c, 0x0083a081, 0x02833506, 0xfdc82edb, 0x01240f8e, 0xffa1d9cf, 0x000f1dd2,
- 0x5bb41a80, 0xf2a2265e, 0x00c54190, 0x02617321, 0xfdd5ad01, 0x01205285, 0xffa25cfe, 0x000f228d,
- 0x5afe1886, 0xf23f393b, 0x010612eb, 0x023fc85c, 0xfde34403, 0x011c7eb2, 0xffa2eb04, 0x000f2434,
- 0x5a4525df, 0xf1df45fd, 0x01460f41, 0x021e3891, 0xfdf0f1d6, 0x011894f0, 0xffa38395, 0x000f22dc,
- 0x59894ff3, 0xf1824c3e, 0x01853165, 0x01fcc78f, 0xfdfeb475, 0x0114961b, 0xffa42668, 0x000f1e99,
- 0x58caa45b, 0xf1284b58, 0x01c37452, 0x01db7914, 0xfe0c89db, 0x0110830f, 0xffa4d332, 0x000f1781,
- 0x580930e1, 0xf0d14267, 0x0200d32c, 0x01ba50d2, 0xfe1a7009, 0x010c5ca6, 0xffa589a6, 0x000f0da8,
- 0x5745037c, 0xf07d3043, 0x023d493c, 0x0199526b, 0xfe286505, 0x010823ba, 0xffa6497c, 0x000f0125,
- 0x567e2a51, 0xf02c138a, 0x0278d1f2, 0x01788170, 0xfe3666d5, 0x0103d927, 0xffa71266, 0x000ef20b,
- 0x55b4b3af, 0xefddea9a, 0x02b368e6, 0x0157e166, 0xfe447389, 0x00ff7dc4, 0xffa7e41a, 0x000ee070,
- 0x54e8ae13, 0xef92b393, 0x02ed09d7, 0x013775bf, 0xfe528931, 0x00fb126b, 0xffa8be4c, 0x000ecc69,
- 0x541a281e, 0xef4a6c58, 0x0325b0ad, 0x011741df, 0xfe60a5e5, 0x00f697f3, 0xffa9a0b1, 0x000eb60b,
- 0x5349309e, 0xef051290, 0x035d5977, 0x00f7491a, 0xfe6ec7c0, 0x00f20f32, 0xffaa8afe, 0x000e9d6b,
- 0x5275d684, 0xeec2a3a3, 0x0394006a, 0x00d78eb3, 0xfe7cece2, 0x00ed78ff, 0xffab7ce7, 0x000e829e,
- 0x51a028e8, 0xee831cc3, 0x03c9a1e5, 0x00b815da, 0xfe8b1373, 0x00e8d62d, 0xffac7621, 0x000e65ba,
- 0x50c83704, 0xee467ae1, 0x03fe3a6f, 0x0098e1b3, 0xfe99399f, 0x00e4278f, 0xffad7662, 0x000e46d3,
- 0x4fee1037, 0xee0cbab9, 0x0431c6b5, 0x0079f54c, 0xfea75d97, 0x00df6df7, 0xffae7d5f, 0x000e25fd,
- 0x4f11c3fe, 0xedd5d8ca, 0x0464438c, 0x005b53a4, 0xfeb57d92, 0x00daaa34, 0xffaf8acd, 0x000e034f,
- 0x4e3361f7, 0xeda1d15c, 0x0495adf2, 0x003cffa9, 0xfec397cf, 0x00d5dd16, 0xffb09e63, 0x000ddedb,
- 0x4d52f9df, 0xed70a07d, 0x04c6030d, 0x001efc35, 0xfed1aa92, 0x00d10769, 0xffb1b7d8, 0x000db8b7,
- 0x4c709b8e, 0xed424205, 0x04f54029, 0x00014c12, 0xfedfb425, 0x00cc29f7, 0xffb2d6e1, 0x000d90f6,
- 0x4b8c56f8, 0xed16b196, 0x052362ba, 0xffe3f1f7, 0xfeedb2da, 0x00c7458a, 0xffb3fb37, 0x000d67ae,
- 0x4aa63c2c, 0xecedea99, 0x0550685d, 0xffc6f08a, 0xfefba508, 0x00c25ae8, 0xffb52490, 0x000d3cf1,
- 0x49be5b50, 0xecc7e845, 0x057c4ed4, 0xffaa4a5d, 0xff09890f, 0x00bd6ad7, 0xffb652a7, 0x000d10d5,
- 0x48d4c4a2, 0xeca4a59b, 0x05a7140b, 0xff8e01f1, 0xff175d53, 0x00b87619, 0xffb78533, 0x000ce36b,
- 0x47e98874, 0xec841d68, 0x05d0b612, 0xff7219b3, 0xff252042, 0x00b37d70, 0xffb8bbed, 0x000cb4c8,
- 0x46fcb72d, 0xec664a48, 0x05f93324, 0xff5693fe, 0xff32d04f, 0x00ae8198, 0xffb9f691, 0x000c84ff,
- 0x460e6148, 0xec4b26a2, 0x0620899e, 0xff3b731b, 0xff406bf8, 0x00a9834e, 0xffbb34d8, 0x000c5422,
- 0x451e9750, 0xec32acb0, 0x0646b808, 0xff20b93e, 0xff4df1be, 0x00a4834c, 0xffbc767f, 0x000c2245,
- 0x442d69de, 0xec1cd677, 0x066bbd0d, 0xff066889, 0xff5b602c, 0x009f8249, 0xffbdbb42, 0x000bef79,
- 0x433ae99c, 0xec099dcf, 0x068f9781, 0xfeec830d, 0xff68b5d5, 0x009a80f8, 0xffbf02dd, 0x000bbbd2,
- 0x4247273f, 0xebf8fc64, 0x06b2465b, 0xfed30ac5, 0xff75f153, 0x0095800c, 0xffc04d0f, 0x000b8760,
- 0x41523389, 0xebeaebaf, 0x06d3c8bb, 0xfeba0199, 0xff831148, 0x00908034, 0xffc19996, 0x000b5235,
- 0x405c1f43, 0xebdf6500, 0x06f41de3, 0xfea16960, 0xff90145e, 0x008b821b, 0xffc2e832, 0x000b1c64,
- 0x3f64fb40, 0xebd6617b, 0x0713453d, 0xfe8943dc, 0xff9cf947, 0x0086866b, 0xffc438a3, 0x000ae5fc,
- 0x3e6cd85b, 0xebcfda19, 0x07313e56, 0xfe7192bd, 0xffa9bebe, 0x00818dcb, 0xffc58aaa, 0x000aaf0f,
- 0x3d73c772, 0xebcbc7a7, 0x074e08e0, 0xfe5a579d, 0xffb66386, 0x007c98de, 0xffc6de09, 0x000a77ac,
- 0x3c79d968, 0xebca22cc, 0x0769a4b2, 0xfe439407, 0xffc2e669, 0x0077a845, 0xffc83285, 0x000a3fe5,
- 0x3b7f1f23, 0xebcae405, 0x078411c7, 0xfe2d496f, 0xffcf463a, 0x0072bc9d, 0xffc987e0, 0x000a07c9,
- 0x3a83a989, 0xebce03aa, 0x079d503b, 0xfe177937, 0xffdb81d6, 0x006dd680, 0xffcadde1, 0x0009cf67,
- 0x3987897f, 0xebd379eb, 0x07b56051, 0xfe0224b0, 0xffe79820, 0x0068f687, 0xffcc344c, 0x000996ce,
- 0x388acfe9, 0xebdb3ed5, 0x07cc426c, 0xfded4d13, 0xfff38806, 0x00641d44, 0xffcd8aeb, 0x00095e0e,
- 0x378d8da8, 0xebe54a4f, 0x07e1f712, 0xfdd8f38b, 0xffff507b, 0x005f4b4a, 0xffcee183, 0x00092535,
- 0x368fd397, 0xebf1941f, 0x07f67eec, 0xfdc5192d, 0x000af07f, 0x005a8125, 0xffd037e0, 0x0008ec50,
- 0x3591b28b, 0xec0013e8, 0x0809dac3, 0xfdb1befc, 0x00166718, 0x0055bf60, 0xffd18dcc, 0x0008b36e,
- 0x34933b50, 0xec10c12c, 0x081c0b84, 0xfd9ee5e7, 0x0021b355, 0x00510682, 0xffd2e311, 0x00087a9c,
- 0x33947eab, 0xec23934f, 0x082d1239, 0xfd8c8ecc, 0x002cd44d, 0x004c570f, 0xffd4377d, 0x000841e8,
- 0x32958d55, 0xec388194, 0x083cf010, 0xfd7aba74, 0x0037c922, 0x0047b186, 0xffd58ade, 0x0008095d,
- 0x319677fa, 0xec4f8322, 0x084ba654, 0xfd696998, 0x004290fc, 0x00431666, 0xffd6dd02, 0x0007d108,
- 0x30974f3b, 0xec688f02, 0x08593671, 0xfd589cdc, 0x004d2b0e, 0x003e8628, 0xffd82dba, 0x000798f5,
- 0x2f9823a8, 0xec839c22, 0x0865a1f1, 0xfd4854d3, 0x00579691, 0x003a0141, 0xffd97cd6, 0x00076130,
- 0x2e9905c1, 0xeca0a156, 0x0870ea7e, 0xfd3891fd, 0x0061d2ca, 0x00358824, 0xffdaca2a, 0x000729c4,
- 0x2d9a05f4, 0xecbf9558, 0x087b11de, 0xfd2954c8, 0x006bdf05, 0x00311b41, 0xffdc1588, 0x0006f2bb,
- 0x2c9b349e, 0xece06ecb, 0x088419f6, 0xfd1a9d91, 0x0075ba95, 0x002cbb03, 0xffdd5ec6, 0x0006bc21,
- 0x2b9ca203, 0xed032439, 0x088c04c8, 0xfd0c6ca2, 0x007f64da, 0x002867d2, 0xffdea5bb, 0x000685ff,
- 0x2a9e5e57, 0xed27ac16, 0x0892d470, 0xfcfec233, 0x0088dd38, 0x00242213, 0xffdfea3c, 0x0006505f,
- 0x29a079b2, 0xed4dfcc2, 0x08988b2a, 0xfcf19e6b, 0x0092231e, 0x001fea27, 0xffe12c22, 0x00061b4b,
- 0x28a30416, 0xed760c88, 0x089d2b4a, 0xfce50161, 0x009b3605, 0x001bc06b, 0xffe26b48, 0x0005e6cb,
- 0x27a60d6a, 0xed9fd1a2, 0x08a0b740, 0xfcd8eb17, 0x00a4156b, 0x0017a53b, 0xffe3a788, 0x0005b2e8,
- 0x26a9a57b, 0xedcb4237, 0x08a33196, 0xfccd5b82, 0x00acc0da, 0x001398ec, 0xffe4e0bf, 0x00057faa,
- 0x25addbf9, 0xedf8545b, 0x08a49cf0, 0xfcc25285, 0x00b537e1, 0x000f9bd2, 0xffe616c8, 0x00054d1a,
- 0x24b2c075, 0xee26fe17, 0x08a4fc0d, 0xfcb7cff0, 0x00bd7a1c, 0x000bae3c, 0xffe74984, 0x00051b3e,
- 0x23b86263, 0xee573562, 0x08a451c0, 0xfcadd386, 0x00c5872a, 0x0007d075, 0xffe878d3, 0x0004ea1d,
- 0x22bed116, 0xee88f026, 0x08a2a0f8, 0xfca45cf7, 0x00cd5eb7, 0x000402c8, 0xffe9a494, 0x0004b9c0,
- 0x21c61bc0, 0xeebc2444, 0x089fecbb, 0xfc9b6be5, 0x00d50075, 0x00004579, 0xffeaccaa, 0x00048a2b,
- 0x20ce516f, 0xeef0c78d, 0x089c3824, 0xfc92ffe1, 0x00dc6c1e, 0xfffc98c9, 0xffebf0fa, 0x00045b65,
- 0x1fd7810f, 0xef26cfca, 0x08978666, 0xfc8b186d, 0x00e3a175, 0xfff8fcf7, 0xffed1166, 0x00042d74,
- 0x1ee1b965, 0xef5e32bd, 0x0891dac8, 0xfc83b4fc, 0x00eaa045, 0xfff5723d, 0xffee2dd7, 0x0004005e,
- 0x1ded0911, 0xef96e61c, 0x088b38a9, 0xfc7cd4f0, 0x00f16861, 0xfff1f8d2, 0xffef4632, 0x0003d426,
- 0x1cf97e8b, 0xefd0df9a, 0x0883a378, 0xfc76779e, 0x00f7f9a3, 0xffee90eb, 0xfff05a60, 0x0003a8d2,
- 0x1c072823, 0xf00c14e1, 0x087b1ebc, 0xfc709c4d, 0x00fe53ef, 0xffeb3ab8, 0xfff16a4a, 0x00037e65,
- 0x1b1613ff, 0xf0487b98, 0x0871ae0d, 0xfc6b4233, 0x0104772e, 0xffe7f666, 0xfff275db, 0x000354e5,
- 0x1a26501b, 0xf0860962, 0x08675516, 0xfc66687a, 0x010a6353, 0xffe4c41e, 0xfff37d00, 0x00032c54,
- 0x1937ea47, 0xf0c4b3e0, 0x085c1794, 0xfc620e3d, 0x01101858, 0xffe1a408, 0xfff47fa5, 0x000304b7,
- 0x184af025, 0xf10470b0, 0x084ff957, 0xfc5e328c, 0x0115963d, 0xffde9646, 0xfff57db8, 0x0002de0e,
- 0x175f6f2b, 0xf1453571, 0x0842fe3d, 0xfc5ad465, 0x011add0b, 0xffdb9af8, 0xfff67729, 0x0002b85f,
- 0x1675749e, 0xf186f7c0, 0x08352a35, 0xfc57f2be, 0x011fecd3, 0xffd8b23b, 0xfff76be9, 0x000293aa,
- 0x158d0d95, 0xf1c9ad40, 0x0826813e, 0xfc558c7c, 0x0124c5ab, 0xffd5dc28, 0xfff85be8, 0x00026ff2,
- 0x14a646f6, 0xf20d4b92, 0x08170767, 0xfc53a07b, 0x012967b1, 0xffd318d6, 0xfff9471b, 0x00024d39,
- 0x13c12d73, 0xf251c85d, 0x0806c0cb, 0xfc522d88, 0x012dd30a, 0xffd06858, 0xfffa2d74, 0x00022b7f,
- 0x12ddcd8f, 0xf297194d, 0x07f5b193, 0xfc513266, 0x013207e4, 0xffcdcabe, 0xfffb0ee9, 0x00020ac7,
- 0x11fc3395, 0xf2dd3411, 0x07e3ddf7, 0xfc50adcc, 0x01360670, 0xffcb4014, 0xfffbeb70, 0x0001eb10,
- 0x111c6ba0, 0xf3240e61, 0x07d14a38, 0xfc509e64, 0x0139cee9, 0xffc8c866, 0xfffcc300, 0x0001cc5c,
+#include "AudioResamplerSincUp.h"
};
/*
@@ -197,135 +69,7 @@
* cmd-line: fir -l 7 -s 48000 -c 17189
*/
const uint32_t AudioResamplerSinc::mFirCoefsDown[] __attribute__ ((aligned (32))) = {
- 0x5bacb6f4, 0x1ded1a1d, 0xf0398d56, 0x0394f674, 0x0193a5f9, 0xfe66dbeb, 0x00791043, 0xfffe6631,
- 0x5bab6c81, 0x1d3ddccd, 0xf0421d2c, 0x03af9995, 0x01818dc9, 0xfe6bb63e, 0x0079812a, 0xfffdc37d,
- 0x5ba78d37, 0x1c8f2cf9, 0xf04beb1d, 0x03c9a04a, 0x016f8aca, 0xfe70a511, 0x0079e34d, 0xfffd2545,
- 0x5ba1194f, 0x1be11231, 0xf056f2c7, 0x03e309fe, 0x015d9e64, 0xfe75a79f, 0x007a36e2, 0xfffc8b86,
- 0x5b981122, 0x1b3393f8, 0xf0632fb7, 0x03fbd625, 0x014bc9fa, 0xfe7abd23, 0x007a7c20, 0xfffbf639,
- 0x5b8c7530, 0x1a86b9bf, 0xf0709d74, 0x04140449, 0x013a0ee9, 0xfe7fe4db, 0x007ab33d, 0xfffb655b,
- 0x5b7e461a, 0x19da8ae5, 0xf07f3776, 0x042b93fd, 0x01286e86, 0xfe851e05, 0x007adc72, 0xfffad8e4,
- 0x5b6d84a8, 0x192f0eb7, 0xf08ef92d, 0x044284e6, 0x0116ea22, 0xfe8a67dd, 0x007af7f6, 0xfffa50ce,
- 0x5b5a31c6, 0x18844c70, 0xf09fddfe, 0x0458d6b7, 0x01058306, 0xfe8fc1a5, 0x007b0603, 0xfff9cd12,
- 0x5b444e81, 0x17da4b37, 0xf0b1e143, 0x046e8933, 0x00f43a74, 0xfe952a9b, 0x007b06d4, 0xfff94da9,
- 0x5b2bdc0e, 0x17311222, 0xf0c4fe50, 0x04839c29, 0x00e311a9, 0xfe9aa201, 0x007afaa1, 0xfff8d28c,
- 0x5b10dbc2, 0x1688a832, 0xf0d9306d, 0x04980f79, 0x00d209db, 0xfea02719, 0x007ae1a7, 0xfff85bb1,
- 0x5af34f18, 0x15e11453, 0xf0ee72db, 0x04abe310, 0x00c12439, 0xfea5b926, 0x007abc20, 0xfff7e910,
- 0x5ad337af, 0x153a5d5e, 0xf104c0d2, 0x04bf16e9, 0x00b061eb, 0xfeab576d, 0x007a8a49, 0xfff77a9f,
- 0x5ab09748, 0x14948a16, 0xf11c1583, 0x04d1ab0d, 0x009fc413, 0xfeb10134, 0x007a4c5d, 0xfff71057,
- 0x5a8b6fc7, 0x13efa12c, 0xf1346c17, 0x04e39f93, 0x008f4bcb, 0xfeb6b5c0, 0x007a029a, 0xfff6aa2b,
- 0x5a63c336, 0x134ba937, 0xf14dbfb1, 0x04f4f4a2, 0x007efa29, 0xfebc745c, 0x0079ad3d, 0xfff64812,
- 0x5a3993c0, 0x12a8a8bb, 0xf1680b6e, 0x0505aa6a, 0x006ed038, 0xfec23c50, 0x00794c82, 0xfff5ea02,
- 0x5a0ce3b2, 0x1206a625, 0xf1834a63, 0x0515c12d, 0x005ecf01, 0xfec80ce8, 0x0078e0a9, 0xfff58ff0,
- 0x59ddb57f, 0x1165a7cc, 0xf19f77a0, 0x05253938, 0x004ef782, 0xfecde571, 0x007869ee, 0xfff539cf,
- 0x59ac0bba, 0x10c5b3ef, 0xf1bc8e31, 0x053412e4, 0x003f4ab4, 0xfed3c538, 0x0077e891, 0xfff4e794,
- 0x5977e919, 0x1026d0b8, 0xf1da891b, 0x05424e9b, 0x002fc98a, 0xfed9ab8f, 0x00775ccf, 0xfff49934,
- 0x59415075, 0x0f890437, 0xf1f96360, 0x054feccf, 0x002074ed, 0xfedf97c6, 0x0076c6e8, 0xfff44ea3,
- 0x590844c9, 0x0eec5465, 0xf21917ff, 0x055cee03, 0x00114dc3, 0xfee58932, 0x00762719, 0xfff407d2,
- 0x58ccc930, 0x0e50c723, 0xf239a1ef, 0x056952c3, 0x000254e8, 0xfeeb7f27, 0x00757da3, 0xfff3c4b7,
- 0x588ee0ea, 0x0db6623b, 0xf25afc29, 0x05751baa, 0xfff38b32, 0xfef178fc, 0x0074cac4, 0xfff38542,
- 0x584e8f56, 0x0d1d2b5d, 0xf27d219f, 0x0580495c, 0xffe4f171, 0xfef7760c, 0x00740ebb, 0xfff34968,
- 0x580bd7f4, 0x0c85281f, 0xf2a00d43, 0x058adc8d, 0xffd6886d, 0xfefd75af, 0x007349c7, 0xfff3111b,
- 0x57c6be67, 0x0bee5dff, 0xf2c3ba04, 0x0594d5fa, 0xffc850e6, 0xff037744, 0x00727c27, 0xfff2dc4c,
- 0x577f4670, 0x0b58d262, 0xf2e822ce, 0x059e366c, 0xffba4b98, 0xff097a29, 0x0071a61b, 0xfff2aaef,
- 0x573573f2, 0x0ac48a92, 0xf30d428e, 0x05a6feb9, 0xffac7936, 0xff0f7dbf, 0x0070c7e1, 0xfff27cf3,
- 0x56e94af1, 0x0a318bc1, 0xf333142f, 0x05af2fbf, 0xff9eda6d, 0xff15816a, 0x006fe1b8, 0xfff2524c,
- 0x569acf90, 0x099fdb04, 0xf359929a, 0x05b6ca6b, 0xff916fe1, 0xff1b848e, 0x006ef3df, 0xfff22aea,
- 0x564a0610, 0x090f7d57, 0xf380b8ba, 0x05bdcfb2, 0xff843a32, 0xff218692, 0x006dfe94, 0xfff206bf,
- 0x55f6f2d3, 0x0880779d, 0xf3a88179, 0x05c44095, 0xff7739f7, 0xff2786e1, 0x006d0217, 0xfff1e5bb,
- 0x55a19a5c, 0x07f2ce9b, 0xf3d0e7c2, 0x05ca1e1f, 0xff6a6fc1, 0xff2d84e5, 0x006bfea4, 0xfff1c7d0,
- 0x554a0148, 0x076686fc, 0xf3f9e680, 0x05cf6965, 0xff5ddc1a, 0xff33800e, 0x006af47b, 0xfff1acef,
- 0x54f02c56, 0x06dba551, 0xf42378a0, 0x05d42387, 0xff517f86, 0xff3977cb, 0x0069e3d9, 0xfff19508,
- 0x54942061, 0x06522e0f, 0xf44d9912, 0x05d84daf, 0xff455a80, 0xff3f6b8f, 0x0068ccfa, 0xfff1800b,
- 0x5435e263, 0x05ca258f, 0xf47842c5, 0x05dbe90f, 0xff396d7f, 0xff455acf, 0x0067b01e, 0xfff16de9,
- 0x53d57774, 0x0543900d, 0xf4a370ad, 0x05def6e4, 0xff2db8f2, 0xff4b4503, 0x00668d80, 0xfff15e93,
- 0x5372e4c6, 0x04be71ab, 0xf4cf1dbf, 0x05e17873, 0xff223d40, 0xff5129a3, 0x0065655d, 0xfff151f9,
- 0x530e2fac, 0x043ace6e, 0xf4fb44f4, 0x05e36f0d, 0xff16faca, 0xff57082e, 0x006437f1, 0xfff1480b,
- 0x52a75d90, 0x03b8aa40, 0xf527e149, 0x05e4dc08, 0xff0bf1ed, 0xff5ce021, 0x00630577, 0xfff140b9,
- 0x523e73fd, 0x033808eb, 0xf554edbd, 0x05e5c0c6, 0xff0122fc, 0xff62b0fd, 0x0061ce2c, 0xfff13bf3,
- 0x51d37897, 0x02b8ee22, 0xf5826555, 0x05e61eae, 0xfef68e45, 0xff687a47, 0x00609249, 0xfff139aa,
- 0x5166711c, 0x023b5d76, 0xf5b0431a, 0x05e5f733, 0xfeec340f, 0xff6e3b84, 0x005f520a, 0xfff139cd,
- 0x50f76368, 0x01bf5a5e, 0xf5de8218, 0x05e54bcd, 0xfee2149b, 0xff73f43d, 0x005e0da8, 0xfff13c4c,
- 0x5086556f, 0x0144e834, 0xf60d1d63, 0x05e41dfe, 0xfed83023, 0xff79a3fe, 0x005cc55c, 0xfff14119,
- 0x50134d3e, 0x00cc0a36, 0xf63c1012, 0x05e26f4e, 0xfece86db, 0xff7f4a54, 0x005b7961, 0xfff14821,
- 0x4f9e50ff, 0x0054c382, 0xf66b5544, 0x05e0414d, 0xfec518f1, 0xff84e6d0, 0x005a29ed, 0xfff15156,
- 0x4f2766f2, 0xffdf171b, 0xf69ae81d, 0x05dd9593, 0xfebbe68c, 0xff8a7905, 0x0058d738, 0xfff15ca8,
- 0x4eae9571, 0xff6b07e7, 0xf6cac3c7, 0x05da6dbe, 0xfeb2efcd, 0xff900089, 0x0057817b, 0xfff16a07,
- 0x4e33e2ee, 0xfef898ae, 0xf6fae373, 0x05d6cb72, 0xfeaa34d0, 0xff957cf4, 0x005628ec, 0xfff17962,
- 0x4db755f3, 0xfe87cc1b, 0xf72b425b, 0x05d2b05c, 0xfea1b5a9, 0xff9aede0, 0x0054cdc0, 0xfff18aab,
- 0x4d38f520, 0xfe18a4bc, 0xf75bdbbd, 0x05ce1e2d, 0xfe997268, 0xffa052ec, 0x0053702d, 0xfff19dd1,
- 0x4cb8c72e, 0xfdab2501, 0xf78caae0, 0x05c9169d, 0xfe916b15, 0xffa5abb8, 0x00521068, 0xfff1b2c5,
- 0x4c36d2eb, 0xfd3f4f3d, 0xf7bdab16, 0x05c39b6a, 0xfe899fb2, 0xffaaf7e6, 0x0050aea5, 0xfff1c976,
- 0x4bb31f3c, 0xfcd525a5, 0xf7eed7b4, 0x05bdae57, 0xfe82103f, 0xffb0371c, 0x004f4b17, 0xfff1e1d6,
- 0x4b2db31a, 0xfc6caa53, 0xf8202c1c, 0x05b7512e, 0xfe7abcb1, 0xffb56902, 0x004de5f1, 0xfff1fbd5,
- 0x4aa69594, 0xfc05df40, 0xf851a3b6, 0x05b085bc, 0xfe73a4fb, 0xffba8d44, 0x004c7f66, 0xfff21764,
- 0x4a1dcdce, 0xfba0c64b, 0xf88339f5, 0x05a94dd5, 0xfe6cc909, 0xffbfa38d, 0x004b17a6, 0xfff23473,
- 0x499362ff, 0xfb3d6133, 0xf8b4ea55, 0x05a1ab52, 0xfe6628c1, 0xffc4ab8f, 0x0049aee3, 0xfff252f3,
- 0x49075c72, 0xfadbb19a, 0xf8e6b059, 0x0599a00e, 0xfe5fc405, 0xffc9a4fc, 0x0048454b, 0xfff272d6,
- 0x4879c185, 0xfa7bb908, 0xf9188793, 0x05912dea, 0xfe599aaf, 0xffce8f8a, 0x0046db0f, 0xfff2940b,
- 0x47ea99a9, 0xfa1d78e3, 0xf94a6b9b, 0x058856cd, 0xfe53ac97, 0xffd36af1, 0x0045705c, 0xfff2b686,
- 0x4759ec60, 0xf9c0f276, 0xf97c5815, 0x057f1c9e, 0xfe4df98e, 0xffd836eb, 0x00440561, 0xfff2da36,
- 0x46c7c140, 0xf96626f0, 0xf9ae48af, 0x0575814c, 0xfe48815e, 0xffdcf336, 0x00429a4a, 0xfff2ff0d,
- 0x46341fed, 0xf90d1761, 0xf9e03924, 0x056b86c6, 0xfe4343d0, 0xffe19f91, 0x00412f43, 0xfff324fd,
- 0x459f101d, 0xf8b5c4be, 0xfa122537, 0x05612f00, 0xfe3e40a6, 0xffe63bc0, 0x003fc478, 0xfff34bf9,
- 0x45089996, 0xf8602fdc, 0xfa4408ba, 0x05567bf1, 0xfe39779a, 0xffeac787, 0x003e5a12, 0xfff373f0,
- 0x4470c42d, 0xf80c5977, 0xfa75df87, 0x054b6f92, 0xfe34e867, 0xffef42af, 0x003cf03d, 0xfff39cd7,
- 0x43d797c7, 0xf7ba422b, 0xfaa7a586, 0x05400be1, 0xfe3092bf, 0xfff3ad01, 0x003b871f, 0xfff3c69f,
- 0x433d1c56, 0xf769ea78, 0xfad956ab, 0x053452dc, 0xfe2c7650, 0xfff8064b, 0x003a1ee3, 0xfff3f13a,
- 0x42a159dc, 0xf71b52c4, 0xfb0aeef6, 0x05284685, 0xfe2892c5, 0xfffc4e5c, 0x0038b7ae, 0xfff41c9c,
- 0x42045865, 0xf6ce7b57, 0xfb3c6a73, 0x051be8dd, 0xfe24e7c3, 0x00008507, 0x003751a7, 0xfff448b7,
- 0x4166200e, 0xf683645a, 0xfb6dc53c, 0x050f3bec, 0xfe2174ec, 0x0004aa1f, 0x0035ecf4, 0xfff4757e,
- 0x40c6b8fd, 0xf63a0ddf, 0xfb9efb77, 0x050241b6, 0xfe1e39da, 0x0008bd7c, 0x003489b9, 0xfff4a2e5,
- 0x40262b65, 0xf5f277d9, 0xfbd00956, 0x04f4fc46, 0xfe1b3628, 0x000cbef7, 0x0033281a, 0xfff4d0de,
- 0x3f847f83, 0xf5aca21f, 0xfc00eb1b, 0x04e76da3, 0xfe18696a, 0x0010ae6e, 0x0031c83a, 0xfff4ff5d,
- 0x3ee1bda2, 0xf5688c6d, 0xfc319d13, 0x04d997d8, 0xfe15d32f, 0x00148bbd, 0x00306a3b, 0xfff52e57,
- 0x3e3dee13, 0xf5263665, 0xfc621b9a, 0x04cb7cf2, 0xfe137304, 0x001856c7, 0x002f0e3f, 0xfff55dbf,
- 0x3d991932, 0xf4e59f8a, 0xfc926319, 0x04bd1efb, 0xfe114872, 0x001c0f6e, 0x002db466, 0xfff58d89,
- 0x3cf34766, 0xf4a6c748, 0xfcc27008, 0x04ae8000, 0xfe0f52fc, 0x001fb599, 0x002c5cd0, 0xfff5bdaa,
- 0x3c4c811c, 0xf469aced, 0xfcf23eec, 0x049fa20f, 0xfe0d9224, 0x0023492f, 0x002b079a, 0xfff5ee17,
- 0x3ba4cec9, 0xf42e4faf, 0xfd21cc59, 0x04908733, 0xfe0c0567, 0x0026ca1c, 0x0029b4e4, 0xfff61ec5,
- 0x3afc38eb, 0xf3f4aea6, 0xfd5114f0, 0x0481317a, 0xfe0aac3f, 0x002a384c, 0x002864c9, 0xfff64fa8,
- 0x3a52c805, 0xf3bcc8d3, 0xfd801564, 0x0471a2ef, 0xfe098622, 0x002d93ae, 0x00271766, 0xfff680b5,
- 0x39a884a1, 0xf3869d1a, 0xfdaeca73, 0x0461dda0, 0xfe089283, 0x0030dc34, 0x0025ccd7, 0xfff6b1e4,
- 0x38fd774e, 0xf3522a49, 0xfddd30eb, 0x0451e396, 0xfe07d0d3, 0x003411d2, 0x00248535, 0xfff6e329,
- 0x3851a8a2, 0xf31f6f0f, 0xfe0b45aa, 0x0441b6dd, 0xfe07407d, 0x0037347d, 0x0023409a, 0xfff7147a,
- 0x37a52135, 0xf2ee6a07, 0xfe39059b, 0x0431597d, 0xfe06e0eb, 0x003a442e, 0x0021ff1f, 0xfff745cd,
- 0x36f7e9a4, 0xf2bf19ae, 0xfe666dbc, 0x0420cd80, 0xfe06b184, 0x003d40e0, 0x0020c0dc, 0xfff7771a,
- 0x364a0a90, 0xf2917c6d, 0xfe937b15, 0x041014eb, 0xfe06b1ac, 0x00402a8e, 0x001f85e6, 0xfff7a857,
- 0x359b8c9d, 0xf265908f, 0xfec02ac2, 0x03ff31c3, 0xfe06e0c4, 0x00430137, 0x001e4e56, 0xfff7d97a,
- 0x34ec786f, 0xf23b544b, 0xfeec79ec, 0x03ee260d, 0xfe073e2a, 0x0045c4dd, 0x001d1a3f, 0xfff80a7c,
- 0x343cd6af, 0xf212c5be, 0xff1865cd, 0x03dcf3ca, 0xfe07c93a, 0x00487582, 0x001be9b7, 0xfff83b52,
- 0x338cb004, 0xf1ebe2ec, 0xff43ebac, 0x03cb9cf9, 0xfe08814e, 0x004b132b, 0x001abcd0, 0xfff86bf6,
- 0x32dc0d17, 0xf1c6a9c3, 0xff6f08e4, 0x03ba2398, 0xfe0965bc, 0x004d9dde, 0x0019939d, 0xfff89c60,
- 0x322af693, 0xf1a3181a, 0xff99badb, 0x03a889a1, 0xfe0a75da, 0x005015a5, 0x00186e31, 0xfff8cc86,
- 0x3179751f, 0xf1812bb0, 0xffc3ff0c, 0x0396d10c, 0xfe0bb0f9, 0x00527a8a, 0x00174c9c, 0xfff8fc62,
- 0x30c79163, 0xf160e22d, 0xffedd2fd, 0x0384fbd1, 0xfe0d166b, 0x0054cc9a, 0x00162eef, 0xfff92bec,
- 0x30155404, 0xf1423924, 0x00173447, 0x03730be0, 0xfe0ea57e, 0x00570be4, 0x00151538, 0xfff95b1e,
- 0x2f62c5a7, 0xf1252e0f, 0x00402092, 0x0361032a, 0xfe105d7e, 0x00593877, 0x0013ff88, 0xfff989ef,
- 0x2eafeeed, 0xf109be56, 0x00689598, 0x034ee39b, 0xfe123db6, 0x005b5267, 0x0012edea, 0xfff9b85b,
- 0x2dfcd873, 0xf0efe748, 0x0090911f, 0x033caf1d, 0xfe144570, 0x005d59c6, 0x0011e06d, 0xfff9e65a,
- 0x2d498ad3, 0xf0d7a622, 0x00b81102, 0x032a6796, 0xfe1673f2, 0x005f4eac, 0x0010d71d, 0xfffa13e5,
- 0x2c960ea3, 0xf0c0f808, 0x00df1328, 0x03180ee7, 0xfe18c884, 0x0061312e, 0x000fd205, 0xfffa40f8,
- 0x2be26c73, 0xf0abda0e, 0x0105958c, 0x0305a6f0, 0xfe1b4268, 0x00630167, 0x000ed130, 0xfffa6d8d,
- 0x2b2eaccf, 0xf0984931, 0x012b9635, 0x02f3318a, 0xfe1de0e2, 0x0064bf71, 0x000dd4a7, 0xfffa999d,
- 0x2a7ad83c, 0xf086425a, 0x0151133e, 0x02e0b08d, 0xfe20a335, 0x00666b68, 0x000cdc74, 0xfffac525,
- 0x29c6f738, 0xf075c260, 0x01760ad1, 0x02ce25ca, 0xfe2388a1, 0x0068056b, 0x000be89f, 0xfffaf01e,
- 0x2913123c, 0xf066c606, 0x019a7b27, 0x02bb9310, 0xfe269065, 0x00698d98, 0x000af931, 0xfffb1a84,
- 0x285f31b7, 0xf05949fb, 0x01be628c, 0x02a8fa2a, 0xfe29b9c1, 0x006b0411, 0x000a0e2f, 0xfffb4453,
- 0x27ab5e12, 0xf04d4ade, 0x01e1bf58, 0x02965cdb, 0xfe2d03f2, 0x006c68f8, 0x000927a0, 0xfffb6d86,
- 0x26f79fab, 0xf042c539, 0x02048ff8, 0x0283bce6, 0xfe306e35, 0x006dbc71, 0x00084589, 0xfffb961a,
- 0x2643feda, 0xf039b587, 0x0226d2e6, 0x02711c05, 0xfe33f7c7, 0x006efea0, 0x000767f0, 0xfffbbe09,
- 0x259083eb, 0xf032182f, 0x024886ad, 0x025e7bf0, 0xfe379fe3, 0x00702fae, 0x00068ed8, 0xfffbe552,
- 0x24dd3721, 0xf02be98a, 0x0269a9e9, 0x024bde5a, 0xfe3b65c4, 0x00714fc0, 0x0005ba46, 0xfffc0bef,
- 0x242a20b3, 0xf02725dc, 0x028a3b44, 0x023944ee, 0xfe3f48a5, 0x00725f02, 0x0004ea3a, 0xfffc31df,
- 0x237748cf, 0xf023c95d, 0x02aa397b, 0x0226b156, 0xfe4347c0, 0x00735d9c, 0x00041eb9, 0xfffc571e,
- 0x22c4b795, 0xf021d031, 0x02c9a359, 0x02142533, 0xfe476250, 0x00744bba, 0x000357c2, 0xfffc7ba9,
- 0x2212751a, 0xf0213671, 0x02e877b9, 0x0201a223, 0xfe4b978e, 0x0075298a, 0x00029558, 0xfffc9f7e,
- 0x21608968, 0xf021f823, 0x0306b586, 0x01ef29be, 0xfe4fe6b3, 0x0075f739, 0x0001d779, 0xfffcc29a,
- 0x20aefc79, 0xf0241140, 0x03245bbc, 0x01dcbd96, 0xfe544efb, 0x0076b4f5, 0x00011e26, 0xfffce4fc,
- 0x1ffdd63b, 0xf0277db1, 0x03416966, 0x01ca5f37, 0xfe58cf9d, 0x007762f0, 0x0000695e, 0xfffd06a1,
- 0x1f4d1e8e, 0xf02c3953, 0x035ddd9e, 0x01b81028, 0xfe5d67d4, 0x0078015a, 0xffffb91f, 0xfffd2787,
- 0x1e9cdd43, 0xf0323ff5, 0x0379b790, 0x01a5d1ea, 0xfe6216db, 0x00789065, 0xffff0d66, 0xfffd47ae,
- 0x1ded1a1d, 0xf0398d56, 0x0394f674, 0x0193a5f9, 0xfe66dbeb, 0x00791043, 0xfffe6631, 0xfffd6713,
+#include "AudioResamplerSincDown.h"
};
// we use 15 bits to interpolate between these samples
@@ -521,7 +265,8 @@
if (mConstants == &veryHighQualityConstants && readResampleCoefficients) {
mFirCoefs = readResampleCoefficients( mInSampleRate <= mSampleRate );
} else {
- mFirCoefs = (const int32_t *) ((mInSampleRate <= mSampleRate) ? mFirCoefsUp : mFirCoefsDown);
+ mFirCoefs = (const int32_t *)
+ ((mInSampleRate <= mSampleRate) ? mFirCoefsUp : mFirCoefsDown);
}
// select the appropriate resampler
@@ -856,4 +601,4 @@
}
}
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index 4691d0a..6d8e85d 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -95,6 +95,6 @@
};
// ----------------------------------------------------------------------------
-}; // namespace android
+} // namespace android
#endif /*ANDROID_AUDIO_RESAMPLER_SINC_H*/
diff --git a/services/audioflinger/AudioResamplerSincDown.h b/services/audioflinger/AudioResamplerSincDown.h
new file mode 100644
index 0000000..2d0fb86
--- /dev/null
+++ b/services/audioflinger/AudioResamplerSincDown.h
@@ -0,0 +1,131 @@
+// cmd-line: fir -l 7 -s48000 -c 17189
+
+ 0x5bacb6f4, 0x1ded1a1d, 0xf0398d56, 0x0394f674, 0x0193a5f9, 0xfe66dbeb, 0x00791043, 0xfffe6631,
+ 0x5bab6c81, 0x1d3ddccd, 0xf0421d2c, 0x03af9995, 0x01818dc9, 0xfe6bb63e, 0x0079812a, 0xfffdc37d,
+ 0x5ba78d37, 0x1c8f2cf9, 0xf04beb1d, 0x03c9a04a, 0x016f8aca, 0xfe70a511, 0x0079e34d, 0xfffd2545,
+ 0x5ba1194f, 0x1be11231, 0xf056f2c7, 0x03e309fe, 0x015d9e64, 0xfe75a79f, 0x007a36e2, 0xfffc8b86,
+ 0x5b981122, 0x1b3393f8, 0xf0632fb7, 0x03fbd625, 0x014bc9fa, 0xfe7abd23, 0x007a7c20, 0xfffbf639,
+ 0x5b8c7530, 0x1a86b9bf, 0xf0709d74, 0x04140449, 0x013a0ee9, 0xfe7fe4db, 0x007ab33d, 0xfffb655b,
+ 0x5b7e461a, 0x19da8ae5, 0xf07f3776, 0x042b93fd, 0x01286e86, 0xfe851e05, 0x007adc72, 0xfffad8e4,
+ 0x5b6d84a8, 0x192f0eb7, 0xf08ef92d, 0x044284e6, 0x0116ea22, 0xfe8a67dd, 0x007af7f6, 0xfffa50ce,
+ 0x5b5a31c6, 0x18844c70, 0xf09fddfe, 0x0458d6b7, 0x01058306, 0xfe8fc1a5, 0x007b0603, 0xfff9cd12,
+ 0x5b444e81, 0x17da4b37, 0xf0b1e143, 0x046e8933, 0x00f43a74, 0xfe952a9b, 0x007b06d4, 0xfff94da9,
+ 0x5b2bdc0e, 0x17311222, 0xf0c4fe50, 0x04839c29, 0x00e311a9, 0xfe9aa201, 0x007afaa1, 0xfff8d28c,
+ 0x5b10dbc2, 0x1688a832, 0xf0d9306d, 0x04980f79, 0x00d209db, 0xfea02719, 0x007ae1a7, 0xfff85bb1,
+ 0x5af34f18, 0x15e11453, 0xf0ee72db, 0x04abe310, 0x00c12439, 0xfea5b926, 0x007abc20, 0xfff7e910,
+ 0x5ad337af, 0x153a5d5e, 0xf104c0d2, 0x04bf16e9, 0x00b061eb, 0xfeab576d, 0x007a8a49, 0xfff77a9f,
+ 0x5ab09748, 0x14948a16, 0xf11c1583, 0x04d1ab0d, 0x009fc413, 0xfeb10134, 0x007a4c5d, 0xfff71057,
+ 0x5a8b6fc7, 0x13efa12c, 0xf1346c17, 0x04e39f93, 0x008f4bcb, 0xfeb6b5c0, 0x007a029a, 0xfff6aa2b,
+ 0x5a63c336, 0x134ba937, 0xf14dbfb1, 0x04f4f4a2, 0x007efa29, 0xfebc745c, 0x0079ad3d, 0xfff64812,
+ 0x5a3993c0, 0x12a8a8bb, 0xf1680b6e, 0x0505aa6a, 0x006ed038, 0xfec23c50, 0x00794c82, 0xfff5ea02,
+ 0x5a0ce3b2, 0x1206a625, 0xf1834a63, 0x0515c12d, 0x005ecf01, 0xfec80ce8, 0x0078e0a9, 0xfff58ff0,
+ 0x59ddb57f, 0x1165a7cc, 0xf19f77a0, 0x05253938, 0x004ef782, 0xfecde571, 0x007869ee, 0xfff539cf,
+ 0x59ac0bba, 0x10c5b3ef, 0xf1bc8e31, 0x053412e4, 0x003f4ab4, 0xfed3c538, 0x0077e891, 0xfff4e794,
+ 0x5977e919, 0x1026d0b8, 0xf1da891b, 0x05424e9b, 0x002fc98a, 0xfed9ab8f, 0x00775ccf, 0xfff49934,
+ 0x59415075, 0x0f890437, 0xf1f96360, 0x054feccf, 0x002074ed, 0xfedf97c6, 0x0076c6e8, 0xfff44ea3,
+ 0x590844c9, 0x0eec5465, 0xf21917ff, 0x055cee03, 0x00114dc3, 0xfee58932, 0x00762719, 0xfff407d2,
+ 0x58ccc930, 0x0e50c723, 0xf239a1ef, 0x056952c3, 0x000254e8, 0xfeeb7f27, 0x00757da3, 0xfff3c4b7,
+ 0x588ee0ea, 0x0db6623b, 0xf25afc29, 0x05751baa, 0xfff38b32, 0xfef178fc, 0x0074cac4, 0xfff38542,
+ 0x584e8f56, 0x0d1d2b5d, 0xf27d219f, 0x0580495c, 0xffe4f171, 0xfef7760c, 0x00740ebb, 0xfff34968,
+ 0x580bd7f4, 0x0c85281f, 0xf2a00d43, 0x058adc8d, 0xffd6886d, 0xfefd75af, 0x007349c7, 0xfff3111b,
+ 0x57c6be67, 0x0bee5dff, 0xf2c3ba04, 0x0594d5fa, 0xffc850e6, 0xff037744, 0x00727c27, 0xfff2dc4c,
+ 0x577f4670, 0x0b58d262, 0xf2e822ce, 0x059e366c, 0xffba4b98, 0xff097a29, 0x0071a61b, 0xfff2aaef,
+ 0x573573f2, 0x0ac48a92, 0xf30d428e, 0x05a6feb9, 0xffac7936, 0xff0f7dbf, 0x0070c7e1, 0xfff27cf3,
+ 0x56e94af1, 0x0a318bc1, 0xf333142f, 0x05af2fbf, 0xff9eda6d, 0xff15816a, 0x006fe1b8, 0xfff2524c,
+ 0x569acf90, 0x099fdb04, 0xf359929a, 0x05b6ca6b, 0xff916fe1, 0xff1b848e, 0x006ef3df, 0xfff22aea,
+ 0x564a0610, 0x090f7d57, 0xf380b8ba, 0x05bdcfb2, 0xff843a32, 0xff218692, 0x006dfe94, 0xfff206bf,
+ 0x55f6f2d3, 0x0880779d, 0xf3a88179, 0x05c44095, 0xff7739f7, 0xff2786e1, 0x006d0217, 0xfff1e5bb,
+ 0x55a19a5c, 0x07f2ce9b, 0xf3d0e7c2, 0x05ca1e1f, 0xff6a6fc1, 0xff2d84e5, 0x006bfea4, 0xfff1c7d0,
+ 0x554a0148, 0x076686fc, 0xf3f9e680, 0x05cf6965, 0xff5ddc1a, 0xff33800e, 0x006af47b, 0xfff1acef,
+ 0x54f02c56, 0x06dba551, 0xf42378a0, 0x05d42387, 0xff517f86, 0xff3977cb, 0x0069e3d9, 0xfff19508,
+ 0x54942061, 0x06522e0f, 0xf44d9912, 0x05d84daf, 0xff455a80, 0xff3f6b8f, 0x0068ccfa, 0xfff1800b,
+ 0x5435e263, 0x05ca258f, 0xf47842c5, 0x05dbe90f, 0xff396d7f, 0xff455acf, 0x0067b01e, 0xfff16de9,
+ 0x53d57774, 0x0543900d, 0xf4a370ad, 0x05def6e4, 0xff2db8f2, 0xff4b4503, 0x00668d80, 0xfff15e93,
+ 0x5372e4c6, 0x04be71ab, 0xf4cf1dbf, 0x05e17873, 0xff223d40, 0xff5129a3, 0x0065655d, 0xfff151f9,
+ 0x530e2fac, 0x043ace6e, 0xf4fb44f4, 0x05e36f0d, 0xff16faca, 0xff57082e, 0x006437f1, 0xfff1480b,
+ 0x52a75d90, 0x03b8aa40, 0xf527e149, 0x05e4dc08, 0xff0bf1ed, 0xff5ce021, 0x00630577, 0xfff140b9,
+ 0x523e73fd, 0x033808eb, 0xf554edbd, 0x05e5c0c6, 0xff0122fc, 0xff62b0fd, 0x0061ce2c, 0xfff13bf3,
+ 0x51d37897, 0x02b8ee22, 0xf5826555, 0x05e61eae, 0xfef68e45, 0xff687a47, 0x00609249, 0xfff139aa,
+ 0x5166711c, 0x023b5d76, 0xf5b0431a, 0x05e5f733, 0xfeec340f, 0xff6e3b84, 0x005f520a, 0xfff139cd,
+ 0x50f76368, 0x01bf5a5e, 0xf5de8218, 0x05e54bcd, 0xfee2149b, 0xff73f43d, 0x005e0da8, 0xfff13c4c,
+ 0x5086556f, 0x0144e834, 0xf60d1d63, 0x05e41dfe, 0xfed83023, 0xff79a3fe, 0x005cc55c, 0xfff14119,
+ 0x50134d3e, 0x00cc0a36, 0xf63c1012, 0x05e26f4e, 0xfece86db, 0xff7f4a54, 0x005b7961, 0xfff14821,
+ 0x4f9e50ff, 0x0054c382, 0xf66b5544, 0x05e0414d, 0xfec518f1, 0xff84e6d0, 0x005a29ed, 0xfff15156,
+ 0x4f2766f2, 0xffdf171b, 0xf69ae81d, 0x05dd9593, 0xfebbe68c, 0xff8a7905, 0x0058d738, 0xfff15ca8,
+ 0x4eae9571, 0xff6b07e7, 0xf6cac3c7, 0x05da6dbe, 0xfeb2efcd, 0xff900089, 0x0057817b, 0xfff16a07,
+ 0x4e33e2ee, 0xfef898ae, 0xf6fae373, 0x05d6cb72, 0xfeaa34d0, 0xff957cf4, 0x005628ec, 0xfff17962,
+ 0x4db755f3, 0xfe87cc1b, 0xf72b425b, 0x05d2b05c, 0xfea1b5a9, 0xff9aede0, 0x0054cdc0, 0xfff18aab,
+ 0x4d38f520, 0xfe18a4bc, 0xf75bdbbd, 0x05ce1e2d, 0xfe997268, 0xffa052ec, 0x0053702d, 0xfff19dd1,
+ 0x4cb8c72e, 0xfdab2501, 0xf78caae0, 0x05c9169d, 0xfe916b15, 0xffa5abb8, 0x00521068, 0xfff1b2c5,
+ 0x4c36d2eb, 0xfd3f4f3d, 0xf7bdab16, 0x05c39b6a, 0xfe899fb2, 0xffaaf7e6, 0x0050aea5, 0xfff1c976,
+ 0x4bb31f3c, 0xfcd525a5, 0xf7eed7b4, 0x05bdae57, 0xfe82103f, 0xffb0371c, 0x004f4b17, 0xfff1e1d6,
+ 0x4b2db31a, 0xfc6caa53, 0xf8202c1c, 0x05b7512e, 0xfe7abcb1, 0xffb56902, 0x004de5f1, 0xfff1fbd5,
+ 0x4aa69594, 0xfc05df40, 0xf851a3b6, 0x05b085bc, 0xfe73a4fb, 0xffba8d44, 0x004c7f66, 0xfff21764,
+ 0x4a1dcdce, 0xfba0c64b, 0xf88339f5, 0x05a94dd5, 0xfe6cc909, 0xffbfa38d, 0x004b17a6, 0xfff23473,
+ 0x499362ff, 0xfb3d6133, 0xf8b4ea55, 0x05a1ab52, 0xfe6628c1, 0xffc4ab8f, 0x0049aee3, 0xfff252f3,
+ 0x49075c72, 0xfadbb19a, 0xf8e6b059, 0x0599a00e, 0xfe5fc405, 0xffc9a4fc, 0x0048454b, 0xfff272d6,
+ 0x4879c185, 0xfa7bb908, 0xf9188793, 0x05912dea, 0xfe599aaf, 0xffce8f8a, 0x0046db0f, 0xfff2940b,
+ 0x47ea99a9, 0xfa1d78e3, 0xf94a6b9b, 0x058856cd, 0xfe53ac97, 0xffd36af1, 0x0045705c, 0xfff2b686,
+ 0x4759ec60, 0xf9c0f276, 0xf97c5815, 0x057f1c9e, 0xfe4df98e, 0xffd836eb, 0x00440561, 0xfff2da36,
+ 0x46c7c140, 0xf96626f0, 0xf9ae48af, 0x0575814c, 0xfe48815e, 0xffdcf336, 0x00429a4a, 0xfff2ff0d,
+ 0x46341fed, 0xf90d1761, 0xf9e03924, 0x056b86c6, 0xfe4343d0, 0xffe19f91, 0x00412f43, 0xfff324fd,
+ 0x459f101d, 0xf8b5c4be, 0xfa122537, 0x05612f00, 0xfe3e40a6, 0xffe63bc0, 0x003fc478, 0xfff34bf9,
+ 0x45089996, 0xf8602fdc, 0xfa4408ba, 0x05567bf1, 0xfe39779a, 0xffeac787, 0x003e5a12, 0xfff373f0,
+ 0x4470c42d, 0xf80c5977, 0xfa75df87, 0x054b6f92, 0xfe34e867, 0xffef42af, 0x003cf03d, 0xfff39cd7,
+ 0x43d797c7, 0xf7ba422b, 0xfaa7a586, 0x05400be1, 0xfe3092bf, 0xfff3ad01, 0x003b871f, 0xfff3c69f,
+ 0x433d1c56, 0xf769ea78, 0xfad956ab, 0x053452dc, 0xfe2c7650, 0xfff8064b, 0x003a1ee3, 0xfff3f13a,
+ 0x42a159dc, 0xf71b52c4, 0xfb0aeef6, 0x05284685, 0xfe2892c5, 0xfffc4e5c, 0x0038b7ae, 0xfff41c9c,
+ 0x42045865, 0xf6ce7b57, 0xfb3c6a73, 0x051be8dd, 0xfe24e7c3, 0x00008507, 0x003751a7, 0xfff448b7,
+ 0x4166200e, 0xf683645a, 0xfb6dc53c, 0x050f3bec, 0xfe2174ec, 0x0004aa1f, 0x0035ecf4, 0xfff4757e,
+ 0x40c6b8fd, 0xf63a0ddf, 0xfb9efb77, 0x050241b6, 0xfe1e39da, 0x0008bd7c, 0x003489b9, 0xfff4a2e5,
+ 0x40262b65, 0xf5f277d9, 0xfbd00956, 0x04f4fc46, 0xfe1b3628, 0x000cbef7, 0x0033281a, 0xfff4d0de,
+ 0x3f847f83, 0xf5aca21f, 0xfc00eb1b, 0x04e76da3, 0xfe18696a, 0x0010ae6e, 0x0031c83a, 0xfff4ff5d,
+ 0x3ee1bda2, 0xf5688c6d, 0xfc319d13, 0x04d997d8, 0xfe15d32f, 0x00148bbd, 0x00306a3b, 0xfff52e57,
+ 0x3e3dee13, 0xf5263665, 0xfc621b9a, 0x04cb7cf2, 0xfe137304, 0x001856c7, 0x002f0e3f, 0xfff55dbf,
+ 0x3d991932, 0xf4e59f8a, 0xfc926319, 0x04bd1efb, 0xfe114872, 0x001c0f6e, 0x002db466, 0xfff58d89,
+ 0x3cf34766, 0xf4a6c748, 0xfcc27008, 0x04ae8000, 0xfe0f52fc, 0x001fb599, 0x002c5cd0, 0xfff5bdaa,
+ 0x3c4c811c, 0xf469aced, 0xfcf23eec, 0x049fa20f, 0xfe0d9224, 0x0023492f, 0x002b079a, 0xfff5ee17,
+ 0x3ba4cec9, 0xf42e4faf, 0xfd21cc59, 0x04908733, 0xfe0c0567, 0x0026ca1c, 0x0029b4e4, 0xfff61ec5,
+ 0x3afc38eb, 0xf3f4aea6, 0xfd5114f0, 0x0481317a, 0xfe0aac3f, 0x002a384c, 0x002864c9, 0xfff64fa8,
+ 0x3a52c805, 0xf3bcc8d3, 0xfd801564, 0x0471a2ef, 0xfe098622, 0x002d93ae, 0x00271766, 0xfff680b5,
+ 0x39a884a1, 0xf3869d1a, 0xfdaeca73, 0x0461dda0, 0xfe089283, 0x0030dc34, 0x0025ccd7, 0xfff6b1e4,
+ 0x38fd774e, 0xf3522a49, 0xfddd30eb, 0x0451e396, 0xfe07d0d3, 0x003411d2, 0x00248535, 0xfff6e329,
+ 0x3851a8a2, 0xf31f6f0f, 0xfe0b45aa, 0x0441b6dd, 0xfe07407d, 0x0037347d, 0x0023409a, 0xfff7147a,
+ 0x37a52135, 0xf2ee6a07, 0xfe39059b, 0x0431597d, 0xfe06e0eb, 0x003a442e, 0x0021ff1f, 0xfff745cd,
+ 0x36f7e9a4, 0xf2bf19ae, 0xfe666dbc, 0x0420cd80, 0xfe06b184, 0x003d40e0, 0x0020c0dc, 0xfff7771a,
+ 0x364a0a90, 0xf2917c6d, 0xfe937b15, 0x041014eb, 0xfe06b1ac, 0x00402a8e, 0x001f85e6, 0xfff7a857,
+ 0x359b8c9d, 0xf265908f, 0xfec02ac2, 0x03ff31c3, 0xfe06e0c4, 0x00430137, 0x001e4e56, 0xfff7d97a,
+ 0x34ec786f, 0xf23b544b, 0xfeec79ec, 0x03ee260d, 0xfe073e2a, 0x0045c4dd, 0x001d1a3f, 0xfff80a7c,
+ 0x343cd6af, 0xf212c5be, 0xff1865cd, 0x03dcf3ca, 0xfe07c93a, 0x00487582, 0x001be9b7, 0xfff83b52,
+ 0x338cb004, 0xf1ebe2ec, 0xff43ebac, 0x03cb9cf9, 0xfe08814e, 0x004b132b, 0x001abcd0, 0xfff86bf6,
+ 0x32dc0d17, 0xf1c6a9c3, 0xff6f08e4, 0x03ba2398, 0xfe0965bc, 0x004d9dde, 0x0019939d, 0xfff89c60,
+ 0x322af693, 0xf1a3181a, 0xff99badb, 0x03a889a1, 0xfe0a75da, 0x005015a5, 0x00186e31, 0xfff8cc86,
+ 0x3179751f, 0xf1812bb0, 0xffc3ff0c, 0x0396d10c, 0xfe0bb0f9, 0x00527a8a, 0x00174c9c, 0xfff8fc62,
+ 0x30c79163, 0xf160e22d, 0xffedd2fd, 0x0384fbd1, 0xfe0d166b, 0x0054cc9a, 0x00162eef, 0xfff92bec,
+ 0x30155404, 0xf1423924, 0x00173447, 0x03730be0, 0xfe0ea57e, 0x00570be4, 0x00151538, 0xfff95b1e,
+ 0x2f62c5a7, 0xf1252e0f, 0x00402092, 0x0361032a, 0xfe105d7e, 0x00593877, 0x0013ff88, 0xfff989ef,
+ 0x2eafeeed, 0xf109be56, 0x00689598, 0x034ee39b, 0xfe123db6, 0x005b5267, 0x0012edea, 0xfff9b85b,
+ 0x2dfcd873, 0xf0efe748, 0x0090911f, 0x033caf1d, 0xfe144570, 0x005d59c6, 0x0011e06d, 0xfff9e65a,
+ 0x2d498ad3, 0xf0d7a622, 0x00b81102, 0x032a6796, 0xfe1673f2, 0x005f4eac, 0x0010d71d, 0xfffa13e5,
+ 0x2c960ea3, 0xf0c0f808, 0x00df1328, 0x03180ee7, 0xfe18c884, 0x0061312e, 0x000fd205, 0xfffa40f8,
+ 0x2be26c73, 0xf0abda0e, 0x0105958c, 0x0305a6f0, 0xfe1b4268, 0x00630167, 0x000ed130, 0xfffa6d8d,
+ 0x2b2eaccf, 0xf0984931, 0x012b9635, 0x02f3318a, 0xfe1de0e2, 0x0064bf71, 0x000dd4a7, 0xfffa999d,
+ 0x2a7ad83c, 0xf086425a, 0x0151133e, 0x02e0b08d, 0xfe20a335, 0x00666b68, 0x000cdc74, 0xfffac525,
+ 0x29c6f738, 0xf075c260, 0x01760ad1, 0x02ce25ca, 0xfe2388a1, 0x0068056b, 0x000be89f, 0xfffaf01e,
+ 0x2913123c, 0xf066c606, 0x019a7b27, 0x02bb9310, 0xfe269065, 0x00698d98, 0x000af931, 0xfffb1a84,
+ 0x285f31b7, 0xf05949fb, 0x01be628c, 0x02a8fa2a, 0xfe29b9c1, 0x006b0411, 0x000a0e2f, 0xfffb4453,
+ 0x27ab5e12, 0xf04d4ade, 0x01e1bf58, 0x02965cdb, 0xfe2d03f2, 0x006c68f8, 0x000927a0, 0xfffb6d86,
+ 0x26f79fab, 0xf042c539, 0x02048ff8, 0x0283bce6, 0xfe306e35, 0x006dbc71, 0x00084589, 0xfffb961a,
+ 0x2643feda, 0xf039b587, 0x0226d2e6, 0x02711c05, 0xfe33f7c7, 0x006efea0, 0x000767f0, 0xfffbbe09,
+ 0x259083eb, 0xf032182f, 0x024886ad, 0x025e7bf0, 0xfe379fe3, 0x00702fae, 0x00068ed8, 0xfffbe552,
+ 0x24dd3721, 0xf02be98a, 0x0269a9e9, 0x024bde5a, 0xfe3b65c4, 0x00714fc0, 0x0005ba46, 0xfffc0bef,
+ 0x242a20b3, 0xf02725dc, 0x028a3b44, 0x023944ee, 0xfe3f48a5, 0x00725f02, 0x0004ea3a, 0xfffc31df,
+ 0x237748cf, 0xf023c95d, 0x02aa397b, 0x0226b156, 0xfe4347c0, 0x00735d9c, 0x00041eb9, 0xfffc571e,
+ 0x22c4b795, 0xf021d031, 0x02c9a359, 0x02142533, 0xfe476250, 0x00744bba, 0x000357c2, 0xfffc7ba9,
+ 0x2212751a, 0xf0213671, 0x02e877b9, 0x0201a223, 0xfe4b978e, 0x0075298a, 0x00029558, 0xfffc9f7e,
+ 0x21608968, 0xf021f823, 0x0306b586, 0x01ef29be, 0xfe4fe6b3, 0x0075f739, 0x0001d779, 0xfffcc29a,
+ 0x20aefc79, 0xf0241140, 0x03245bbc, 0x01dcbd96, 0xfe544efb, 0x0076b4f5, 0x00011e26, 0xfffce4fc,
+ 0x1ffdd63b, 0xf0277db1, 0x03416966, 0x01ca5f37, 0xfe58cf9d, 0x007762f0, 0x0000695e, 0xfffd06a1,
+ 0x1f4d1e8e, 0xf02c3953, 0x035ddd9e, 0x01b81028, 0xfe5d67d4, 0x0078015a, 0xffffb91f, 0xfffd2787,
+ 0x1e9cdd43, 0xf0323ff5, 0x0379b790, 0x01a5d1ea, 0xfe6216db, 0x00789065, 0xffff0d66, 0xfffd47ae,
+ 0x1ded1a1d, 0xf0398d56, 0x0394f674, 0x0193a5f9, 0xfe66dbeb, 0x00791043, 0xfffe6631, 0xfffd6713,
diff --git a/services/audioflinger/AudioResamplerSincUp.h b/services/audioflinger/AudioResamplerSincUp.h
new file mode 100644
index 0000000..fd5367e
--- /dev/null
+++ b/services/audioflinger/AudioResamplerSincUp.h
@@ -0,0 +1,131 @@
+// cmd-line: fir -l 7 -s48000 -c 20478
+
+ 0x6d374bc7, 0x111c6ba0, 0xf3240e61, 0x07d14a38, 0xfc509e64, 0x0139cee9, 0xffc8c866, 0xfffcc300,
+ 0x6d35278a, 0x103e8192, 0xf36b9dfd, 0x07bdfaa5, 0xfc5102d0, 0x013d618d, 0xffc663b9, 0xfffd9592,
+ 0x6d2ebafe, 0x0f62811a, 0xf3b3d8ac, 0x07a9f399, 0xfc51d9a6, 0x0140bea5, 0xffc41212, 0xfffe631e,
+ 0x6d24069d, 0x0e8875ad, 0xf3fcb43e, 0x07953976, 0xfc53216f, 0x0143e67c, 0xffc1d373, 0xffff2b9f,
+ 0x6d150b35, 0x0db06a89, 0xf4462690, 0x077fd0ac, 0xfc54d8ae, 0x0146d965, 0xffbfa7d9, 0xffffef10,
+ 0x6d01c9e3, 0x0cda6ab5, 0xf4902587, 0x0769bdaf, 0xfc56fdda, 0x014997bb, 0xffbd8f40, 0x0000ad6e,
+ 0x6cea4418, 0x0c0680fe, 0xf4daa718, 0x07530501, 0xfc598f60, 0x014c21db, 0xffbb89a1, 0x000166b6,
+ 0x6cce7b97, 0x0b34b7f5, 0xf525a143, 0x073bab28, 0xfc5c8ba5, 0x014e782a, 0xffb996f3, 0x00021ae5,
+ 0x6cae7272, 0x0a6519f4, 0xf5710a17, 0x0723b4b4, 0xfc5ff105, 0x01509b14, 0xffb7b728, 0x0002c9fd,
+ 0x6c8a2b0f, 0x0997b116, 0xf5bcd7b1, 0x070b2639, 0xfc63bdd3, 0x01528b08, 0xffb5ea31, 0x000373fb,
+ 0x6c61a823, 0x08cc873c, 0xf609003f, 0x06f20453, 0xfc67f05a, 0x0154487b, 0xffb42ffc, 0x000418e2,
+ 0x6c34ecb5, 0x0803a60a, 0xf6557a00, 0x06d853a2, 0xfc6c86dd, 0x0155d3e8, 0xffb28876, 0x0004b8b3,
+ 0x6c03fc1c, 0x073d16e7, 0xf6a23b44, 0x06be18cd, 0xfc717f97, 0x01572dcf, 0xffb0f388, 0x00055371,
+ 0x6bced9ff, 0x0678e2fc, 0xf6ef3a6e, 0x06a3587e, 0xfc76d8bc, 0x015856b6, 0xffaf7118, 0x0005e921,
+ 0x6b958a54, 0x05b71332, 0xf73c6df4, 0x06881761, 0xfc7c9079, 0x01594f25, 0xffae010b, 0x000679c5,
+ 0x6b581163, 0x04f7b037, 0xf789cc61, 0x066c5a27, 0xfc82a4f4, 0x015a17ab, 0xffaca344, 0x00070564,
+ 0x6b1673c1, 0x043ac276, 0xf7d74c53, 0x06502583, 0xfc89144d, 0x015ab0db, 0xffab57a1, 0x00078c04,
+ 0x6ad0b652, 0x0380521c, 0xf824e480, 0x06337e2a, 0xfc8fdc9f, 0x015b1b4e, 0xffaa1e02, 0x00080dab,
+ 0x6a86de48, 0x02c86715, 0xf8728bb3, 0x061668d2, 0xfc96fbfc, 0x015b579e, 0xffa8f641, 0x00088a62,
+ 0x6a38f123, 0x0213090c, 0xf8c038d0, 0x05f8ea30, 0xfc9e7074, 0x015b666c, 0xffa7e039, 0x00090230,
+ 0x69e6f4b1, 0x01603f6e, 0xf90de2d1, 0x05db06fc, 0xfca63810, 0x015b485b, 0xffa6dbc0, 0x0009751e,
+ 0x6990ef0b, 0x00b01162, 0xf95b80cb, 0x05bcc3ed, 0xfcae50d6, 0x015afe14, 0xffa5e8ad, 0x0009e337,
+ 0x6936e697, 0x000285d0, 0xf9a909ea, 0x059e25b5, 0xfcb6b8c4, 0x015a8843, 0xffa506d2, 0x000a4c85,
+ 0x68d8e206, 0xff57a35e, 0xf9f67577, 0x057f310a, 0xfcbf6dd8, 0x0159e796, 0xffa43603, 0x000ab112,
+ 0x6876e855, 0xfeaf706f, 0xfa43bad2, 0x055fea9d, 0xfcc86e09, 0x01591cc0, 0xffa3760e, 0x000b10ec,
+ 0x681100c9, 0xfe09f323, 0xfa90d17b, 0x0540571a, 0xfcd1b74c, 0x01582878, 0xffa2c6c2, 0x000b6c1d,
+ 0x67a732f4, 0xfd673159, 0xfaddb10c, 0x05207b2f, 0xfcdb4793, 0x01570b77, 0xffa227ec, 0x000bc2b3,
+ 0x673986ac, 0xfcc730aa, 0xfb2a513b, 0x05005b82, 0xfce51ccb, 0x0155c678, 0xffa19957, 0x000c14bb,
+ 0x66c80413, 0xfc29f670, 0xfb76a9dd, 0x04dffcb6, 0xfcef34e1, 0x01545a3c, 0xffa11acb, 0x000c6244,
+ 0x6652b392, 0xfb8f87bd, 0xfbc2b2e4, 0x04bf6369, 0xfcf98dbe, 0x0152c783, 0xffa0ac11, 0x000cab5c,
+ 0x65d99dd5, 0xfaf7e963, 0xfc0e6461, 0x049e9433, 0xfd04254a, 0x01510f13, 0xffa04cf0, 0x000cf012,
+ 0x655ccbd3, 0xfa631fef, 0xfc59b685, 0x047d93a8, 0xfd0ef969, 0x014f31b2, 0xff9ffd2c, 0x000d3075,
+ 0x64dc46c3, 0xf9d12fab, 0xfca4a19f, 0x045c6654, 0xfd1a0801, 0x014d3029, 0xff9fbc89, 0x000d6c97,
+ 0x64581823, 0xf9421c9d, 0xfcef1e20, 0x043b10bd, 0xfd254ef4, 0x014b0b45, 0xff9f8ac9, 0x000da486,
+ 0x63d049b4, 0xf8b5ea87, 0xfd392498, 0x04199760, 0xfd30cc24, 0x0148c3d2, 0xff9f67ae, 0x000dd854,
+ 0x6344e578, 0xf82c9ce7, 0xfd82adba, 0x03f7feb4, 0xfd3c7d73, 0x01465a9f, 0xff9f52f7, 0x000e0812,
+ 0x62b5f5b2, 0xf7a636fa, 0xfdcbb25a, 0x03d64b27, 0xfd4860c2, 0x0143d07f, 0xff9f4c65, 0x000e33d3,
+ 0x622384e8, 0xf722bbb5, 0xfe142b6e, 0x03b4811d, 0xfd5473f3, 0x01412643, 0xff9f53b4, 0x000e5ba7,
+ 0x618d9ddc, 0xf6a22dcf, 0xfe5c120f, 0x0392a4f4, 0xfd60b4e7, 0x013e5cc0, 0xff9f68a1, 0x000e7fa1,
+ 0x60f44b91, 0xf6248fb6, 0xfea35f79, 0x0370bafc, 0xfd6d2180, 0x013b74ca, 0xff9f8ae9, 0x000e9fd5,
+ 0x60579947, 0xf5a9e398, 0xfeea0d0c, 0x034ec77f, 0xfd79b7a1, 0x01386f3a, 0xff9fba47, 0x000ebc54,
+ 0x5fb79278, 0xf5322b61, 0xff30144a, 0x032ccebb, 0xfd86752e, 0x01354ce7, 0xff9ff674, 0x000ed533,
+ 0x5f1442dc, 0xf4bd68b6, 0xff756edc, 0x030ad4e1, 0xfd93580d, 0x01320ea9, 0xffa03f2b, 0x000eea84,
+ 0x5e6db665, 0xf44b9cfe, 0xffba168d, 0x02e8de19, 0xfda05e23, 0x012eb55a, 0xffa09425, 0x000efc5c,
+ 0x5dc3f93c, 0xf3dcc959, 0xfffe054e, 0x02c6ee7f, 0xfdad855b, 0x012b41d3, 0xffa0f519, 0x000f0ace,
+ 0x5d1717c4, 0xf370eea9, 0x00413536, 0x02a50a22, 0xfdbacb9e, 0x0127b4f1, 0xffa161bf, 0x000f15ef,
+ 0x5c671e96, 0xf3080d8c, 0x0083a081, 0x02833506, 0xfdc82edb, 0x01240f8e, 0xffa1d9cf, 0x000f1dd2,
+ 0x5bb41a80, 0xf2a2265e, 0x00c54190, 0x02617321, 0xfdd5ad01, 0x01205285, 0xffa25cfe, 0x000f228d,
+ 0x5afe1886, 0xf23f393b, 0x010612eb, 0x023fc85c, 0xfde34403, 0x011c7eb2, 0xffa2eb04, 0x000f2434,
+ 0x5a4525df, 0xf1df45fd, 0x01460f41, 0x021e3891, 0xfdf0f1d6, 0x011894f0, 0xffa38395, 0x000f22dc,
+ 0x59894ff3, 0xf1824c3e, 0x01853165, 0x01fcc78f, 0xfdfeb475, 0x0114961b, 0xffa42668, 0x000f1e99,
+ 0x58caa45b, 0xf1284b58, 0x01c37452, 0x01db7914, 0xfe0c89db, 0x0110830f, 0xffa4d332, 0x000f1781,
+ 0x580930e1, 0xf0d14267, 0x0200d32c, 0x01ba50d2, 0xfe1a7009, 0x010c5ca6, 0xffa589a6, 0x000f0da8,
+ 0x5745037c, 0xf07d3043, 0x023d493c, 0x0199526b, 0xfe286505, 0x010823ba, 0xffa6497c, 0x000f0125,
+ 0x567e2a51, 0xf02c138a, 0x0278d1f2, 0x01788170, 0xfe3666d5, 0x0103d927, 0xffa71266, 0x000ef20b,
+ 0x55b4b3af, 0xefddea9a, 0x02b368e6, 0x0157e166, 0xfe447389, 0x00ff7dc4, 0xffa7e41a, 0x000ee070,
+ 0x54e8ae13, 0xef92b393, 0x02ed09d7, 0x013775bf, 0xfe528931, 0x00fb126b, 0xffa8be4c, 0x000ecc69,
+ 0x541a281e, 0xef4a6c58, 0x0325b0ad, 0x011741df, 0xfe60a5e5, 0x00f697f3, 0xffa9a0b1, 0x000eb60b,
+ 0x5349309e, 0xef051290, 0x035d5977, 0x00f7491a, 0xfe6ec7c0, 0x00f20f32, 0xffaa8afe, 0x000e9d6b,
+ 0x5275d684, 0xeec2a3a3, 0x0394006a, 0x00d78eb3, 0xfe7cece2, 0x00ed78ff, 0xffab7ce7, 0x000e829e,
+ 0x51a028e8, 0xee831cc3, 0x03c9a1e5, 0x00b815da, 0xfe8b1373, 0x00e8d62d, 0xffac7621, 0x000e65ba,
+ 0x50c83704, 0xee467ae1, 0x03fe3a6f, 0x0098e1b3, 0xfe99399f, 0x00e4278f, 0xffad7662, 0x000e46d3,
+ 0x4fee1037, 0xee0cbab9, 0x0431c6b5, 0x0079f54c, 0xfea75d97, 0x00df6df7, 0xffae7d5f, 0x000e25fd,
+ 0x4f11c3fe, 0xedd5d8ca, 0x0464438c, 0x005b53a4, 0xfeb57d92, 0x00daaa34, 0xffaf8acd, 0x000e034f,
+ 0x4e3361f7, 0xeda1d15c, 0x0495adf2, 0x003cffa9, 0xfec397cf, 0x00d5dd16, 0xffb09e63, 0x000ddedb,
+ 0x4d52f9df, 0xed70a07d, 0x04c6030d, 0x001efc35, 0xfed1aa92, 0x00d10769, 0xffb1b7d8, 0x000db8b7,
+ 0x4c709b8e, 0xed424205, 0x04f54029, 0x00014c12, 0xfedfb425, 0x00cc29f7, 0xffb2d6e1, 0x000d90f6,
+ 0x4b8c56f8, 0xed16b196, 0x052362ba, 0xffe3f1f7, 0xfeedb2da, 0x00c7458a, 0xffb3fb37, 0x000d67ae,
+ 0x4aa63c2c, 0xecedea99, 0x0550685d, 0xffc6f08a, 0xfefba508, 0x00c25ae8, 0xffb52490, 0x000d3cf1,
+ 0x49be5b50, 0xecc7e845, 0x057c4ed4, 0xffaa4a5d, 0xff09890f, 0x00bd6ad7, 0xffb652a7, 0x000d10d5,
+ 0x48d4c4a2, 0xeca4a59b, 0x05a7140b, 0xff8e01f1, 0xff175d53, 0x00b87619, 0xffb78533, 0x000ce36b,
+ 0x47e98874, 0xec841d68, 0x05d0b612, 0xff7219b3, 0xff252042, 0x00b37d70, 0xffb8bbed, 0x000cb4c8,
+ 0x46fcb72d, 0xec664a48, 0x05f93324, 0xff5693fe, 0xff32d04f, 0x00ae8198, 0xffb9f691, 0x000c84ff,
+ 0x460e6148, 0xec4b26a2, 0x0620899e, 0xff3b731b, 0xff406bf8, 0x00a9834e, 0xffbb34d8, 0x000c5422,
+ 0x451e9750, 0xec32acb0, 0x0646b808, 0xff20b93e, 0xff4df1be, 0x00a4834c, 0xffbc767f, 0x000c2245,
+ 0x442d69de, 0xec1cd677, 0x066bbd0d, 0xff066889, 0xff5b602c, 0x009f8249, 0xffbdbb42, 0x000bef79,
+ 0x433ae99c, 0xec099dcf, 0x068f9781, 0xfeec830d, 0xff68b5d5, 0x009a80f8, 0xffbf02dd, 0x000bbbd2,
+ 0x4247273f, 0xebf8fc64, 0x06b2465b, 0xfed30ac5, 0xff75f153, 0x0095800c, 0xffc04d0f, 0x000b8760,
+ 0x41523389, 0xebeaebaf, 0x06d3c8bb, 0xfeba0199, 0xff831148, 0x00908034, 0xffc19996, 0x000b5235,
+ 0x405c1f43, 0xebdf6500, 0x06f41de3, 0xfea16960, 0xff90145e, 0x008b821b, 0xffc2e832, 0x000b1c64,
+ 0x3f64fb40, 0xebd6617b, 0x0713453d, 0xfe8943dc, 0xff9cf947, 0x0086866b, 0xffc438a3, 0x000ae5fc,
+ 0x3e6cd85b, 0xebcfda19, 0x07313e56, 0xfe7192bd, 0xffa9bebe, 0x00818dcb, 0xffc58aaa, 0x000aaf0f,
+ 0x3d73c772, 0xebcbc7a7, 0x074e08e0, 0xfe5a579d, 0xffb66386, 0x007c98de, 0xffc6de09, 0x000a77ac,
+ 0x3c79d968, 0xebca22cc, 0x0769a4b2, 0xfe439407, 0xffc2e669, 0x0077a845, 0xffc83285, 0x000a3fe5,
+ 0x3b7f1f23, 0xebcae405, 0x078411c7, 0xfe2d496f, 0xffcf463a, 0x0072bc9d, 0xffc987e0, 0x000a07c9,
+ 0x3a83a989, 0xebce03aa, 0x079d503b, 0xfe177937, 0xffdb81d6, 0x006dd680, 0xffcadde1, 0x0009cf67,
+ 0x3987897f, 0xebd379eb, 0x07b56051, 0xfe0224b0, 0xffe79820, 0x0068f687, 0xffcc344c, 0x000996ce,
+ 0x388acfe9, 0xebdb3ed5, 0x07cc426c, 0xfded4d13, 0xfff38806, 0x00641d44, 0xffcd8aeb, 0x00095e0e,
+ 0x378d8da8, 0xebe54a4f, 0x07e1f712, 0xfdd8f38b, 0xffff507b, 0x005f4b4a, 0xffcee183, 0x00092535,
+ 0x368fd397, 0xebf1941f, 0x07f67eec, 0xfdc5192d, 0x000af07f, 0x005a8125, 0xffd037e0, 0x0008ec50,
+ 0x3591b28b, 0xec0013e8, 0x0809dac3, 0xfdb1befc, 0x00166718, 0x0055bf60, 0xffd18dcc, 0x0008b36e,
+ 0x34933b50, 0xec10c12c, 0x081c0b84, 0xfd9ee5e7, 0x0021b355, 0x00510682, 0xffd2e311, 0x00087a9c,
+ 0x33947eab, 0xec23934f, 0x082d1239, 0xfd8c8ecc, 0x002cd44d, 0x004c570f, 0xffd4377d, 0x000841e8,
+ 0x32958d55, 0xec388194, 0x083cf010, 0xfd7aba74, 0x0037c922, 0x0047b186, 0xffd58ade, 0x0008095d,
+ 0x319677fa, 0xec4f8322, 0x084ba654, 0xfd696998, 0x004290fc, 0x00431666, 0xffd6dd02, 0x0007d108,
+ 0x30974f3b, 0xec688f02, 0x08593671, 0xfd589cdc, 0x004d2b0e, 0x003e8628, 0xffd82dba, 0x000798f5,
+ 0x2f9823a8, 0xec839c22, 0x0865a1f1, 0xfd4854d3, 0x00579691, 0x003a0141, 0xffd97cd6, 0x00076130,
+ 0x2e9905c1, 0xeca0a156, 0x0870ea7e, 0xfd3891fd, 0x0061d2ca, 0x00358824, 0xffdaca2a, 0x000729c4,
+ 0x2d9a05f4, 0xecbf9558, 0x087b11de, 0xfd2954c8, 0x006bdf05, 0x00311b41, 0xffdc1588, 0x0006f2bb,
+ 0x2c9b349e, 0xece06ecb, 0x088419f6, 0xfd1a9d91, 0x0075ba95, 0x002cbb03, 0xffdd5ec6, 0x0006bc21,
+ 0x2b9ca203, 0xed032439, 0x088c04c8, 0xfd0c6ca2, 0x007f64da, 0x002867d2, 0xffdea5bb, 0x000685ff,
+ 0x2a9e5e57, 0xed27ac16, 0x0892d470, 0xfcfec233, 0x0088dd38, 0x00242213, 0xffdfea3c, 0x0006505f,
+ 0x29a079b2, 0xed4dfcc2, 0x08988b2a, 0xfcf19e6b, 0x0092231e, 0x001fea27, 0xffe12c22, 0x00061b4b,
+ 0x28a30416, 0xed760c88, 0x089d2b4a, 0xfce50161, 0x009b3605, 0x001bc06b, 0xffe26b48, 0x0005e6cb,
+ 0x27a60d6a, 0xed9fd1a2, 0x08a0b740, 0xfcd8eb17, 0x00a4156b, 0x0017a53b, 0xffe3a788, 0x0005b2e8,
+ 0x26a9a57b, 0xedcb4237, 0x08a33196, 0xfccd5b82, 0x00acc0da, 0x001398ec, 0xffe4e0bf, 0x00057faa,
+ 0x25addbf9, 0xedf8545b, 0x08a49cf0, 0xfcc25285, 0x00b537e1, 0x000f9bd2, 0xffe616c8, 0x00054d1a,
+ 0x24b2c075, 0xee26fe17, 0x08a4fc0d, 0xfcb7cff0, 0x00bd7a1c, 0x000bae3c, 0xffe74984, 0x00051b3e,
+ 0x23b86263, 0xee573562, 0x08a451c0, 0xfcadd386, 0x00c5872a, 0x0007d075, 0xffe878d3, 0x0004ea1d,
+ 0x22bed116, 0xee88f026, 0x08a2a0f8, 0xfca45cf7, 0x00cd5eb7, 0x000402c8, 0xffe9a494, 0x0004b9c0,
+ 0x21c61bc0, 0xeebc2444, 0x089fecbb, 0xfc9b6be5, 0x00d50075, 0x00004579, 0xffeaccaa, 0x00048a2b,
+ 0x20ce516f, 0xeef0c78d, 0x089c3824, 0xfc92ffe1, 0x00dc6c1e, 0xfffc98c9, 0xffebf0fa, 0x00045b65,
+ 0x1fd7810f, 0xef26cfca, 0x08978666, 0xfc8b186d, 0x00e3a175, 0xfff8fcf7, 0xffed1166, 0x00042d74,
+ 0x1ee1b965, 0xef5e32bd, 0x0891dac8, 0xfc83b4fc, 0x00eaa045, 0xfff5723d, 0xffee2dd7, 0x0004005e,
+ 0x1ded0911, 0xef96e61c, 0x088b38a9, 0xfc7cd4f0, 0x00f16861, 0xfff1f8d2, 0xffef4632, 0x0003d426,
+ 0x1cf97e8b, 0xefd0df9a, 0x0883a378, 0xfc76779e, 0x00f7f9a3, 0xffee90eb, 0xfff05a60, 0x0003a8d2,
+ 0x1c072823, 0xf00c14e1, 0x087b1ebc, 0xfc709c4d, 0x00fe53ef, 0xffeb3ab8, 0xfff16a4a, 0x00037e65,
+ 0x1b1613ff, 0xf0487b98, 0x0871ae0d, 0xfc6b4233, 0x0104772e, 0xffe7f666, 0xfff275db, 0x000354e5,
+ 0x1a26501b, 0xf0860962, 0x08675516, 0xfc66687a, 0x010a6353, 0xffe4c41e, 0xfff37d00, 0x00032c54,
+ 0x1937ea47, 0xf0c4b3e0, 0x085c1794, 0xfc620e3d, 0x01101858, 0xffe1a408, 0xfff47fa5, 0x000304b7,
+ 0x184af025, 0xf10470b0, 0x084ff957, 0xfc5e328c, 0x0115963d, 0xffde9646, 0xfff57db8, 0x0002de0e,
+ 0x175f6f2b, 0xf1453571, 0x0842fe3d, 0xfc5ad465, 0x011add0b, 0xffdb9af8, 0xfff67729, 0x0002b85f,
+ 0x1675749e, 0xf186f7c0, 0x08352a35, 0xfc57f2be, 0x011fecd3, 0xffd8b23b, 0xfff76be9, 0x000293aa,
+ 0x158d0d95, 0xf1c9ad40, 0x0826813e, 0xfc558c7c, 0x0124c5ab, 0xffd5dc28, 0xfff85be8, 0x00026ff2,
+ 0x14a646f6, 0xf20d4b92, 0x08170767, 0xfc53a07b, 0x012967b1, 0xffd318d6, 0xfff9471b, 0x00024d39,
+ 0x13c12d73, 0xf251c85d, 0x0806c0cb, 0xfc522d88, 0x012dd30a, 0xffd06858, 0xfffa2d74, 0x00022b7f,
+ 0x12ddcd8f, 0xf297194d, 0x07f5b193, 0xfc513266, 0x013207e4, 0xffcdcabe, 0xfffb0ee9, 0x00020ac7,
+ 0x11fc3395, 0xf2dd3411, 0x07e3ddf7, 0xfc50adcc, 0x01360670, 0xffcb4014, 0xfffbeb70, 0x0001eb10,
+ 0x111c6ba0, 0xf3240e61, 0x07d14a38, 0xfc509e64, 0x0139cee9, 0xffc8c866, 0xfffcc300, 0x0001cc5c,
diff --git a/services/audioflinger/AudioStreamOut.cpp b/services/audioflinger/AudioStreamOut.cpp
new file mode 100644
index 0000000..e6d8f09
--- /dev/null
+++ b/services/audioflinger/AudioStreamOut.cpp
@@ -0,0 +1,117 @@
+/*
+**
+** Copyright 2015, 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 "AudioFlinger"
+//#define LOG_NDEBUG 0
+
+#include <hardware/audio.h>
+#include <utils/Log.h>
+
+#include "AudioHwDevice.h"
+#include "AudioStreamOut.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+AudioStreamOut::AudioStreamOut(AudioHwDevice *dev, audio_output_flags_t flags)
+ : audioHwDev(dev)
+ , stream(NULL)
+ , flags(flags)
+{
+}
+
+audio_hw_device_t* AudioStreamOut::hwDev() const
+{
+ return audioHwDev->hwDevice();
+}
+
+status_t AudioStreamOut::getRenderPosition(uint32_t *frames)
+{
+ if (stream == NULL) {
+ return NO_INIT;
+ }
+ return stream->get_render_position(stream, frames);
+}
+
+status_t AudioStreamOut::getPresentationPosition(uint64_t *frames, struct timespec *timestamp)
+{
+ if (stream == NULL) {
+ return NO_INIT;
+ }
+ return stream->get_presentation_position(stream, frames, timestamp);
+}
+
+status_t AudioStreamOut::open(
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ struct audio_config *config,
+ const char *address)
+{
+ audio_stream_out_t* outStream;
+ int status = hwDev()->open_output_stream(
+ hwDev(),
+ handle,
+ devices,
+ flags,
+ config,
+ &outStream,
+ address);
+ ALOGV("AudioStreamOut::open(), HAL open_output_stream returned "
+ " %p, sampleRate %d, Format %#x, "
+ "channelMask %#x, status %d",
+ outStream,
+ config->sample_rate,
+ config->format,
+ config->channel_mask,
+ status);
+
+ if (status == NO_ERROR) {
+ stream = outStream;
+ }
+
+ return status;
+}
+
+size_t AudioStreamOut::getFrameSize()
+{
+ ALOG_ASSERT(stream != NULL);
+ return audio_stream_out_frame_size(stream);
+}
+
+int AudioStreamOut::flush()
+{
+ ALOG_ASSERT(stream != NULL);
+ if (stream->flush != NULL) {
+ return stream->flush(stream);
+ }
+ return NO_ERROR;
+}
+
+int AudioStreamOut::standby()
+{
+ ALOG_ASSERT(stream != NULL);
+ return stream->common.standby(&stream->common);
+}
+
+ssize_t AudioStreamOut::write(const void* buffer, size_t bytes)
+{
+ ALOG_ASSERT(stream != NULL);
+ return stream->write(stream, buffer, bytes);
+}
+
+} // namespace android
diff --git a/services/audioflinger/AudioStreamOut.h b/services/audioflinger/AudioStreamOut.h
new file mode 100644
index 0000000..e91ca9c
--- /dev/null
+++ b/services/audioflinger/AudioStreamOut.h
@@ -0,0 +1,83 @@
+/*
+**
+** Copyright 2015, 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_STREAM_OUT_H
+#define ANDROID_AUDIO_STREAM_OUT_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <system/audio.h>
+
+#include "AudioStreamOut.h"
+
+namespace android {
+
+class AudioHwDevice;
+
+/**
+ * Managed access to a HAL output stream.
+ */
+class AudioStreamOut {
+public:
+// AudioStreamOut is immutable, so its fields are const.
+// For emphasis, we could also make all pointers to them be "const *",
+// but that would clutter the code unnecessarily.
+ AudioHwDevice * const audioHwDev;
+ audio_stream_out_t *stream;
+ const audio_output_flags_t flags;
+
+ audio_hw_device_t *hwDev() const;
+
+ AudioStreamOut(AudioHwDevice *dev, audio_output_flags_t flags);
+
+ virtual status_t open(
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ struct audio_config *config,
+ const char *address);
+
+ virtual ~AudioStreamOut() { }
+
+ virtual status_t getRenderPosition(uint32_t *frames);
+
+ virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
+
+ /**
+ * Write audio buffer to driver. Returns number of bytes written, or a
+ * negative status_t. If at least one frame was written successfully prior to the error,
+ * it is suggested that the driver return that successful (short) byte count
+ * and then return an error in the subsequent call.
+ *
+ * If set_callback() has previously been called to enable non-blocking mode
+ * the write() is not allowed to block. It must write only the number of
+ * bytes that currently fit in the driver/hardware buffer and then return
+ * this byte count. If this is less than the requested write size the
+ * callback function must be called when more space is available in the
+ * driver/hardware buffer.
+ */
+ virtual ssize_t write(const void *buffer, size_t bytes);
+
+ virtual size_t getFrameSize();
+
+ virtual status_t flush();
+ virtual status_t standby();
+};
+
+} // namespace android
+
+#endif // ANDROID_AUDIO_STREAM_OUT_H
diff --git a/services/audioflinger/Configuration.h b/services/audioflinger/Configuration.h
index 6a8aeb1..845697a 100644
--- a/services/audioflinger/Configuration.h
+++ b/services/audioflinger/Configuration.h
@@ -29,9 +29,8 @@
// uncomment to display CPU load adjusted for CPU frequency
//#define CPU_FREQUENCY_STATISTICS
-// uncomment to enable fast mixer to take performance samples for later statistical analysis
-#define FAST_MIXER_STATISTICS
-// FIXME rename to FAST_THREAD_STATISTICS
+// uncomment to enable fast threads to take performance samples for later statistical analysis
+#define FAST_THREAD_STATISTICS
// uncomment for debugging timing problems related to StateQueue::push()
//#define STATE_QUEUE_DUMP
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index bcaf8ae..8bccb47 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -1953,4 +1953,4 @@
}
}
-}; // namespace android
+} // namespace android
diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
index 0c9b976..9e7e8a4 100644
--- a/services/audioflinger/FastCapture.cpp
+++ b/services/audioflinger/FastCapture.cpp
@@ -29,18 +29,18 @@
namespace android {
-/*static*/ const FastCaptureState FastCapture::initial;
+/*static*/ const FastCaptureState FastCapture::sInitial;
FastCapture::FastCapture() : FastThread(),
- inputSource(NULL), inputSourceGen(0), pipeSink(NULL), pipeSinkGen(0),
- readBuffer(NULL), readBufferState(-1), format(Format_Invalid), sampleRate(0),
- // dummyDumpState
- totalNativeFramesRead(0)
+ mInputSource(NULL), mInputSourceGen(0), mPipeSink(NULL), mPipeSinkGen(0),
+ mReadBuffer(NULL), mReadBufferState(-1), mFormat(Format_Invalid), mSampleRate(0),
+ // mDummyDumpState
+ mTotalNativeFramesRead(0)
{
- previous = &initial;
- current = &initial;
+ mPrevious = &sInitial;
+ mCurrent = &sInitial;
- mDummyDumpState = &dummyDumpState;
+ mDummyDumpState = &mDummyFastCaptureDumpState;
}
FastCapture::~FastCapture()
@@ -63,13 +63,13 @@
void FastCapture::onIdle()
{
- preIdle = *(const FastCaptureState *)current;
- current = &preIdle;
+ mPreIdle = *(const FastCaptureState *)mCurrent;
+ mCurrent = &mPreIdle;
}
void FastCapture::onExit()
{
- delete[] readBuffer;
+ free(mReadBuffer);
}
bool FastCapture::isSubClassCommand(FastThreadState::Command command)
@@ -86,67 +86,67 @@
void FastCapture::onStateChange()
{
- const FastCaptureState * const current = (const FastCaptureState *) this->current;
- const FastCaptureState * const previous = (const FastCaptureState *) this->previous;
- FastCaptureDumpState * const dumpState = (FastCaptureDumpState *) this->dumpState;
+ const FastCaptureState * const current = (const FastCaptureState *) mCurrent;
+ const FastCaptureState * const previous = (const FastCaptureState *) mPrevious;
+ FastCaptureDumpState * const dumpState = (FastCaptureDumpState *) mDumpState;
const size_t frameCount = current->mFrameCount;
bool eitherChanged = false;
// check for change in input HAL configuration
- NBAIO_Format previousFormat = format;
- if (current->mInputSourceGen != inputSourceGen) {
- inputSource = current->mInputSource;
- inputSourceGen = current->mInputSourceGen;
- if (inputSource == NULL) {
- format = Format_Invalid;
- sampleRate = 0;
+ NBAIO_Format previousFormat = mFormat;
+ if (current->mInputSourceGen != mInputSourceGen) {
+ mInputSource = current->mInputSource;
+ mInputSourceGen = current->mInputSourceGen;
+ if (mInputSource == NULL) {
+ mFormat = Format_Invalid;
+ mSampleRate = 0;
} else {
- format = inputSource->format();
- sampleRate = Format_sampleRate(format);
- unsigned channelCount = Format_channelCount(format);
+ mFormat = mInputSource->format();
+ mSampleRate = Format_sampleRate(mFormat);
+ unsigned channelCount = Format_channelCount(mFormat);
ALOG_ASSERT(channelCount == 1 || channelCount == 2);
}
- dumpState->mSampleRate = sampleRate;
+ dumpState->mSampleRate = mSampleRate;
eitherChanged = true;
}
// check for change in pipe
- if (current->mPipeSinkGen != pipeSinkGen) {
- pipeSink = current->mPipeSink;
- pipeSinkGen = current->mPipeSinkGen;
+ if (current->mPipeSinkGen != mPipeSinkGen) {
+ mPipeSink = current->mPipeSink;
+ mPipeSinkGen = current->mPipeSinkGen;
eitherChanged = true;
}
// input source and pipe sink must be compatible
- if (eitherChanged && inputSource != NULL && pipeSink != NULL) {
- ALOG_ASSERT(Format_isEqual(format, pipeSink->format()));
+ if (eitherChanged && mInputSource != NULL && mPipeSink != NULL) {
+ ALOG_ASSERT(Format_isEqual(mFormat, mPipeSink->format()));
}
- if ((!Format_isEqual(format, previousFormat)) || (frameCount != previous->mFrameCount)) {
- // FIXME to avoid priority inversion, don't delete here
- delete[] readBuffer;
- readBuffer = NULL;
- if (frameCount > 0 && sampleRate > 0) {
+ if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) {
+ // FIXME to avoid priority inversion, don't free here
+ free(mReadBuffer);
+ mReadBuffer = NULL;
+ if (frameCount > 0 && mSampleRate > 0) {
// FIXME new may block for unbounded time at internal mutex of the heap
// implementation; it would be better to have normal capture thread allocate for
// us to avoid blocking here and to prevent possible priority inversion
- unsigned channelCount = Format_channelCount(format);
- // FIXME frameSize
- readBuffer = new short[frameCount * channelCount];
- periodNs = (frameCount * 1000000000LL) / sampleRate; // 1.00
- underrunNs = (frameCount * 1750000000LL) / sampleRate; // 1.75
- overrunNs = (frameCount * 500000000LL) / sampleRate; // 0.50
- forceNs = (frameCount * 950000000LL) / sampleRate; // 0.95
- warmupNs = (frameCount * 500000000LL) / sampleRate; // 0.50
+ (void)posix_memalign(&mReadBuffer, 32, frameCount * Format_frameSize(mFormat));
+ mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00
+ mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75
+ mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50
+ mForceNs = (frameCount * 950000000LL) / mSampleRate; // 0.95
+ mWarmupNsMin = (frameCount * 750000000LL) / mSampleRate; // 0.75
+ mWarmupNsMax = (frameCount * 1250000000LL) / mSampleRate; // 1.25
} else {
- periodNs = 0;
- underrunNs = 0;
- overrunNs = 0;
- forceNs = 0;
- warmupNs = 0;
+ mPeriodNs = 0;
+ mUnderrunNs = 0;
+ mOverrunNs = 0;
+ mForceNs = 0;
+ mWarmupNsMin = 0;
+ mWarmupNsMax = LONG_MAX;
}
- readBufferState = -1;
+ mReadBufferState = -1;
dumpState->mFrameCount = frameCount;
}
@@ -154,44 +154,43 @@
void FastCapture::onWork()
{
- const FastCaptureState * const current = (const FastCaptureState *) this->current;
- FastCaptureDumpState * const dumpState = (FastCaptureDumpState *) this->dumpState;
- const FastCaptureState::Command command = this->command;
+ const FastCaptureState * const current = (const FastCaptureState *) mCurrent;
+ FastCaptureDumpState * const dumpState = (FastCaptureDumpState *) mDumpState;
+ const FastCaptureState::Command command = mCommand;
const size_t frameCount = current->mFrameCount;
if ((command & FastCaptureState::READ) /*&& isWarm*/) {
- ALOG_ASSERT(inputSource != NULL);
- ALOG_ASSERT(readBuffer != NULL);
+ ALOG_ASSERT(mInputSource != NULL);
+ ALOG_ASSERT(mReadBuffer != NULL);
dumpState->mReadSequence++;
ATRACE_BEGIN("read");
- ssize_t framesRead = inputSource->read(readBuffer, frameCount,
+ ssize_t framesRead = mInputSource->read(mReadBuffer, frameCount,
AudioBufferProvider::kInvalidPTS);
ATRACE_END();
dumpState->mReadSequence++;
if (framesRead >= 0) {
LOG_ALWAYS_FATAL_IF((size_t) framesRead > frameCount);
- totalNativeFramesRead += framesRead;
- dumpState->mFramesRead = totalNativeFramesRead;
- readBufferState = framesRead;
+ mTotalNativeFramesRead += framesRead;
+ dumpState->mFramesRead = mTotalNativeFramesRead;
+ mReadBufferState = framesRead;
} else {
dumpState->mReadErrors++;
- readBufferState = 0;
+ mReadBufferState = 0;
}
// FIXME rename to attemptedIO
- attemptedWrite = true;
+ mAttemptedWrite = true;
}
if (command & FastCaptureState::WRITE) {
- ALOG_ASSERT(pipeSink != NULL);
- ALOG_ASSERT(readBuffer != NULL);
- if (readBufferState < 0) {
- unsigned channelCount = Format_channelCount(format);
- // FIXME frameSize
- memset(readBuffer, 0, frameCount * channelCount * sizeof(short));
- readBufferState = frameCount;
+ ALOG_ASSERT(mPipeSink != NULL);
+ ALOG_ASSERT(mReadBuffer != NULL);
+ if (mReadBufferState < 0) {
+ unsigned channelCount = Format_channelCount(mFormat);
+ memset(mReadBuffer, 0, frameCount * Format_frameSize(mFormat));
+ mReadBufferState = frameCount;
}
- if (readBufferState > 0) {
- ssize_t framesWritten = pipeSink->write(readBuffer, readBufferState);
+ if (mReadBufferState > 0) {
+ ssize_t framesWritten = mPipeSink->write(mReadBuffer, mReadBufferState);
// FIXME This supports at most one fast capture client.
// To handle multiple clients this could be converted to an array,
// or with a lot more work the control block could be shared by all clients.
@@ -210,13 +209,4 @@
}
}
-FastCaptureDumpState::FastCaptureDumpState() : FastThreadDumpState(),
- mReadSequence(0), mFramesRead(0), mReadErrors(0), mSampleRate(0), mFrameCount(0)
-{
-}
-
-FastCaptureDumpState::~FastCaptureDumpState()
-{
-}
-
} // namespace android
diff --git a/services/audioflinger/FastCapture.h b/services/audioflinger/FastCapture.h
index e535b9d..e258a4d 100644
--- a/services/audioflinger/FastCapture.h
+++ b/services/audioflinger/FastCapture.h
@@ -20,23 +20,12 @@
#include "FastThread.h"
#include "StateQueue.h"
#include "FastCaptureState.h"
+#include "FastCaptureDumpState.h"
namespace android {
typedef StateQueue<FastCaptureState> FastCaptureStateQueue;
-struct FastCaptureDumpState : FastThreadDumpState {
- FastCaptureDumpState();
- /*virtual*/ ~FastCaptureDumpState();
-
- // FIXME by renaming, could pull up many of these to FastThreadDumpState
- uint32_t mReadSequence; // incremented before and after each read()
- uint32_t mFramesRead; // total number of frames read successfully
- uint32_t mReadErrors; // total number of read() errors
- uint32_t mSampleRate;
- size_t mFrameCount;
-};
-
class FastCapture : public FastThread {
public:
@@ -57,19 +46,21 @@
virtual void onStateChange();
virtual void onWork();
- static const FastCaptureState initial;
- FastCaptureState preIdle; // copy of state before we went into idle
+ static const FastCaptureState sInitial;
+
+ FastCaptureState mPreIdle; // copy of state before we went into idle
// FIXME by renaming, could pull up many of these to FastThread
- NBAIO_Source *inputSource;
- int inputSourceGen;
- NBAIO_Sink *pipeSink;
- int pipeSinkGen;
- short *readBuffer;
- ssize_t readBufferState; // number of initialized frames in readBuffer, or -1 to clear
- NBAIO_Format format;
- unsigned sampleRate;
- FastCaptureDumpState dummyDumpState;
- uint32_t totalNativeFramesRead; // copied to dumpState->mFramesRead
+ NBAIO_Source* mInputSource;
+ int mInputSourceGen;
+ NBAIO_Sink* mPipeSink;
+ int mPipeSinkGen;
+ void* mReadBuffer;
+ ssize_t mReadBufferState; // number of initialized frames in readBuffer,
+ // or -1 to clear
+ NBAIO_Format mFormat;
+ unsigned mSampleRate;
+ FastCaptureDumpState mDummyFastCaptureDumpState;
+ uint32_t mTotalNativeFramesRead; // copied to dumpState->mFramesRead
}; // class FastCapture
diff --git a/services/audioflinger/FastCaptureDumpState.cpp b/services/audioflinger/FastCaptureDumpState.cpp
new file mode 100644
index 0000000..53eeba5
--- /dev/null
+++ b/services/audioflinger/FastCaptureDumpState.cpp
@@ -0,0 +1,53 @@
+/*
+ * 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_TAG "FastCaptureDumpState"
+//define LOG_NDEBUG 0
+
+#include "Configuration.h"
+#include <utils/Log.h>
+#include "FastCaptureDumpState.h"
+#include "FastCaptureState.h"
+
+namespace android {
+
+FastCaptureDumpState::FastCaptureDumpState() : FastThreadDumpState(),
+ mReadSequence(0), mFramesRead(0), mReadErrors(0), mSampleRate(0), mFrameCount(0)
+{
+}
+
+FastCaptureDumpState::~FastCaptureDumpState()
+{
+}
+
+void FastCaptureDumpState::dump(int fd) const
+{
+ if (mCommand == FastCaptureState::INITIAL) {
+ dprintf(fd, " FastCapture not initialized\n");
+ return;
+ }
+ double measuredWarmupMs = (mMeasuredWarmupTs.tv_sec * 1000.0) +
+ (mMeasuredWarmupTs.tv_nsec / 1000000.0);
+ double periodSec = (double) mFrameCount / mSampleRate;
+ dprintf(fd, " FastCapture command=%s readSequence=%u framesRead=%u\n"
+ " readErrors=%u sampleRate=%u frameCount=%zu\n"
+ " measuredWarmup=%.3g ms, warmupCycles=%u period=%.2f ms\n",
+ FastCaptureState::commandToString(mCommand), mReadSequence, mFramesRead,
+ mReadErrors, mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles,
+ periodSec * 1e3);
+}
+
+} // android
diff --git a/services/audioflinger/FastCaptureDumpState.h b/services/audioflinger/FastCaptureDumpState.h
new file mode 100644
index 0000000..6f9c4c3
--- /dev/null
+++ b/services/audioflinger/FastCaptureDumpState.h
@@ -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.
+ */
+
+#ifndef ANDROID_AUDIO_FAST_CAPTURE_DUMP_STATE_H
+#define ANDROID_AUDIO_FAST_CAPTURE_DUMP_STATE_H
+
+#include <stdint.h>
+#include "Configuration.h"
+#include "FastThreadDumpState.h"
+
+namespace android {
+
+struct FastCaptureDumpState : FastThreadDumpState {
+ FastCaptureDumpState();
+ /*virtual*/ ~FastCaptureDumpState();
+
+ void dump(int fd) const; // should only be called on a stable copy, not the original
+
+ // FIXME by renaming, could pull up many of these to FastThreadDumpState
+ uint32_t mReadSequence; // incremented before and after each read()
+ uint32_t mFramesRead; // total number of frames read successfully
+ uint32_t mReadErrors; // total number of read() errors
+ uint32_t mSampleRate;
+ size_t mFrameCount;
+};
+
+} // android
+
+#endif // ANDROID_AUDIO_FAST_CAPTURE_DUMP_STATE_H
diff --git a/services/audioflinger/FastCaptureState.cpp b/services/audioflinger/FastCaptureState.cpp
index 1d029b7..c4d5e45 100644
--- a/services/audioflinger/FastCaptureState.cpp
+++ b/services/audioflinger/FastCaptureState.cpp
@@ -27,4 +27,19 @@
{
}
+// static
+const char *FastCaptureState::commandToString(Command command)
+{
+ const char *str = FastThreadState::commandToString(command);
+ if (str != NULL) {
+ return str;
+ }
+ switch (command) {
+ case FastCaptureState::READ: return "READ";
+ case FastCaptureState::WRITE: return "WRITE";
+ case FastCaptureState::READ_WRITE: return "READ_WRITE";
+ }
+ LOG_ALWAYS_FATAL("%s", __func__);
+}
+
} // android
diff --git a/services/audioflinger/FastCaptureState.h b/services/audioflinger/FastCaptureState.h
index 29c865a..9bca2d4 100644
--- a/services/audioflinger/FastCaptureState.h
+++ b/services/audioflinger/FastCaptureState.h
@@ -29,21 +29,23 @@
/*virtual*/ ~FastCaptureState();
// all pointer fields use raw pointers; objects are owned and ref-counted by RecordThread
- NBAIO_Source *mInputSource; // HAL input device, must already be negotiated
+ NBAIO_Source* mInputSource; // HAL input device, must already be negotiated
// FIXME by renaming, could pull up these fields to FastThreadState
int mInputSourceGen; // increment when mInputSource is assigned
- NBAIO_Sink *mPipeSink; // after reading from input source, write to this pipe sink
+ NBAIO_Sink* mPipeSink; // after reading from input source, write to this pipe sink
int mPipeSinkGen; // increment when mPipeSink is assigned
size_t mFrameCount; // number of frames per fast capture buffer
- audio_track_cblk_t *mCblk; // control block for the single fast client, or NULL
+ audio_track_cblk_t* mCblk; // control block for the single fast client, or NULL
// Extends FastThreadState::Command
static const Command
// The following commands also process configuration changes, and can be "or"ed:
- READ = 0x8, // read from input source
- WRITE = 0x10, // write to pipe sink
- READ_WRITE = 0x18; // read from input source and write to pipe sink
+ READ = 0x8, // read from input source
+ WRITE = 0x10, // write to pipe sink
+ READ_WRITE = 0x18; // read from input source and write to pipe sink
+ // never returns NULL; asserts if command is invalid
+ static const char *commandToString(Command command);
}; // struct FastCaptureState
} // namespace android
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 2678cbf..f1cf0aa 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -27,10 +27,11 @@
#include "Configuration.h"
#include <time.h>
+#include <utils/Debug.h>
#include <utils/Log.h>
#include <utils/Trace.h>
#include <system/audio.h>
-#ifdef FAST_MIXER_STATISTICS
+#ifdef FAST_THREAD_STATISTICS
#include <cpustats/CentralTendencyStatistics.h>
#ifdef CPU_FREQUENCY_STATISTICS
#include <cpustats/ThreadCpuUsage.h>
@@ -44,15 +45,15 @@
namespace android {
-/*static*/ const FastMixerState FastMixer::initial;
+/*static*/ const FastMixerState FastMixer::sInitial;
FastMixer::FastMixer() : FastThread(),
- slopNs(0),
- // fastTrackNames
- // generations
- outputSink(NULL),
- outputSinkGen(0),
- mixer(NULL),
+ mSlopNs(0),
+ // mFastTrackNames
+ // mGenerations
+ mOutputSink(NULL),
+ mOutputSinkGen(0),
+ mMixer(NULL),
mSinkBuffer(NULL),
mSinkBufferSize(0),
mSinkChannelCount(FCC_2),
@@ -60,30 +61,30 @@
mMixerBufferSize(0),
mMixerBufferFormat(AUDIO_FORMAT_PCM_16_BIT),
mMixerBufferState(UNDEFINED),
- format(Format_Invalid),
- sampleRate(0),
- fastTracksGen(0),
- totalNativeFramesWritten(0),
+ mFormat(Format_Invalid),
+ mSampleRate(0),
+ mFastTracksGen(0),
+ mTotalNativeFramesWritten(0),
// timestamp
- nativeFramesWrittenButNotPresented(0) // the = 0 is to silence the compiler
+ mNativeFramesWrittenButNotPresented(0) // the = 0 is to silence the compiler
{
- // FIXME pass initial as parameter to base class constructor, and make it static local
- previous = &initial;
- current = &initial;
+ // FIXME pass sInitial as parameter to base class constructor, and make it static local
+ mPrevious = &sInitial;
+ mCurrent = &sInitial;
- mDummyDumpState = &dummyDumpState;
+ mDummyDumpState = &mDummyFastMixerDumpState;
// TODO: Add channel mask to NBAIO_Format.
// We assume that the channel mask must be a valid positional channel mask.
mSinkChannelMask = audio_channel_out_mask_from_count(mSinkChannelCount);
unsigned i;
for (i = 0; i < FastMixerState::kMaxFastTracks; ++i) {
- fastTrackNames[i] = -1;
- generations[i] = 0;
+ mFastTrackNames[i] = -1;
+ mGenerations[i] = 0;
}
-#ifdef FAST_MIXER_STATISTICS
- oldLoad.tv_sec = 0;
- oldLoad.tv_nsec = 0;
+#ifdef FAST_THREAD_STATISTICS
+ mOldLoad.tv_sec = 0;
+ mOldLoad.tv_nsec = 0;
#endif
}
@@ -103,20 +104,20 @@
void FastMixer::setLog(NBLog::Writer *logWriter)
{
- if (mixer != NULL) {
- mixer->setLog(logWriter);
+ if (mMixer != NULL) {
+ mMixer->setLog(logWriter);
}
}
void FastMixer::onIdle()
{
- preIdle = *(const FastMixerState *)current;
- current = &preIdle;
+ mPreIdle = *(const FastMixerState *)mCurrent;
+ mCurrent = &mPreIdle;
}
void FastMixer::onExit()
{
- delete mixer;
+ delete mMixer;
free(mMixerBuffer);
free(mSinkBuffer);
}
@@ -135,82 +136,84 @@
void FastMixer::onStateChange()
{
- const FastMixerState * const current = (const FastMixerState *) this->current;
- const FastMixerState * const previous = (const FastMixerState *) this->previous;
- FastMixerDumpState * const dumpState = (FastMixerDumpState *) this->dumpState;
+ const FastMixerState * const current = (const FastMixerState *) mCurrent;
+ const FastMixerState * const previous = (const FastMixerState *) mPrevious;
+ FastMixerDumpState * const dumpState = (FastMixerDumpState *) mDumpState;
const size_t frameCount = current->mFrameCount;
// handle state change here, but since we want to diff the state,
- // we're prepared for previous == &initial the first time through
+ // we're prepared for previous == &sInitial the first time through
unsigned previousTrackMask;
// check for change in output HAL configuration
- NBAIO_Format previousFormat = format;
- if (current->mOutputSinkGen != outputSinkGen) {
- outputSink = current->mOutputSink;
- outputSinkGen = current->mOutputSinkGen;
- if (outputSink == NULL) {
- format = Format_Invalid;
- sampleRate = 0;
+ NBAIO_Format previousFormat = mFormat;
+ if (current->mOutputSinkGen != mOutputSinkGen) {
+ mOutputSink = current->mOutputSink;
+ mOutputSinkGen = current->mOutputSinkGen;
+ if (mOutputSink == NULL) {
+ mFormat = Format_Invalid;
+ mSampleRate = 0;
mSinkChannelCount = 0;
mSinkChannelMask = AUDIO_CHANNEL_NONE;
} else {
- format = outputSink->format();
- sampleRate = Format_sampleRate(format);
- mSinkChannelCount = Format_channelCount(format);
+ mFormat = mOutputSink->format();
+ mSampleRate = Format_sampleRate(mFormat);
+ mSinkChannelCount = Format_channelCount(mFormat);
LOG_ALWAYS_FATAL_IF(mSinkChannelCount > AudioMixer::MAX_NUM_CHANNELS);
// TODO: Add channel mask to NBAIO_Format
// We assume that the channel mask must be a valid positional channel mask.
mSinkChannelMask = audio_channel_out_mask_from_count(mSinkChannelCount);
}
- dumpState->mSampleRate = sampleRate;
+ dumpState->mSampleRate = mSampleRate;
}
- if ((!Format_isEqual(format, previousFormat)) || (frameCount != previous->mFrameCount)) {
+ if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) {
// FIXME to avoid priority inversion, don't delete here
- delete mixer;
- mixer = NULL;
+ delete mMixer;
+ mMixer = NULL;
free(mMixerBuffer);
mMixerBuffer = NULL;
free(mSinkBuffer);
mSinkBuffer = NULL;
- if (frameCount > 0 && sampleRate > 0) {
+ if (frameCount > 0 && mSampleRate > 0) {
// FIXME new may block for unbounded time at internal mutex of the heap
// implementation; it would be better to have normal mixer allocate for us
// to avoid blocking here and to prevent possible priority inversion
- mixer = new AudioMixer(frameCount, sampleRate, FastMixerState::kMaxFastTracks);
+ mMixer = new AudioMixer(frameCount, mSampleRate, FastMixerState::kMaxFastTracks);
const size_t mixerFrameSize = mSinkChannelCount
* audio_bytes_per_sample(mMixerBufferFormat);
mMixerBufferSize = mixerFrameSize * frameCount;
(void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
const size_t sinkFrameSize = mSinkChannelCount
- * audio_bytes_per_sample(format.mFormat);
+ * audio_bytes_per_sample(mFormat.mFormat);
if (sinkFrameSize > mixerFrameSize) { // need a sink buffer
mSinkBufferSize = sinkFrameSize * frameCount;
(void)posix_memalign(&mSinkBuffer, 32, mSinkBufferSize);
}
- periodNs = (frameCount * 1000000000LL) / sampleRate; // 1.00
- underrunNs = (frameCount * 1750000000LL) / sampleRate; // 1.75
- overrunNs = (frameCount * 500000000LL) / sampleRate; // 0.50
- forceNs = (frameCount * 950000000LL) / sampleRate; // 0.95
- warmupNs = (frameCount * 500000000LL) / sampleRate; // 0.50
+ mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00
+ mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75
+ mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50
+ mForceNs = (frameCount * 950000000LL) / mSampleRate; // 0.95
+ mWarmupNsMin = (frameCount * 750000000LL) / mSampleRate; // 0.75
+ mWarmupNsMax = (frameCount * 1250000000LL) / mSampleRate; // 1.25
} else {
- periodNs = 0;
- underrunNs = 0;
- overrunNs = 0;
- forceNs = 0;
- warmupNs = 0;
+ mPeriodNs = 0;
+ mUnderrunNs = 0;
+ mOverrunNs = 0;
+ mForceNs = 0;
+ mWarmupNsMin = 0;
+ mWarmupNsMax = LONG_MAX;
}
mMixerBufferState = UNDEFINED;
#if !LOG_NDEBUG
for (unsigned i = 0; i < FastMixerState::kMaxFastTracks; ++i) {
- fastTrackNames[i] = -1;
+ mFastTrackNames[i] = -1;
}
#endif
// we need to reconfigure all active tracks
previousTrackMask = 0;
- fastTracksGen = current->mFastTracksGen - 1;
+ mFastTracksGen = current->mFastTracksGen - 1;
dumpState->mFrameCount = frameCount;
} else {
previousTrackMask = previous->mTrackMask;
@@ -219,7 +222,7 @@
// check for change in active track set
const unsigned currentTrackMask = current->mTrackMask;
dumpState->mTrackMask = currentTrackMask;
- if (current->mFastTracksGen != fastTracksGen) {
+ if (current->mFastTracksGen != mFastTracksGen) {
ALOG_ASSERT(mMixerBuffer != NULL);
int name;
@@ -230,16 +233,16 @@
removedTracks &= ~(1 << i);
const FastTrack* fastTrack = ¤t->mFastTracks[i];
ALOG_ASSERT(fastTrack->mBufferProvider == NULL);
- if (mixer != NULL) {
- name = fastTrackNames[i];
+ if (mMixer != NULL) {
+ name = mFastTrackNames[i];
ALOG_ASSERT(name >= 0);
- mixer->deleteTrackName(name);
+ mMixer->deleteTrackName(name);
}
#if !LOG_NDEBUG
- fastTrackNames[i] = -1;
+ mFastTrackNames[i] = -1;
#endif
// don't reset track dump state, since other side is ignoring it
- generations[i] = fastTrack->mGeneration;
+ mGenerations[i] = fastTrack->mGeneration;
}
// now process added tracks
@@ -249,29 +252,29 @@
addedTracks &= ~(1 << i);
const FastTrack* fastTrack = ¤t->mFastTracks[i];
AudioBufferProvider *bufferProvider = fastTrack->mBufferProvider;
- ALOG_ASSERT(bufferProvider != NULL && fastTrackNames[i] == -1);
- if (mixer != NULL) {
- name = mixer->getTrackName(fastTrack->mChannelMask,
+ ALOG_ASSERT(bufferProvider != NULL && mFastTrackNames[i] == -1);
+ if (mMixer != NULL) {
+ name = mMixer->getTrackName(fastTrack->mChannelMask,
fastTrack->mFormat, AUDIO_SESSION_OUTPUT_MIX);
ALOG_ASSERT(name >= 0);
- fastTrackNames[i] = name;
- mixer->setBufferProvider(name, bufferProvider);
- mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
+ mFastTrackNames[i] = name;
+ mMixer->setBufferProvider(name, bufferProvider);
+ mMixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
(void *)mMixerBuffer);
// newly allocated track names default to full scale volume
- mixer->setParameter(
+ mMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
- mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
+ mMixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
(void *)(uintptr_t)fastTrack->mFormat);
- mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
+ mMixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
(void *)(uintptr_t)fastTrack->mChannelMask);
- mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MIXER_CHANNEL_MASK,
+ mMixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MIXER_CHANNEL_MASK,
(void *)(uintptr_t)mSinkChannelMask);
- mixer->enable(name);
+ mMixer->enable(name);
}
- generations[i] = fastTrack->mGeneration;
+ mGenerations[i] = fastTrack->mGeneration;
}
// finally process (potentially) modified tracks; these use the same slot
@@ -281,38 +284,38 @@
int i = __builtin_ctz(modifiedTracks);
modifiedTracks &= ~(1 << i);
const FastTrack* fastTrack = ¤t->mFastTracks[i];
- if (fastTrack->mGeneration != generations[i]) {
+ if (fastTrack->mGeneration != mGenerations[i]) {
// this track was actually modified
AudioBufferProvider *bufferProvider = fastTrack->mBufferProvider;
ALOG_ASSERT(bufferProvider != NULL);
- if (mixer != NULL) {
- name = fastTrackNames[i];
+ if (mMixer != NULL) {
+ name = mFastTrackNames[i];
ALOG_ASSERT(name >= 0);
- mixer->setBufferProvider(name, bufferProvider);
+ mMixer->setBufferProvider(name, bufferProvider);
if (fastTrack->mVolumeProvider == NULL) {
float f = AudioMixer::UNITY_GAIN_FLOAT;
- mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f);
- mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f);
+ mMixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f);
+ mMixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f);
}
- mixer->setParameter(name, AudioMixer::RESAMPLE,
+ mMixer->setParameter(name, AudioMixer::RESAMPLE,
AudioMixer::REMOVE, NULL);
- mixer->setParameter(
+ mMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
- mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
+ mMixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
(void *)(uintptr_t)fastTrack->mFormat);
- mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
+ mMixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
(void *)(uintptr_t)fastTrack->mChannelMask);
- mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MIXER_CHANNEL_MASK,
+ mMixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MIXER_CHANNEL_MASK,
(void *)(uintptr_t)mSinkChannelMask);
// already enabled
}
- generations[i] = fastTrack->mGeneration;
+ mGenerations[i] = fastTrack->mGeneration;
}
}
- fastTracksGen = current->mFastTracksGen;
+ mFastTracksGen = current->mFastTracksGen;
dumpState->mNumTracks = popcount(currentTrackMask);
}
@@ -320,12 +323,12 @@
void FastMixer::onWork()
{
- const FastMixerState * const current = (const FastMixerState *) this->current;
- FastMixerDumpState * const dumpState = (FastMixerDumpState *) this->dumpState;
- const FastMixerState::Command command = this->command;
+ const FastMixerState * const current = (const FastMixerState *) mCurrent;
+ FastMixerDumpState * const dumpState = (FastMixerDumpState *) mDumpState;
+ const FastMixerState::Command command = mCommand;
const size_t frameCount = current->mFrameCount;
- if ((command & FastMixerState::MIX) && (mixer != NULL) && isWarm) {
+ if ((command & FastMixerState::MIX) && (mMixer != NULL) && mIsWarm) {
ALOG_ASSERT(mMixerBuffer != NULL);
// for each track, update volume and check for underrun
unsigned currentTrackMask = current->mTrackMask;
@@ -335,9 +338,9 @@
const FastTrack* fastTrack = ¤t->mFastTracks[i];
// Refresh the per-track timestamp
- if (timestampStatus == NO_ERROR) {
+ if (mTimestampStatus == NO_ERROR) {
uint32_t trackFramesWrittenButNotPresented =
- nativeFramesWrittenButNotPresented;
+ mNativeFramesWrittenButNotPresented;
uint32_t trackFramesWritten = fastTrack->mBufferProvider->framesReleased();
// Can't provide an AudioTimestamp before first frame presented,
// or during the brief 32-bit wraparound window
@@ -345,20 +348,20 @@
AudioTimestamp perTrackTimestamp;
perTrackTimestamp.mPosition =
trackFramesWritten - trackFramesWrittenButNotPresented;
- perTrackTimestamp.mTime = timestamp.mTime;
+ perTrackTimestamp.mTime = mTimestamp.mTime;
fastTrack->mBufferProvider->onTimestamp(perTrackTimestamp);
}
}
- int name = fastTrackNames[i];
+ int name = mFastTrackNames[i];
ALOG_ASSERT(name >= 0);
if (fastTrack->mVolumeProvider != NULL) {
gain_minifloat_packed_t vlr = fastTrack->mVolumeProvider->getVolumeLR();
float vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
float vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
- mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &vlf);
- mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &vrf);
+ mMixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &vlf);
+ mMixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &vrf);
}
// FIXME The current implementation of framesReady() for fast tracks
// takes a tryLock, which can block
@@ -379,43 +382,44 @@
if (framesReady == 0) {
underruns.mBitFields.mEmpty++;
underruns.mBitFields.mMostRecent = UNDERRUN_EMPTY;
- mixer->disable(name);
+ mMixer->disable(name);
} else {
// allow mixing partial buffer
underruns.mBitFields.mPartial++;
underruns.mBitFields.mMostRecent = UNDERRUN_PARTIAL;
- mixer->enable(name);
+ mMixer->enable(name);
}
} else {
underruns.mBitFields.mFull++;
underruns.mBitFields.mMostRecent = UNDERRUN_FULL;
- mixer->enable(name);
+ mMixer->enable(name);
}
ftDump->mUnderruns = underruns;
ftDump->mFramesReady = framesReady;
}
int64_t pts;
- if (outputSink == NULL || (OK != outputSink->getNextWriteTimestamp(&pts))) {
+ if (mOutputSink == NULL || (OK != mOutputSink->getNextWriteTimestamp(&pts))) {
pts = AudioBufferProvider::kInvalidPTS;
}
// process() is CPU-bound
- mixer->process(pts);
+ mMixer->process(pts);
mMixerBufferState = MIXED;
} else if (mMixerBufferState == MIXED) {
mMixerBufferState = UNDEFINED;
}
//bool didFullWrite = false; // dumpsys could display a count of partial writes
- if ((command & FastMixerState::WRITE) && (outputSink != NULL) && (mMixerBuffer != NULL)) {
+ if ((command & FastMixerState::WRITE) && (mOutputSink != NULL) && (mMixerBuffer != NULL)) {
if (mMixerBufferState == UNDEFINED) {
memset(mMixerBuffer, 0, mMixerBufferSize);
mMixerBufferState = ZEROED;
}
+ // prepare the buffer used to write to sink
void *buffer = mSinkBuffer != NULL ? mSinkBuffer : mMixerBuffer;
- if (format.mFormat != mMixerBufferFormat) { // sink format not the same as mixer format
- memcpy_by_audio_format(buffer, format.mFormat, mMixerBuffer, mMixerBufferFormat,
- frameCount * Format_channelCount(format));
+ if (mFormat.mFormat != mMixerBufferFormat) { // sink format not the same as mixer format
+ memcpy_by_audio_format(buffer, mFormat.mFormat, mMixerBuffer, mMixerBufferFormat,
+ frameCount * Format_channelCount(mFormat));
}
// if non-NULL, then duplicate write() to this non-blocking sink
NBAIO_Sink* teeSink;
@@ -426,252 +430,34 @@
// but this code should be modified to handle both non-blocking and blocking sinks
dumpState->mWriteSequence++;
ATRACE_BEGIN("write");
- ssize_t framesWritten = outputSink->write(buffer, frameCount);
+ ssize_t framesWritten = mOutputSink->write(buffer, frameCount);
ATRACE_END();
dumpState->mWriteSequence++;
if (framesWritten >= 0) {
ALOG_ASSERT((size_t) framesWritten <= frameCount);
- totalNativeFramesWritten += framesWritten;
- dumpState->mFramesWritten = totalNativeFramesWritten;
+ mTotalNativeFramesWritten += framesWritten;
+ dumpState->mFramesWritten = mTotalNativeFramesWritten;
//if ((size_t) framesWritten == frameCount) {
// didFullWrite = true;
//}
} else {
dumpState->mWriteErrors++;
}
- attemptedWrite = true;
+ mAttemptedWrite = true;
// FIXME count # of writes blocked excessively, CPU usage, etc. for dump
- timestampStatus = outputSink->getTimestamp(timestamp);
- if (timestampStatus == NO_ERROR) {
- uint32_t totalNativeFramesPresented = timestamp.mPosition;
- if (totalNativeFramesPresented <= totalNativeFramesWritten) {
- nativeFramesWrittenButNotPresented =
- totalNativeFramesWritten - totalNativeFramesPresented;
+ mTimestampStatus = mOutputSink->getTimestamp(mTimestamp);
+ if (mTimestampStatus == NO_ERROR) {
+ uint32_t totalNativeFramesPresented = mTimestamp.mPosition;
+ if (totalNativeFramesPresented <= mTotalNativeFramesWritten) {
+ mNativeFramesWrittenButNotPresented =
+ mTotalNativeFramesWritten - totalNativeFramesPresented;
} else {
// HAL reported that more frames were presented than were written
- timestampStatus = INVALID_OPERATION;
+ mTimestampStatus = INVALID_OPERATION;
}
}
}
}
-FastMixerDumpState::FastMixerDumpState(
-#ifdef FAST_MIXER_STATISTICS
- uint32_t samplingN
-#endif
- ) : FastThreadDumpState(),
- mWriteSequence(0), mFramesWritten(0),
- mNumTracks(0), mWriteErrors(0),
- mSampleRate(0), mFrameCount(0),
- mTrackMask(0)
-{
-#ifdef FAST_MIXER_STATISTICS
- increaseSamplingN(samplingN);
-#endif
-}
-
-#ifdef FAST_MIXER_STATISTICS
-void FastMixerDumpState::increaseSamplingN(uint32_t samplingN)
-{
- if (samplingN <= mSamplingN || samplingN > kSamplingN || roundup(samplingN) != samplingN) {
- return;
- }
- uint32_t additional = samplingN - mSamplingN;
- // sample arrays aren't accessed atomically with respect to the bounds,
- // so clearing reduces chance for dumpsys to read random uninitialized samples
- memset(&mMonotonicNs[mSamplingN], 0, sizeof(mMonotonicNs[0]) * additional);
- memset(&mLoadNs[mSamplingN], 0, sizeof(mLoadNs[0]) * additional);
-#ifdef CPU_FREQUENCY_STATISTICS
- memset(&mCpukHz[mSamplingN], 0, sizeof(mCpukHz[0]) * additional);
-#endif
- mSamplingN = samplingN;
-}
-#endif
-
-FastMixerDumpState::~FastMixerDumpState()
-{
-}
-
-// helper function called by qsort()
-static int compare_uint32_t(const void *pa, const void *pb)
-{
- uint32_t a = *(const uint32_t *)pa;
- uint32_t b = *(const uint32_t *)pb;
- if (a < b) {
- return -1;
- } else if (a > b) {
- return 1;
- } else {
- return 0;
- }
-}
-
-void FastMixerDumpState::dump(int fd) const
-{
- if (mCommand == FastMixerState::INITIAL) {
- dprintf(fd, " FastMixer not initialized\n");
- return;
- }
-#define COMMAND_MAX 32
- char string[COMMAND_MAX];
- switch (mCommand) {
- case FastMixerState::INITIAL:
- strcpy(string, "INITIAL");
- break;
- case FastMixerState::HOT_IDLE:
- strcpy(string, "HOT_IDLE");
- break;
- case FastMixerState::COLD_IDLE:
- strcpy(string, "COLD_IDLE");
- break;
- case FastMixerState::EXIT:
- strcpy(string, "EXIT");
- break;
- case FastMixerState::MIX:
- strcpy(string, "MIX");
- break;
- case FastMixerState::WRITE:
- strcpy(string, "WRITE");
- break;
- case FastMixerState::MIX_WRITE:
- strcpy(string, "MIX_WRITE");
- break;
- default:
- snprintf(string, COMMAND_MAX, "%d", mCommand);
- break;
- }
- double measuredWarmupMs = (mMeasuredWarmupTs.tv_sec * 1000.0) +
- (mMeasuredWarmupTs.tv_nsec / 1000000.0);
- double mixPeriodSec = (double) mFrameCount / (double) mSampleRate;
- dprintf(fd, " FastMixer command=%s writeSequence=%u framesWritten=%u\n"
- " numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
- " sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n"
- " mixPeriod=%.2f ms\n",
- string, mWriteSequence, mFramesWritten,
- mNumTracks, mWriteErrors, mUnderruns, mOverruns,
- mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles,
- mixPeriodSec * 1e3);
-#ifdef FAST_MIXER_STATISTICS
- // find the interval of valid samples
- uint32_t bounds = mBounds;
- uint32_t newestOpen = bounds & 0xFFFF;
- uint32_t oldestClosed = bounds >> 16;
- uint32_t n = (newestOpen - oldestClosed) & 0xFFFF;
- if (n > mSamplingN) {
- ALOGE("too many samples %u", n);
- n = mSamplingN;
- }
- // statistics for monotonic (wall clock) time, thread raw CPU load in time, CPU clock frequency,
- // and adjusted CPU load in MHz normalized for CPU clock frequency
- CentralTendencyStatistics wall, loadNs;
-#ifdef CPU_FREQUENCY_STATISTICS
- CentralTendencyStatistics kHz, loadMHz;
- uint32_t previousCpukHz = 0;
-#endif
- // Assuming a normal distribution for cycle times, three standard deviations on either side of
- // the mean account for 99.73% of the population. So if we take each tail to be 1/1000 of the
- // sample set, we get 99.8% combined, or close to three standard deviations.
- static const uint32_t kTailDenominator = 1000;
- uint32_t *tail = n >= kTailDenominator ? new uint32_t[n] : NULL;
- // loop over all the samples
- for (uint32_t j = 0; j < n; ++j) {
- size_t i = oldestClosed++ & (mSamplingN - 1);
- uint32_t wallNs = mMonotonicNs[i];
- if (tail != NULL) {
- tail[j] = wallNs;
- }
- wall.sample(wallNs);
- uint32_t sampleLoadNs = mLoadNs[i];
- loadNs.sample(sampleLoadNs);
-#ifdef CPU_FREQUENCY_STATISTICS
- uint32_t sampleCpukHz = mCpukHz[i];
- // skip bad kHz samples
- if ((sampleCpukHz & ~0xF) != 0) {
- kHz.sample(sampleCpukHz >> 4);
- if (sampleCpukHz == previousCpukHz) {
- double megacycles = (double) sampleLoadNs * (double) (sampleCpukHz >> 4) * 1e-12;
- double adjMHz = megacycles / mixPeriodSec; // _not_ wallNs * 1e9
- loadMHz.sample(adjMHz);
- }
- }
- previousCpukHz = sampleCpukHz;
-#endif
- }
- if (n) {
- dprintf(fd, " Simple moving statistics over last %.1f seconds:\n",
- wall.n() * mixPeriodSec);
- dprintf(fd, " wall clock time in ms per mix cycle:\n"
- " mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
- wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6,
- wall.stddev()*1e-6);
- dprintf(fd, " raw CPU load in us per mix cycle:\n"
- " mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
- loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
- loadNs.stddev()*1e-3);
- } else {
- dprintf(fd, " No FastMixer statistics available currently\n");
- }
-#ifdef CPU_FREQUENCY_STATISTICS
- dprintf(fd, " CPU clock frequency in MHz:\n"
- " mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
- kHz.mean()*1e-3, kHz.minimum()*1e-3, kHz.maximum()*1e-3, kHz.stddev()*1e-3);
- dprintf(fd, " adjusted CPU load in MHz (i.e. normalized for CPU clock frequency):\n"
- " mean=%.1f min=%.1f max=%.1f stddev=%.1f\n",
- loadMHz.mean(), loadMHz.minimum(), loadMHz.maximum(), loadMHz.stddev());
-#endif
- if (tail != NULL) {
- qsort(tail, n, sizeof(uint32_t), compare_uint32_t);
- // assume same number of tail samples on each side, left and right
- uint32_t count = n / kTailDenominator;
- CentralTendencyStatistics left, right;
- for (uint32_t i = 0; i < count; ++i) {
- left.sample(tail[i]);
- right.sample(tail[n - (i + 1)]);
- }
- dprintf(fd, " Distribution of mix cycle times in ms for the tails (> ~3 stddev outliers):\n"
- " left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
- " right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
- left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6,
- right.mean()*1e-6, right.minimum()*1e-6, right.maximum()*1e-6,
- right.stddev()*1e-6);
- delete[] tail;
- }
-#endif
- // The active track mask and track states are updated non-atomically.
- // So if we relied on isActive to decide whether to display,
- // then we might display an obsolete track or omit an active track.
- // Instead we always display all tracks, with an indication
- // of whether we think the track is active.
- uint32_t trackMask = mTrackMask;
- dprintf(fd, " Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
- FastMixerState::kMaxFastTracks, trackMask);
- dprintf(fd, " Index Active Full Partial Empty Recent Ready\n");
- for (uint32_t i = 0; i < FastMixerState::kMaxFastTracks; ++i, trackMask >>= 1) {
- bool isActive = trackMask & 1;
- const FastTrackDump *ftDump = &mTracks[i];
- const FastTrackUnderruns& underruns = ftDump->mUnderruns;
- const char *mostRecent;
- switch (underruns.mBitFields.mMostRecent) {
- case UNDERRUN_FULL:
- mostRecent = "full";
- break;
- case UNDERRUN_PARTIAL:
- mostRecent = "partial";
- break;
- case UNDERRUN_EMPTY:
- mostRecent = "empty";
- break;
- default:
- mostRecent = "?";
- break;
- }
- dprintf(fd, " %5u %6s %4u %7u %5u %7s %5zu\n", i, isActive ? "yes" : "no",
- (underruns.mBitFields.mFull) & UNDERRUN_MASK,
- (underruns.mBitFields.mPartial) & UNDERRUN_MASK,
- (underruns.mBitFields.mEmpty) & UNDERRUN_MASK,
- mostRecent, ftDump->mFramesReady);
- }
-}
-
} // namespace android
diff --git a/services/audioflinger/FastMixer.h b/services/audioflinger/FastMixer.h
index fde8c2b..06a68fb 100644
--- a/services/audioflinger/FastMixer.h
+++ b/services/audioflinger/FastMixer.h
@@ -17,11 +17,7 @@
#ifndef ANDROID_AUDIO_FAST_MIXER_H
#define ANDROID_AUDIO_FAST_MIXER_H
-#include <linux/futex.h>
-#include <sys/syscall.h>
-#include <utils/Debug.h>
#include "FastThread.h"
-#include <utils/Thread.h>
#include "StateQueue.h"
#include "FastMixerState.h"
#include "FastMixerDumpState.h"
@@ -52,36 +48,39 @@
virtual void onStateChange();
virtual void onWork();
- // FIXME these former local variables need comments and to be renamed to have "m" prefix
- static const FastMixerState initial;
- FastMixerState preIdle; // copy of state before we went into idle
- long slopNs; // accumulated time we've woken up too early (> 0) or too late (< 0)
- int fastTrackNames[FastMixerState::kMaxFastTracks]; // handles used by mixer to identify tracks
- int generations[FastMixerState::kMaxFastTracks]; // last observed mFastTracks[i].mGeneration
- NBAIO_Sink *outputSink;
- int outputSinkGen;
- AudioMixer* mixer;
+ // FIXME these former local variables need comments
+ static const FastMixerState sInitial;
+
+ FastMixerState mPreIdle; // copy of state before we went into idle
+ long mSlopNs; // accumulated time we've woken up too early (> 0) or too late (< 0)
+ int mFastTrackNames[FastMixerState::kMaxFastTracks];
+ // handles used by mixer to identify tracks
+ int mGenerations[FastMixerState::kMaxFastTracks];
+ // last observed mFastTracks[i].mGeneration
+ NBAIO_Sink* mOutputSink;
+ int mOutputSinkGen;
+ AudioMixer* mMixer;
// mSinkBuffer audio format is stored in format.mFormat.
- void* mSinkBuffer; // used for mixer output format translation
+ void* mSinkBuffer; // used for mixer output format translation
// if sink format is different than mixer output.
- size_t mSinkBufferSize;
- uint32_t mSinkChannelCount;
+ size_t mSinkBufferSize;
+ uint32_t mSinkChannelCount;
audio_channel_mask_t mSinkChannelMask;
- void* mMixerBuffer; // mixer output buffer.
- size_t mMixerBufferSize;
- audio_format_t mMixerBufferFormat; // mixer output format: AUDIO_FORMAT_PCM_(16_BIT|FLOAT).
+ void* mMixerBuffer; // mixer output buffer.
+ size_t mMixerBufferSize;
+ audio_format_t mMixerBufferFormat; // mixer output format: AUDIO_FORMAT_PCM_(16_BIT|FLOAT).
enum {UNDEFINED, MIXED, ZEROED} mMixerBufferState;
- NBAIO_Format format;
- unsigned sampleRate;
- int fastTracksGen;
- FastMixerDumpState dummyDumpState;
- uint32_t totalNativeFramesWritten; // copied to dumpState->mFramesWritten
+ NBAIO_Format mFormat;
+ unsigned mSampleRate;
+ int mFastTracksGen;
+ FastMixerDumpState mDummyFastMixerDumpState;
+ uint32_t mTotalNativeFramesWritten; // copied to dumpState->mFramesWritten
// next 2 fields are valid only when timestampStatus == NO_ERROR
- AudioTimestamp timestamp;
- uint32_t nativeFramesWrittenButNotPresented;
+ AudioTimestamp mTimestamp;
+ uint32_t mNativeFramesWrittenButNotPresented;
}; // class FastMixer
diff --git a/services/audioflinger/FastMixerDumpState.cpp b/services/audioflinger/FastMixerDumpState.cpp
new file mode 100644
index 0000000..b10942b
--- /dev/null
+++ b/services/audioflinger/FastMixerDumpState.cpp
@@ -0,0 +1,199 @@
+/*
+ * 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_TAG "FastMixerDumpState"
+//#define LOG_NDEBUG 0
+
+#include "Configuration.h"
+#ifdef FAST_THREAD_STATISTICS
+#include <cpustats/CentralTendencyStatistics.h>
+#ifdef CPU_FREQUENCY_STATISTICS
+#include <cpustats/ThreadCpuUsage.h>
+#endif
+#endif
+#include <utils/Debug.h>
+#include <utils/Log.h>
+#include "FastMixerDumpState.h"
+
+namespace android {
+
+FastMixerDumpState::FastMixerDumpState() : FastThreadDumpState(),
+ mWriteSequence(0), mFramesWritten(0),
+ mNumTracks(0), mWriteErrors(0),
+ mSampleRate(0), mFrameCount(0),
+ mTrackMask(0)
+{
+}
+
+FastMixerDumpState::~FastMixerDumpState()
+{
+}
+
+// helper function called by qsort()
+static int compare_uint32_t(const void *pa, const void *pb)
+{
+ uint32_t a = *(const uint32_t *)pa;
+ uint32_t b = *(const uint32_t *)pb;
+ if (a < b) {
+ return -1;
+ } else if (a > b) {
+ return 1;
+ } else {
+ return 0;
+ }
+}
+
+void FastMixerDumpState::dump(int fd) const
+{
+ if (mCommand == FastMixerState::INITIAL) {
+ dprintf(fd, " FastMixer not initialized\n");
+ return;
+ }
+ double measuredWarmupMs = (mMeasuredWarmupTs.tv_sec * 1000.0) +
+ (mMeasuredWarmupTs.tv_nsec / 1000000.0);
+ double mixPeriodSec = (double) mFrameCount / mSampleRate;
+ dprintf(fd, " FastMixer command=%s writeSequence=%u framesWritten=%u\n"
+ " numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
+ " sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n"
+ " mixPeriod=%.2f ms\n",
+ FastMixerState::commandToString(mCommand), mWriteSequence, mFramesWritten,
+ mNumTracks, mWriteErrors, mUnderruns, mOverruns,
+ mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles,
+ mixPeriodSec * 1e3);
+#ifdef FAST_THREAD_STATISTICS
+ // find the interval of valid samples
+ uint32_t bounds = mBounds;
+ uint32_t newestOpen = bounds & 0xFFFF;
+ uint32_t oldestClosed = bounds >> 16;
+ uint32_t n = (newestOpen - oldestClosed) & 0xFFFF;
+ if (n > mSamplingN) {
+ ALOGE("too many samples %u", n);
+ n = mSamplingN;
+ }
+ // statistics for monotonic (wall clock) time, thread raw CPU load in time, CPU clock frequency,
+ // and adjusted CPU load in MHz normalized for CPU clock frequency
+ CentralTendencyStatistics wall, loadNs;
+#ifdef CPU_FREQUENCY_STATISTICS
+ CentralTendencyStatistics kHz, loadMHz;
+ uint32_t previousCpukHz = 0;
+#endif
+ // Assuming a normal distribution for cycle times, three standard deviations on either side of
+ // the mean account for 99.73% of the population. So if we take each tail to be 1/1000 of the
+ // sample set, we get 99.8% combined, or close to three standard deviations.
+ static const uint32_t kTailDenominator = 1000;
+ uint32_t *tail = n >= kTailDenominator ? new uint32_t[n] : NULL;
+ // loop over all the samples
+ for (uint32_t j = 0; j < n; ++j) {
+ size_t i = oldestClosed++ & (mSamplingN - 1);
+ uint32_t wallNs = mMonotonicNs[i];
+ if (tail != NULL) {
+ tail[j] = wallNs;
+ }
+ wall.sample(wallNs);
+ uint32_t sampleLoadNs = mLoadNs[i];
+ loadNs.sample(sampleLoadNs);
+#ifdef CPU_FREQUENCY_STATISTICS
+ uint32_t sampleCpukHz = mCpukHz[i];
+ // skip bad kHz samples
+ if ((sampleCpukHz & ~0xF) != 0) {
+ kHz.sample(sampleCpukHz >> 4);
+ if (sampleCpukHz == previousCpukHz) {
+ double megacycles = (double) sampleLoadNs * (double) (sampleCpukHz >> 4) * 1e-12;
+ double adjMHz = megacycles / mixPeriodSec; // _not_ wallNs * 1e9
+ loadMHz.sample(adjMHz);
+ }
+ }
+ previousCpukHz = sampleCpukHz;
+#endif
+ }
+ if (n) {
+ dprintf(fd, " Simple moving statistics over last %.1f seconds:\n",
+ wall.n() * mixPeriodSec);
+ dprintf(fd, " wall clock time in ms per mix cycle:\n"
+ " mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
+ wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6,
+ wall.stddev()*1e-6);
+ dprintf(fd, " raw CPU load in us per mix cycle:\n"
+ " mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
+ loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
+ loadNs.stddev()*1e-3);
+ } else {
+ dprintf(fd, " No FastMixer statistics available currently\n");
+ }
+#ifdef CPU_FREQUENCY_STATISTICS
+ dprintf(fd, " CPU clock frequency in MHz:\n"
+ " mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
+ kHz.mean()*1e-3, kHz.minimum()*1e-3, kHz.maximum()*1e-3, kHz.stddev()*1e-3);
+ dprintf(fd, " adjusted CPU load in MHz (i.e. normalized for CPU clock frequency):\n"
+ " mean=%.1f min=%.1f max=%.1f stddev=%.1f\n",
+ loadMHz.mean(), loadMHz.minimum(), loadMHz.maximum(), loadMHz.stddev());
+#endif
+ if (tail != NULL) {
+ qsort(tail, n, sizeof(uint32_t), compare_uint32_t);
+ // assume same number of tail samples on each side, left and right
+ uint32_t count = n / kTailDenominator;
+ CentralTendencyStatistics left, right;
+ for (uint32_t i = 0; i < count; ++i) {
+ left.sample(tail[i]);
+ right.sample(tail[n - (i + 1)]);
+ }
+ dprintf(fd, " Distribution of mix cycle times in ms for the tails "
+ "(> ~3 stddev outliers):\n"
+ " left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
+ " right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
+ left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6,
+ right.mean()*1e-6, right.minimum()*1e-6, right.maximum()*1e-6,
+ right.stddev()*1e-6);
+ delete[] tail;
+ }
+#endif
+ // The active track mask and track states are updated non-atomically.
+ // So if we relied on isActive to decide whether to display,
+ // then we might display an obsolete track or omit an active track.
+ // Instead we always display all tracks, with an indication
+ // of whether we think the track is active.
+ uint32_t trackMask = mTrackMask;
+ dprintf(fd, " Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
+ FastMixerState::kMaxFastTracks, trackMask);
+ dprintf(fd, " Index Active Full Partial Empty Recent Ready\n");
+ for (uint32_t i = 0; i < FastMixerState::kMaxFastTracks; ++i, trackMask >>= 1) {
+ bool isActive = trackMask & 1;
+ const FastTrackDump *ftDump = &mTracks[i];
+ const FastTrackUnderruns& underruns = ftDump->mUnderruns;
+ const char *mostRecent;
+ switch (underruns.mBitFields.mMostRecent) {
+ case UNDERRUN_FULL:
+ mostRecent = "full";
+ break;
+ case UNDERRUN_PARTIAL:
+ mostRecent = "partial";
+ break;
+ case UNDERRUN_EMPTY:
+ mostRecent = "empty";
+ break;
+ default:
+ mostRecent = "?";
+ break;
+ }
+ dprintf(fd, " %5u %6s %4u %7u %5u %7s %5zu\n", i, isActive ? "yes" : "no",
+ (underruns.mBitFields.mFull) & UNDERRUN_MASK,
+ (underruns.mBitFields.mPartial) & UNDERRUN_MASK,
+ (underruns.mBitFields.mEmpty) & UNDERRUN_MASK,
+ mostRecent, ftDump->mFramesReady);
+ }
+}
+
+} // android
diff --git a/services/audioflinger/FastMixerDumpState.h b/services/audioflinger/FastMixerDumpState.h
index 6a1e4649..ac15e7c 100644
--- a/services/audioflinger/FastMixerDumpState.h
+++ b/services/audioflinger/FastMixerDumpState.h
@@ -17,7 +17,10 @@
#ifndef ANDROID_AUDIO_FAST_MIXER_DUMP_STATE_H
#define ANDROID_AUDIO_FAST_MIXER_DUMP_STATE_H
+#include <stdint.h>
#include "Configuration.h"
+#include "FastThreadDumpState.h"
+#include "FastMixerState.h"
namespace android {
@@ -52,22 +55,12 @@
struct FastTrackDump {
FastTrackDump() : mFramesReady(0) { }
/*virtual*/ ~FastTrackDump() { }
- FastTrackUnderruns mUnderruns;
- size_t mFramesReady; // most recent value only; no long-term statistics kept
+ FastTrackUnderruns mUnderruns;
+ size_t mFramesReady; // most recent value only; no long-term statistics kept
};
-// The FastMixerDumpState keeps a cache of FastMixer statistics that can be logged by dumpsys.
-// Each individual native word-sized field is accessed atomically. But the
-// overall structure is non-atomic, that is there may be an inconsistency between fields.
-// No barriers or locks are used for either writing or reading.
-// Only POD types are permitted, and the contents shouldn't be trusted (i.e. do range checks).
-// It has a different lifetime than the FastMixer, and so it can't be a member of FastMixer.
struct FastMixerDumpState : FastThreadDumpState {
- FastMixerDumpState(
-#ifdef FAST_MIXER_STATISTICS
- uint32_t samplingN = kSamplingNforLowRamDevice
-#endif
- );
+ FastMixerDumpState();
/*virtual*/ ~FastMixerDumpState();
void dump(int fd) const; // should only be called on a stable copy, not the original
@@ -80,14 +73,6 @@
size_t mFrameCount;
uint32_t mTrackMask; // mask of active tracks
FastTrackDump mTracks[FastMixerState::kMaxFastTracks];
-
-#ifdef FAST_MIXER_STATISTICS
- // Compile-time constant for a "low RAM device", must be a power of 2 <= kSamplingN.
- // This value was chosen such that each array uses 1 small page (4 Kbytes).
- static const uint32_t kSamplingNforLowRamDevice = 0x400;
- // Increase sampling window after construction, must be a power of 2 <= kSamplingN
- void increaseSamplingN(uint32_t samplingN);
-#endif
};
} // android
diff --git a/services/audioflinger/FastMixerState.cpp b/services/audioflinger/FastMixerState.cpp
index 3aa8dad..a8c2634 100644
--- a/services/audioflinger/FastMixerState.cpp
+++ b/services/audioflinger/FastMixerState.cpp
@@ -39,4 +39,19 @@
{
}
+// static
+const char *FastMixerState::commandToString(Command command)
+{
+ const char *str = FastThreadState::commandToString(command);
+ if (str != NULL) {
+ return str;
+ }
+ switch (command) {
+ case FastMixerState::MIX: return "MIX";
+ case FastMixerState::WRITE: return "WRITE";
+ case FastMixerState::MIX_WRITE: return "MIX_WRITE";
+ }
+ LOG_ALWAYS_FATAL("%s", __func__);
+}
+
} // namespace android
diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index 661c9ca..916514f 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -73,6 +73,9 @@
// This might be a one-time configuration rather than per-state
NBAIO_Sink* mTeeSink; // if non-NULL, then duplicate write()s to this non-blocking sink
+
+ // never returns NULL; asserts if command is invalid
+ static const char *commandToString(Command command);
}; // struct FastMixerState
} // namespace android
diff --git a/services/audioflinger/FastThread.cpp b/services/audioflinger/FastThread.cpp
index 216dace..5ca579b 100644
--- a/services/audioflinger/FastThread.cpp
+++ b/services/audioflinger/FastThread.cpp
@@ -25,54 +25,58 @@
#include <utils/Log.h>
#include <utils/Trace.h>
#include "FastThread.h"
+#include "FastThreadDumpState.h"
#define FAST_DEFAULT_NS 999999999L // ~1 sec: default time to sleep
#define FAST_HOT_IDLE_NS 1000000L // 1 ms: time to sleep while hot idling
-#define MIN_WARMUP_CYCLES 2 // minimum number of loop cycles to wait for warmup
+#define MIN_WARMUP_CYCLES 2 // minimum number of consecutive in-range loop cycles
+ // to wait for warmup
#define MAX_WARMUP_CYCLES 10 // maximum number of loop cycles to wait for warmup
namespace android {
FastThread::FastThread() : Thread(false /*canCallJava*/),
- // re-initialized to &initial by subclass constructor
- previous(NULL), current(NULL),
- /* oldTs({0, 0}), */
- oldTsValid(false),
- sleepNs(-1),
- periodNs(0),
- underrunNs(0),
- overrunNs(0),
- forceNs(0),
- warmupNs(0),
- // re-initialized to &dummyDumpState by subclass constructor
+ // re-initialized to &sInitial by subclass constructor
+ mPrevious(NULL), mCurrent(NULL),
+ /* mOldTs({0, 0}), */
+ mOldTsValid(false),
+ mSleepNs(-1),
+ mPeriodNs(0),
+ mUnderrunNs(0),
+ mOverrunNs(0),
+ mForceNs(0),
+ mWarmupNsMin(0),
+ mWarmupNsMax(LONG_MAX),
+ // re-initialized to &mDummySubclassDumpState by subclass constructor
mDummyDumpState(NULL),
- dumpState(NULL),
- ignoreNextOverrun(true),
-#ifdef FAST_MIXER_STATISTICS
- // oldLoad
- oldLoadValid(false),
- bounds(0),
- full(false),
- // tcu
+ mDumpState(NULL),
+ mIgnoreNextOverrun(true),
+#ifdef FAST_THREAD_STATISTICS
+ // mOldLoad
+ mOldLoadValid(false),
+ mBounds(0),
+ mFull(false),
+ // mTcu
#endif
- coldGen(0),
- isWarm(false),
- /* measuredWarmupTs({0, 0}), */
- warmupCycles(0),
- // dummyLogWriter
- logWriter(&dummyLogWriter),
- timestampStatus(INVALID_OPERATION),
+ mColdGen(0),
+ mIsWarm(false),
+ /* mMeasuredWarmupTs({0, 0}), */
+ mWarmupCycles(0),
+ mWarmupConsecutiveInRangeCycles(0),
+ // mDummyLogWriter
+ mLogWriter(&mDummyLogWriter),
+ mTimestampStatus(INVALID_OPERATION),
- command(FastThreadState::INITIAL),
+ mCommand(FastThreadState::INITIAL),
#if 0
frameCount(0),
#endif
- attemptedWrite(false)
+ mAttemptedWrite(false)
{
- oldTs.tv_sec = 0;
- oldTs.tv_nsec = 0;
- measuredWarmupTs.tv_sec = 0;
- measuredWarmupTs.tv_nsec = 0;
+ mOldTs.tv_sec = 0;
+ mOldTs.tv_nsec = 0;
+ mMeasuredWarmupTs.tv_sec = 0;
+ mMeasuredWarmupTs.tv_nsec = 0;
}
FastThread::~FastThread()
@@ -84,34 +88,34 @@
for (;;) {
// either nanosleep, sched_yield, or busy wait
- if (sleepNs >= 0) {
- if (sleepNs > 0) {
- ALOG_ASSERT(sleepNs < 1000000000);
- const struct timespec req = {0, sleepNs};
+ if (mSleepNs >= 0) {
+ if (mSleepNs > 0) {
+ ALOG_ASSERT(mSleepNs < 1000000000);
+ const struct timespec req = {0, mSleepNs};
nanosleep(&req, NULL);
} else {
sched_yield();
}
}
// default to long sleep for next cycle
- sleepNs = FAST_DEFAULT_NS;
+ mSleepNs = FAST_DEFAULT_NS;
// poll for state change
const FastThreadState *next = poll();
if (next == NULL) {
// continue to use the default initial state until a real state is available
- // FIXME &initial not available, should save address earlier
- //ALOG_ASSERT(current == &initial && previous == &initial);
- next = current;
+ // FIXME &sInitial not available, should save address earlier
+ //ALOG_ASSERT(mCurrent == &sInitial && previous == &sInitial);
+ next = mCurrent;
}
- command = next->mCommand;
- if (next != current) {
+ mCommand = next->mCommand;
+ if (next != mCurrent) {
// As soon as possible of learning of a new dump area, start using it
- dumpState = next->mDumpState != NULL ? next->mDumpState : mDummyDumpState;
- logWriter = next->mNBLogWriter != NULL ? next->mNBLogWriter : &dummyLogWriter;
- setLog(logWriter);
+ mDumpState = next->mDumpState != NULL ? next->mDumpState : mDummyDumpState;
+ mLogWriter = next->mNBLogWriter != NULL ? next->mNBLogWriter : &mDummyLogWriter;
+ setLog(mLogWriter);
// We want to always have a valid reference to the previous (non-idle) state.
// However, the state queue only guarantees access to current and previous states.
@@ -122,37 +126,38 @@
// non-idle -> idle update previous from copy of current
// idle -> idle don't update previous
// idle -> non-idle don't update previous
- if (!(current->mCommand & FastThreadState::IDLE)) {
- if (command & FastThreadState::IDLE) {
+ if (!(mCurrent->mCommand & FastThreadState::IDLE)) {
+ if (mCommand & FastThreadState::IDLE) {
onIdle();
- oldTsValid = false;
-#ifdef FAST_MIXER_STATISTICS
- oldLoadValid = false;
+ mOldTsValid = false;
+#ifdef FAST_THREAD_STATISTICS
+ mOldLoadValid = false;
#endif
- ignoreNextOverrun = true;
+ mIgnoreNextOverrun = true;
}
- previous = current;
+ mPrevious = mCurrent;
}
- current = next;
+ mCurrent = next;
}
#if !LOG_NDEBUG
next = NULL; // not referenced again
#endif
- dumpState->mCommand = command;
+ mDumpState->mCommand = mCommand;
+ // FIXME what does this comment mean?
// << current, previous, command, dumpState >>
- switch (command) {
+ switch (mCommand) {
case FastThreadState::INITIAL:
case FastThreadState::HOT_IDLE:
- sleepNs = FAST_HOT_IDLE_NS;
+ mSleepNs = FAST_HOT_IDLE_NS;
continue;
case FastThreadState::COLD_IDLE:
// only perform a cold idle command once
// FIXME consider checking previous state and only perform if previous != COLD_IDLE
- if (current->mColdGen != coldGen) {
- int32_t *coldFutexAddr = current->mColdFutexAddr;
+ if (mCurrent->mColdGen != mColdGen) {
+ int32_t *coldFutexAddr = mCurrent->mColdFutexAddr;
ALOG_ASSERT(coldFutexAddr != NULL);
int32_t old = android_atomic_dec(coldFutexAddr);
if (old <= 0) {
@@ -164,41 +169,42 @@
}
// This may be overly conservative; there could be times that the normal mixer
// requests such a brief cold idle that it doesn't require resetting this flag.
- isWarm = false;
- measuredWarmupTs.tv_sec = 0;
- measuredWarmupTs.tv_nsec = 0;
- warmupCycles = 0;
- sleepNs = -1;
- coldGen = current->mColdGen;
-#ifdef FAST_MIXER_STATISTICS
- bounds = 0;
- full = false;
+ mIsWarm = false;
+ mMeasuredWarmupTs.tv_sec = 0;
+ mMeasuredWarmupTs.tv_nsec = 0;
+ mWarmupCycles = 0;
+ mWarmupConsecutiveInRangeCycles = 0;
+ mSleepNs = -1;
+ mColdGen = mCurrent->mColdGen;
+#ifdef FAST_THREAD_STATISTICS
+ mBounds = 0;
+ mFull = false;
#endif
- oldTsValid = !clock_gettime(CLOCK_MONOTONIC, &oldTs);
- timestampStatus = INVALID_OPERATION;
+ mOldTsValid = !clock_gettime(CLOCK_MONOTONIC, &mOldTs);
+ mTimestampStatus = INVALID_OPERATION;
} else {
- sleepNs = FAST_HOT_IDLE_NS;
+ mSleepNs = FAST_HOT_IDLE_NS;
}
continue;
case FastThreadState::EXIT:
onExit();
return false;
default:
- LOG_ALWAYS_FATAL_IF(!isSubClassCommand(command));
+ LOG_ALWAYS_FATAL_IF(!isSubClassCommand(mCommand));
break;
}
// there is a non-idle state available to us; did the state change?
- if (current != previous) {
+ if (mCurrent != mPrevious) {
onStateChange();
#if 1 // FIXME shouldn't need this
// only process state change once
- previous = current;
+ mPrevious = mCurrent;
#endif
}
// do work using current state here
- attemptedWrite = false;
+ mAttemptedWrite = false;
onWork();
// To be exactly periodic, compute the next sleep time based on current time.
@@ -207,13 +213,13 @@
struct timespec newTs;
int rc = clock_gettime(CLOCK_MONOTONIC, &newTs);
if (rc == 0) {
- //logWriter->logTimestamp(newTs);
- if (oldTsValid) {
- time_t sec = newTs.tv_sec - oldTs.tv_sec;
- long nsec = newTs.tv_nsec - oldTs.tv_nsec;
+ //mLogWriter->logTimestamp(newTs);
+ if (mOldTsValid) {
+ time_t sec = newTs.tv_sec - mOldTs.tv_sec;
+ long nsec = newTs.tv_nsec - mOldTs.tv_nsec;
ALOGE_IF(sec < 0 || (sec == 0 && nsec < 0),
"clock_gettime(CLOCK_MONOTONIC) failed: was %ld.%09ld but now %ld.%09ld",
- oldTs.tv_sec, oldTs.tv_nsec, newTs.tv_sec, newTs.tv_nsec);
+ mOldTs.tv_sec, mOldTs.tv_nsec, newTs.tv_sec, newTs.tv_nsec);
if (nsec < 0) {
--sec;
nsec += 1000000000;
@@ -221,62 +227,70 @@
// To avoid an initial underrun on fast tracks after exiting standby,
// do not start pulling data from tracks and mixing until warmup is complete.
// Warmup is considered complete after the earlier of:
- // MIN_WARMUP_CYCLES write() attempts and last one blocks for at least warmupNs
+ // MIN_WARMUP_CYCLES consecutive in-range write() attempts,
+ // where "in-range" means mWarmupNsMin <= cycle time <= mWarmupNsMax
// MAX_WARMUP_CYCLES write() attempts.
// This is overly conservative, but to get better accuracy requires a new HAL API.
- if (!isWarm && attemptedWrite) {
- measuredWarmupTs.tv_sec += sec;
- measuredWarmupTs.tv_nsec += nsec;
- if (measuredWarmupTs.tv_nsec >= 1000000000) {
- measuredWarmupTs.tv_sec++;
- measuredWarmupTs.tv_nsec -= 1000000000;
+ if (!mIsWarm && mAttemptedWrite) {
+ mMeasuredWarmupTs.tv_sec += sec;
+ mMeasuredWarmupTs.tv_nsec += nsec;
+ if (mMeasuredWarmupTs.tv_nsec >= 1000000000) {
+ mMeasuredWarmupTs.tv_sec++;
+ mMeasuredWarmupTs.tv_nsec -= 1000000000;
}
- ++warmupCycles;
- if ((nsec > warmupNs && warmupCycles >= MIN_WARMUP_CYCLES) ||
- (warmupCycles >= MAX_WARMUP_CYCLES)) {
- isWarm = true;
- dumpState->mMeasuredWarmupTs = measuredWarmupTs;
- dumpState->mWarmupCycles = warmupCycles;
+ ++mWarmupCycles;
+ if (mWarmupNsMin <= nsec && nsec <= mWarmupNsMax) {
+ ALOGV("warmup cycle %d in range: %.03f ms", mWarmupCycles, nsec * 1e-9);
+ ++mWarmupConsecutiveInRangeCycles;
+ } else {
+ ALOGV("warmup cycle %d out of range: %.03f ms", mWarmupCycles, nsec * 1e-9);
+ mWarmupConsecutiveInRangeCycles = 0;
+ }
+ if ((mWarmupConsecutiveInRangeCycles >= MIN_WARMUP_CYCLES) ||
+ (mWarmupCycles >= MAX_WARMUP_CYCLES)) {
+ mIsWarm = true;
+ mDumpState->mMeasuredWarmupTs = mMeasuredWarmupTs;
+ mDumpState->mWarmupCycles = mWarmupCycles;
}
}
- sleepNs = -1;
- if (isWarm) {
- if (sec > 0 || nsec > underrunNs) {
+ mSleepNs = -1;
+ if (mIsWarm) {
+ if (sec > 0 || nsec > mUnderrunNs) {
ATRACE_NAME("underrun");
// FIXME only log occasionally
ALOGV("underrun: time since last cycle %d.%03ld sec",
(int) sec, nsec / 1000000L);
- dumpState->mUnderruns++;
- ignoreNextOverrun = true;
- } else if (nsec < overrunNs) {
- if (ignoreNextOverrun) {
- ignoreNextOverrun = false;
+ mDumpState->mUnderruns++;
+ mIgnoreNextOverrun = true;
+ } else if (nsec < mOverrunNs) {
+ if (mIgnoreNextOverrun) {
+ mIgnoreNextOverrun = false;
} else {
// FIXME only log occasionally
ALOGV("overrun: time since last cycle %d.%03ld sec",
(int) sec, nsec / 1000000L);
- dumpState->mOverruns++;
+ mDumpState->mOverruns++;
}
// This forces a minimum cycle time. It:
// - compensates for an audio HAL with jitter due to sample rate conversion
// - works with a variable buffer depth audio HAL that never pulls at a
- // rate < than overrunNs per buffer.
+ // rate < than mOverrunNs per buffer.
// - recovers from overrun immediately after underrun
// It doesn't work with a non-blocking audio HAL.
- sleepNs = forceNs - nsec;
+ mSleepNs = mForceNs - nsec;
} else {
- ignoreNextOverrun = false;
+ mIgnoreNextOverrun = false;
}
}
-#ifdef FAST_MIXER_STATISTICS
- if (isWarm) {
+#ifdef FAST_THREAD_STATISTICS
+ if (mIsWarm) {
// advance the FIFO queue bounds
- size_t i = bounds & (dumpState->mSamplingN - 1);
- bounds = (bounds & 0xFFFF0000) | ((bounds + 1) & 0xFFFF);
- if (full) {
- bounds += 0x10000;
- } else if (!(bounds & (dumpState->mSamplingN - 1))) {
- full = true;
+ size_t i = mBounds & (mDumpState->mSamplingN - 1);
+ mBounds = (mBounds & 0xFFFF0000) | ((mBounds + 1) & 0xFFFF);
+ if (mFull) {
+ mBounds += 0x10000;
+ } else if (!(mBounds & (mDumpState->mSamplingN - 1))) {
+ mFull = true;
}
// compute the delta value of clock_gettime(CLOCK_MONOTONIC)
uint32_t monotonicNs = nsec;
@@ -288,9 +302,9 @@
struct timespec newLoad;
rc = clock_gettime(CLOCK_THREAD_CPUTIME_ID, &newLoad);
if (rc == 0) {
- if (oldLoadValid) {
- sec = newLoad.tv_sec - oldLoad.tv_sec;
- nsec = newLoad.tv_nsec - oldLoad.tv_nsec;
+ if (mOldLoadValid) {
+ sec = newLoad.tv_sec - mOldLoad.tv_sec;
+ nsec = newLoad.tv_nsec - mOldLoad.tv_nsec;
if (nsec < 0) {
--sec;
nsec += 1000000000;
@@ -301,42 +315,42 @@
}
} else {
// first time through the loop
- oldLoadValid = true;
+ mOldLoadValid = true;
}
- oldLoad = newLoad;
+ mOldLoad = newLoad;
}
#ifdef CPU_FREQUENCY_STATISTICS
// get the absolute value of CPU clock frequency in kHz
int cpuNum = sched_getcpu();
- uint32_t kHz = tcu.getCpukHz(cpuNum);
+ uint32_t kHz = mTcu.getCpukHz(cpuNum);
kHz = (kHz << 4) | (cpuNum & 0xF);
#endif
// save values in FIFO queues for dumpsys
// these stores #1, #2, #3 are not atomic with respect to each other,
// or with respect to store #4 below
- dumpState->mMonotonicNs[i] = monotonicNs;
- dumpState->mLoadNs[i] = loadNs;
+ mDumpState->mMonotonicNs[i] = monotonicNs;
+ mDumpState->mLoadNs[i] = loadNs;
#ifdef CPU_FREQUENCY_STATISTICS
- dumpState->mCpukHz[i] = kHz;
+ mDumpState->mCpukHz[i] = kHz;
#endif
// this store #4 is not atomic with respect to stores #1, #2, #3 above, but
// the newest open & oldest closed halves are atomic with respect to each other
- dumpState->mBounds = bounds;
+ mDumpState->mBounds = mBounds;
ATRACE_INT("cycle_ms", monotonicNs / 1000000);
ATRACE_INT("load_us", loadNs / 1000);
}
#endif
} else {
// first time through the loop
- oldTsValid = true;
- sleepNs = periodNs;
- ignoreNextOverrun = true;
+ mOldTsValid = true;
+ mSleepNs = mPeriodNs;
+ mIgnoreNextOverrun = true;
}
- oldTs = newTs;
+ mOldTs = newTs;
} else {
// monotonic clock is broken
- oldTsValid = false;
- sleepNs = periodNs;
+ mOldTsValid = false;
+ mSleepNs = mPeriodNs;
}
} // for (;;)
diff --git a/services/audioflinger/FastThread.h b/services/audioflinger/FastThread.h
index 1330334..2efb6de 100644
--- a/services/audioflinger/FastThread.h
+++ b/services/audioflinger/FastThread.h
@@ -48,42 +48,45 @@
virtual void onStateChange() = 0;
virtual void onWork() = 0;
- // FIXME these former local variables need comments and to be renamed to have an "m" prefix
- const FastThreadState *previous;
- const FastThreadState *current;
- struct timespec oldTs;
- bool oldTsValid;
- long sleepNs; // -1: busy wait, 0: sched_yield, > 0: nanosleep
- long periodNs; // expected period; the time required to render one mix buffer
- long underrunNs; // underrun likely when write cycle is greater than this value
- long overrunNs; // overrun likely when write cycle is less than this value
- long forceNs; // if overrun detected, force the write cycle to take this much time
- long warmupNs; // warmup complete when write cycle is greater than to this value
- FastThreadDumpState *mDummyDumpState;
- FastThreadDumpState *dumpState;
- bool ignoreNextOverrun; // used to ignore initial overrun and first after an underrun
-#ifdef FAST_MIXER_STATISTICS
- struct timespec oldLoad; // previous value of clock_gettime(CLOCK_THREAD_CPUTIME_ID)
- bool oldLoadValid; // whether oldLoad is valid
- uint32_t bounds;
- bool full; // whether we have collected at least mSamplingN samples
+ // FIXME these former local variables need comments
+ const FastThreadState* mPrevious;
+ const FastThreadState* mCurrent;
+ struct timespec mOldTs;
+ bool mOldTsValid;
+ long mSleepNs; // -1: busy wait, 0: sched_yield, > 0: nanosleep
+ long mPeriodNs; // expected period; the time required to render one mix buffer
+ long mUnderrunNs; // underrun likely when write cycle is greater than this value
+ long mOverrunNs; // overrun likely when write cycle is less than this value
+ long mForceNs; // if overrun detected,
+ // force the write cycle to take this much time
+ long mWarmupNsMin; // warmup complete when write cycle is greater than or equal to
+ // this value
+ long mWarmupNsMax; // and less than or equal to this value
+ FastThreadDumpState* mDummyDumpState;
+ FastThreadDumpState* mDumpState;
+ bool mIgnoreNextOverrun; // used to ignore initial overrun and first after an
+ // underrun
+#ifdef FAST_THREAD_STATISTICS
+ struct timespec mOldLoad; // previous value of clock_gettime(CLOCK_THREAD_CPUTIME_ID)
+ bool mOldLoadValid; // whether oldLoad is valid
+ uint32_t mBounds;
+ bool mFull; // whether we have collected at least mSamplingN samples
#ifdef CPU_FREQUENCY_STATISTICS
- ThreadCpuUsage tcu; // for reading the current CPU clock frequency in kHz
+ ThreadCpuUsage mTcu; // for reading the current CPU clock frequency in kHz
#endif
#endif
- unsigned coldGen; // last observed mColdGen
- bool isWarm; // true means ready to mix, false means wait for warmup before mixing
- struct timespec measuredWarmupTs; // how long did it take for warmup to complete
- uint32_t warmupCycles; // counter of number of loop cycles required to warmup
- NBLog::Writer dummyLogWriter;
- NBLog::Writer *logWriter;
- status_t timestampStatus;
+ unsigned mColdGen; // last observed mColdGen
+ bool mIsWarm; // true means ready to mix,
+ // false means wait for warmup before mixing
+ struct timespec mMeasuredWarmupTs; // how long did it take for warmup to complete
+ uint32_t mWarmupCycles; // counter of number of loop cycles during warmup phase
+ uint32_t mWarmupConsecutiveInRangeCycles; // number of consecutive cycles in range
+ NBLog::Writer mDummyLogWriter;
+ NBLog::Writer* mLogWriter;
+ status_t mTimestampStatus;
- FastThreadState::Command command;
-#if 0
- size_t frameCount;
-#endif
- bool attemptedWrite;
+ FastThreadState::Command mCommand;
+ bool mAttemptedWrite;
}; // class FastThread
diff --git a/services/audioflinger/FastThreadDumpState.cpp b/services/audioflinger/FastThreadDumpState.cpp
new file mode 100644
index 0000000..9df5c4c
--- /dev/null
+++ b/services/audioflinger/FastThreadDumpState.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.
+ */
+
+#include "FastThreadDumpState.h"
+
+namespace android {
+
+FastThreadDumpState::FastThreadDumpState() :
+ mCommand(FastThreadState::INITIAL), mUnderruns(0), mOverruns(0),
+ /* mMeasuredWarmupTs({0, 0}), */
+ mWarmupCycles(0)
+#ifdef FAST_THREAD_STATISTICS
+ , mSamplingN(0), mBounds(0)
+#endif
+{
+ mMeasuredWarmupTs.tv_sec = 0;
+ mMeasuredWarmupTs.tv_nsec = 0;
+#ifdef FAST_THREAD_STATISTICS
+ increaseSamplingN(1);
+#endif
+}
+
+FastThreadDumpState::~FastThreadDumpState()
+{
+}
+
+#ifdef FAST_THREAD_STATISTICS
+void FastThreadDumpState::increaseSamplingN(uint32_t samplingN)
+{
+ if (samplingN <= mSamplingN || samplingN > kSamplingN || roundup(samplingN) != samplingN) {
+ return;
+ }
+ uint32_t additional = samplingN - mSamplingN;
+ // sample arrays aren't accessed atomically with respect to the bounds,
+ // so clearing reduces chance for dumpsys to read random uninitialized samples
+ memset(&mMonotonicNs[mSamplingN], 0, sizeof(mMonotonicNs[0]) * additional);
+ memset(&mLoadNs[mSamplingN], 0, sizeof(mLoadNs[0]) * additional);
+#ifdef CPU_FREQUENCY_STATISTICS
+ memset(&mCpukHz[mSamplingN], 0, sizeof(mCpukHz[0]) * additional);
+#endif
+ mSamplingN = samplingN;
+}
+#endif
+
+} // android
diff --git a/services/audioflinger/FastThreadDumpState.h b/services/audioflinger/FastThreadDumpState.h
new file mode 100644
index 0000000..1ce0914
--- /dev/null
+++ b/services/audioflinger/FastThreadDumpState.h
@@ -0,0 +1,72 @@
+/*
+ * 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 ANDROID_AUDIO_FAST_THREAD_DUMP_STATE_H
+#define ANDROID_AUDIO_FAST_THREAD_DUMP_STATE_H
+
+#include "Configuration.h"
+#include "FastThreadState.h"
+
+namespace android {
+
+// The FastThreadDumpState keeps a cache of FastThread statistics that can be logged by dumpsys.
+// Each individual native word-sized field is accessed atomically. But the
+// overall structure is non-atomic, that is there may be an inconsistency between fields.
+// No barriers or locks are used for either writing or reading.
+// Only POD types are permitted, and the contents shouldn't be trusted (i.e. do range checks).
+// It has a different lifetime than the FastThread, and so it can't be a member of FastThread.
+struct FastThreadDumpState {
+ FastThreadDumpState();
+ /*virtual*/ ~FastThreadDumpState();
+
+ FastThreadState::Command mCommand; // current command
+ uint32_t mUnderruns; // total number of underruns
+ uint32_t mOverruns; // total number of overruns
+ struct timespec mMeasuredWarmupTs; // measured warmup time
+ uint32_t mWarmupCycles; // number of loop cycles required to warmup
+
+#ifdef FAST_THREAD_STATISTICS
+ // Recently collected samples of per-cycle monotonic time, thread CPU time, and CPU frequency.
+ // kSamplingN is max size of sampling frame (statistics), and must be a power of 2 <= 0x8000.
+ // The sample arrays are virtually allocated based on this compile-time constant,
+ // but are only initialized and used based on the runtime parameter mSamplingN.
+ static const uint32_t kSamplingN = 0x8000;
+ // Compile-time constant for a "low RAM device", must be a power of 2 <= kSamplingN.
+ // This value was chosen such that each array uses 1 small page (4 Kbytes).
+ static const uint32_t kSamplingNforLowRamDevice = 0x400;
+ // Corresponding runtime maximum size of sample arrays, must be a power of 2 <= kSamplingN.
+ uint32_t mSamplingN;
+ // The bounds define the interval of valid samples, and are represented as follows:
+ // newest open (excluded) endpoint = lower 16 bits of bounds, modulo N
+ // oldest closed (included) endpoint = upper 16 bits of bounds, modulo N
+ // Number of valid samples is newest - oldest.
+ uint32_t mBounds; // bounds for mMonotonicNs, mThreadCpuNs, and mCpukHz
+ // The elements in the *Ns arrays are in units of nanoseconds <= 3999999999.
+ uint32_t mMonotonicNs[kSamplingN]; // delta monotonic (wall clock) time
+ uint32_t mLoadNs[kSamplingN]; // delta CPU load in time
+#ifdef CPU_FREQUENCY_STATISTICS
+ uint32_t mCpukHz[kSamplingN]; // absolute CPU clock frequency in kHz, bits 0-3 are CPU#
+#endif
+
+ // Increase sampling window after construction, must be a power of 2 <= kSamplingN
+ void increaseSamplingN(uint32_t samplingN);
+#endif
+
+}; // struct FastThreadDumpState
+
+} // android
+
+#endif // ANDROID_AUDIO_FAST_THREAD_DUMP_STATE_H
diff --git a/services/audioflinger/FastThreadState.cpp b/services/audioflinger/FastThreadState.cpp
index 6994872..ad5f31f 100644
--- a/services/audioflinger/FastThreadState.cpp
+++ b/services/audioflinger/FastThreadState.cpp
@@ -29,21 +29,16 @@
{
}
-
-FastThreadDumpState::FastThreadDumpState() :
- mCommand(FastThreadState::INITIAL), mUnderruns(0), mOverruns(0),
- /* mMeasuredWarmupTs({0, 0}), */
- mWarmupCycles(0)
-#ifdef FAST_MIXER_STATISTICS
- , mSamplingN(1), mBounds(0)
-#endif
+// static
+const char *FastThreadState::commandToString(FastThreadState::Command command)
{
- mMeasuredWarmupTs.tv_sec = 0;
- mMeasuredWarmupTs.tv_nsec = 0;
-}
-
-FastThreadDumpState::~FastThreadDumpState()
-{
+ switch (command) {
+ case FastThreadState::INITIAL: return "INITIAL";
+ case FastThreadState::HOT_IDLE: return "HOT_IDLE";
+ case FastThreadState::COLD_IDLE: return "COLD_IDLE";
+ case FastThreadState::EXIT: return "EXIT";
+ }
+ return NULL;
}
} // namespace android
diff --git a/services/audioflinger/FastThreadState.h b/services/audioflinger/FastThreadState.h
index 1ab8a0a..f18f846 100644
--- a/services/audioflinger/FastThreadState.h
+++ b/services/audioflinger/FastThreadState.h
@@ -46,43 +46,10 @@
FastThreadDumpState* mDumpState; // if non-NULL, then update dump state periodically
NBLog::Writer* mNBLogWriter; // non-blocking logger
+ // returns NULL if command belongs to a subclass
+ static const char *commandToString(Command command);
}; // struct FastThreadState
-
-// FIXME extract common part of comment at FastMixerDumpState
-struct FastThreadDumpState {
- FastThreadDumpState();
- /*virtual*/ ~FastThreadDumpState();
-
- FastThreadState::Command mCommand; // current command
- uint32_t mUnderruns; // total number of underruns
- uint32_t mOverruns; // total number of overruns
- struct timespec mMeasuredWarmupTs; // measured warmup time
- uint32_t mWarmupCycles; // number of loop cycles required to warmup
-
-#ifdef FAST_MIXER_STATISTICS
- // Recently collected samples of per-cycle monotonic time, thread CPU time, and CPU frequency.
- // kSamplingN is max size of sampling frame (statistics), and must be a power of 2 <= 0x8000.
- // The sample arrays are virtually allocated based on this compile-time constant,
- // but are only initialized and used based on the runtime parameter mSamplingN.
- static const uint32_t kSamplingN = 0x8000;
- // Corresponding runtime maximum size of sample arrays, must be a power of 2 <= kSamplingN.
- uint32_t mSamplingN;
- // The bounds define the interval of valid samples, and are represented as follows:
- // newest open (excluded) endpoint = lower 16 bits of bounds, modulo N
- // oldest closed (included) endpoint = upper 16 bits of bounds, modulo N
- // Number of valid samples is newest - oldest.
- uint32_t mBounds; // bounds for mMonotonicNs, mThreadCpuNs, and mCpukHz
- // The elements in the *Ns arrays are in units of nanoseconds <= 3999999999.
- uint32_t mMonotonicNs[kSamplingN]; // delta monotonic (wall clock) time
- uint32_t mLoadNs[kSamplingN]; // delta CPU load in time
-#ifdef CPU_FREQUENCY_STATISTICS
- uint32_t mCpukHz[kSamplingN]; // absolute CPU clock frequency in kHz, bits 0-3 are CPU#
-#endif
-#endif
-
-}; // struct FastThreadDumpState
-
} // android
#endif // ANDROID_AUDIO_FAST_THREAD_STATE_H
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 4f0c6b1..efbdcff 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -694,4 +694,4 @@
}
-}; // namespace android
+} // namespace android
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index ee48276..45df6a9 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -157,8 +157,9 @@
bool mFlushHwPending; // track requests for thread flush
// for last call to getTimestamp
- bool mPreviousValid;
- uint32_t mPreviousFramesWritten;
+ bool mPreviousTimestampValid;
+ // This is either the first timestamp or one that has passed
+ // the check to prevent retrograde motion.
AudioTimestamp mPreviousTimestamp;
}; // end of Track
@@ -255,7 +256,7 @@
class Buffer : public AudioBufferProvider::Buffer {
public:
- int16_t *mBuffer;
+ void *mBuffer;
};
OutputTrack(PlaybackThread *thread,
@@ -271,7 +272,7 @@
AudioSystem::SYNC_EVENT_NONE,
int triggerSession = 0);
virtual void stop();
- bool write(int16_t* data, uint32_t frames);
+ bool write(void* data, uint32_t frames);
bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
bool isActive() const { return mActive; }
const wp<ThreadBase>& thread() const { return mThread; }
diff --git a/services/audioflinger/SpdifStreamOut.cpp b/services/audioflinger/SpdifStreamOut.cpp
new file mode 100644
index 0000000..d23588e
--- /dev/null
+++ b/services/audioflinger/SpdifStreamOut.cpp
@@ -0,0 +1,166 @@
+/*
+**
+** Copyright 2015, 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 "AudioFlinger"
+//#define LOG_NDEBUG 0
+#include <hardware/audio.h>
+#include <utils/Log.h>
+
+#include <audio_utils/spdif/SPDIFEncoder.h>
+
+#include "AudioHwDevice.h"
+#include "AudioStreamOut.h"
+#include "SpdifStreamOut.h"
+
+namespace android {
+
+/**
+ * If the AudioFlinger is processing encoded data and the HAL expects
+ * PCM then we need to wrap the data in an SPDIF wrapper.
+ */
+SpdifStreamOut::SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags)
+ : AudioStreamOut(dev,flags)
+ , mRateMultiplier(1)
+ , mSpdifEncoder(this)
+ , mRenderPositionHal(0)
+ , mPreviousHalPosition32(0)
+{
+}
+
+status_t SpdifStreamOut::open(
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ struct audio_config *config,
+ const char *address)
+{
+ struct audio_config customConfig = *config;
+
+ customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+ customConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+
+ // Some data bursts run at a higher sample rate.
+ switch(config->format) {
+ case AUDIO_FORMAT_E_AC3:
+ mRateMultiplier = 4;
+ break;
+ case AUDIO_FORMAT_AC3:
+ mRateMultiplier = 1;
+ break;
+ default:
+ ALOGE("ERROR SpdifStreamOut::open() unrecognized format 0x%08X\n",
+ config->format);
+ return BAD_VALUE;
+ }
+ customConfig.sample_rate = config->sample_rate * mRateMultiplier;
+
+ // Always print this because otherwise it could be very confusing if the
+ // HAL and AudioFlinger are using different formats.
+ // Print before open() because HAL may modify customConfig.
+ ALOGI("SpdifStreamOut::open() AudioFlinger requested"
+ " sampleRate %d, format %#x, channelMask %#x",
+ config->sample_rate,
+ config->format,
+ config->channel_mask);
+ ALOGI("SpdifStreamOut::open() HAL configured for"
+ " sampleRate %d, format %#x, channelMask %#x",
+ customConfig.sample_rate,
+ customConfig.format,
+ customConfig.channel_mask);
+
+ status_t status = AudioStreamOut::open(
+ handle,
+ devices,
+ &customConfig,
+ address);
+
+ ALOGI("SpdifStreamOut::open() status = %d", status);
+
+ return status;
+}
+
+// Account for possibly higher sample rate.
+status_t SpdifStreamOut::getRenderPosition(uint32_t *frames)
+{
+ uint32_t halPosition = 0;
+ status_t status = AudioStreamOut::getRenderPosition(&halPosition);
+ if (status != NO_ERROR) {
+ return status;
+ }
+
+ // Accumulate a 64-bit position so that we wrap at the right place.
+ if (mRateMultiplier != 1) {
+ // Maintain a 64-bit render position.
+ int32_t deltaHalPosition = (int32_t)(halPosition - mPreviousHalPosition32);
+ mPreviousHalPosition32 = halPosition;
+ mRenderPositionHal += deltaHalPosition;
+
+ // Scale from device sample rate to application rate.
+ uint64_t renderPositionApp = mRenderPositionHal / mRateMultiplier;
+ ALOGV("SpdifStreamOut::getRenderPosition() "
+ "renderPositionAppRate = %llu = %llu / %u\n",
+ renderPositionApp, mRenderPositionHal, mRateMultiplier);
+
+ *frames = (uint32_t)renderPositionApp;
+ } else {
+ *frames = halPosition;
+ }
+ return status;
+}
+
+int SpdifStreamOut::flush()
+{
+ // FIXME Is there an issue here with flush being asynchronous?
+ mRenderPositionHal = 0;
+ mPreviousHalPosition32 = 0;
+ return AudioStreamOut::flush();
+}
+
+int SpdifStreamOut::standby()
+{
+ mRenderPositionHal = 0;
+ mPreviousHalPosition32 = 0;
+ return AudioStreamOut::standby();
+}
+
+// Account for possibly higher sample rate.
+// This is much easier when all the values are 64-bit.
+status_t SpdifStreamOut::getPresentationPosition(uint64_t *frames,
+ struct timespec *timestamp)
+{
+ uint64_t halFrames = 0;
+ status_t status = AudioStreamOut::getPresentationPosition(&halFrames, timestamp);
+ *frames = halFrames / mRateMultiplier;
+ return status;
+}
+
+size_t SpdifStreamOut::getFrameSize()
+{
+ return sizeof(int8_t);
+}
+
+ssize_t SpdifStreamOut::writeDataBurst(const void* buffer, size_t bytes)
+{
+ return AudioStreamOut::write(buffer, bytes);
+}
+
+ssize_t SpdifStreamOut::write(const void* buffer, size_t bytes)
+{
+ // Write to SPDIF wrapper. It will call back to writeDataBurst().
+ return mSpdifEncoder.write(buffer, bytes);
+}
+
+} // namespace android
diff --git a/services/audioflinger/SpdifStreamOut.h b/services/audioflinger/SpdifStreamOut.h
new file mode 100644
index 0000000..cb82ac7
--- /dev/null
+++ b/services/audioflinger/SpdifStreamOut.h
@@ -0,0 +1,107 @@
+/*
+**
+** Copyright 2015, 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_SPDIF_STREAM_OUT_H
+#define ANDROID_SPDIF_STREAM_OUT_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <system/audio.h>
+
+#include "AudioHwDevice.h"
+#include "AudioStreamOut.h"
+#include "SpdifStreamOut.h"
+
+#include <audio_utils/spdif/SPDIFEncoder.h>
+
+namespace android {
+
+/**
+ * Stream that is a PCM data burst in the HAL but looks like an encoded stream
+ * to the AudioFlinger. Wraps encoded data in an SPDIF wrapper per IEC61973-3.
+ */
+class SpdifStreamOut : public AudioStreamOut {
+public:
+
+ SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags);
+
+ virtual ~SpdifStreamOut() { }
+
+ virtual status_t open(
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ struct audio_config *config,
+ const char *address);
+
+ virtual status_t getRenderPosition(uint32_t *frames);
+
+ virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
+
+ /**
+ * Write audio buffer to driver. Returns number of bytes written, or a
+ * negative status_t. If at least one frame was written successfully prior to the error,
+ * it is suggested that the driver return that successful (short) byte count
+ * and then return an error in the subsequent call.
+ *
+ * If set_callback() has previously been called to enable non-blocking mode
+ * the write() is not allowed to block. It must write only the number of
+ * bytes that currently fit in the driver/hardware buffer and then return
+ * this byte count. If this is less than the requested write size the
+ * callback function must be called when more space is available in the
+ * driver/hardware buffer.
+ */
+ virtual ssize_t write(const void* buffer, size_t bytes);
+
+ virtual size_t getFrameSize();
+
+ virtual status_t flush();
+ virtual status_t standby();
+
+private:
+
+ class MySPDIFEncoder : public SPDIFEncoder
+ {
+ public:
+ MySPDIFEncoder(SpdifStreamOut *spdifStreamOut)
+ : mSpdifStreamOut(spdifStreamOut)
+ {
+ }
+
+ virtual ssize_t writeOutput(const void* buffer, size_t bytes)
+ {
+ return mSpdifStreamOut->writeDataBurst(buffer, bytes);
+ }
+ protected:
+ SpdifStreamOut * const mSpdifStreamOut;
+ };
+
+ int mRateMultiplier;
+ MySPDIFEncoder mSpdifEncoder;
+
+ // Used to implement getRenderPosition()
+ int64_t mRenderPositionHal;
+ uint32_t mPreviousHalPosition32;
+
+ ssize_t writeDataBurst(const void* data, size_t bytes);
+ ssize_t writeInternal(const void* buffer, size_t bytes);
+
+};
+
+} // namespace android
+
+#endif // ANDROID_SPDIF_STREAM_OUT_H
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 51025fe..4efb3d7 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -23,7 +23,9 @@
#include "Configuration.h"
#include <math.h>
#include <fcntl.h>
+#include <linux/futex.h>
#include <sys/stat.h>
+#include <sys/syscall.h>
#include <cutils/properties.h>
#include <media/AudioParameter.h>
#include <media/AudioResamplerPublic.h>
@@ -314,6 +316,165 @@
// ThreadBase
// ----------------------------------------------------------------------------
+// static
+const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type)
+{
+ switch (type) {
+ case MIXER:
+ return "MIXER";
+ case DIRECT:
+ return "DIRECT";
+ case DUPLICATING:
+ return "DUPLICATING";
+ case RECORD:
+ return "RECORD";
+ case OFFLOAD:
+ return "OFFLOAD";
+ default:
+ return "unknown";
+ }
+}
+
+String8 devicesToString(audio_devices_t devices)
+{
+ static const struct mapping {
+ audio_devices_t mDevices;
+ const char * mString;
+ } mappingsOut[] = {
+ AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE",
+ AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER",
+ AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET",
+ AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE",
+ AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX",
+ AUDIO_DEVICE_NONE, "NONE", // must be last
+ }, mappingsIn[] = {
+ AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC",
+ AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET",
+ AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL",
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX",
+ AUDIO_DEVICE_NONE, "NONE", // must be last
+ };
+ String8 result;
+ audio_devices_t allDevices = AUDIO_DEVICE_NONE;
+ const mapping *entry;
+ if (devices & AUDIO_DEVICE_BIT_IN) {
+ devices &= ~AUDIO_DEVICE_BIT_IN;
+ entry = mappingsIn;
+ } else {
+ entry = mappingsOut;
+ }
+ for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) {
+ allDevices = (audio_devices_t) (allDevices | entry->mDevices);
+ if (devices & entry->mDevices) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.append(entry->mString);
+ }
+ }
+ if (devices & ~allDevices) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.appendFormat("0x%X", devices & ~allDevices);
+ }
+ if (result.isEmpty()) {
+ result.append(entry->mString);
+ }
+ return result;
+}
+
+String8 inputFlagsToString(audio_input_flags_t flags)
+{
+ static const struct mapping {
+ audio_input_flags_t mFlag;
+ const char * mString;
+ } mappings[] = {
+ AUDIO_INPUT_FLAG_FAST, "FAST",
+ AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD",
+ AUDIO_INPUT_FLAG_NONE, "NONE", // must be last
+ };
+ String8 result;
+ audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE;
+ const mapping *entry;
+ for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) {
+ allFlags = (audio_input_flags_t) (allFlags | entry->mFlag);
+ if (flags & entry->mFlag) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.append(entry->mString);
+ }
+ }
+ if (flags & ~allFlags) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.appendFormat("0x%X", flags & ~allFlags);
+ }
+ if (result.isEmpty()) {
+ result.append(entry->mString);
+ }
+ return result;
+}
+
+String8 outputFlagsToString(audio_output_flags_t flags)
+{
+ static const struct mapping {
+ audio_output_flags_t mFlag;
+ const char * mString;
+ } mappings[] = {
+ AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT",
+ AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY",
+ AUDIO_OUTPUT_FLAG_FAST, "FAST",
+ AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER",
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD",
+ AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING",
+ AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC",
+ AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last
+ };
+ String8 result;
+ audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE;
+ const mapping *entry;
+ for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) {
+ allFlags = (audio_output_flags_t) (allFlags | entry->mFlag);
+ if (flags & entry->mFlag) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.append(entry->mString);
+ }
+ }
+ if (flags & ~allFlags) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.appendFormat("0x%X", flags & ~allFlags);
+ }
+ if (result.isEmpty()) {
+ result.append(entry->mString);
+ }
+ return result;
+}
+
+const char *sourceToString(audio_source_t source)
+{
+ switch (source) {
+ case AUDIO_SOURCE_DEFAULT: return "default";
+ case AUDIO_SOURCE_MIC: return "mic";
+ case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink";
+ case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink";
+ case AUDIO_SOURCE_VOICE_CALL: return "voice call";
+ case AUDIO_SOURCE_CAMCORDER: return "camcorder";
+ case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition";
+ case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication";
+ case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix";
+ case AUDIO_SOURCE_FM_TUNER: return "FM tuner";
+ case AUDIO_SOURCE_HOTWORD: return "hotword";
+ default: return "unknown";
+ }
+}
+
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
: Thread(false /*canCallJava*/),
@@ -577,20 +738,22 @@
bool locked = AudioFlinger::dumpTryLock(mLock);
if (!locked) {
- dprintf(fd, "thread %p maybe dead locked\n", this);
+ dprintf(fd, "thread %p may be deadlocked\n", this);
}
+ dprintf(fd, " Thread name: %s\n", mThreadName);
dprintf(fd, " I/O handle: %d\n", mId);
dprintf(fd, " TID: %d\n", getTid());
dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
- dprintf(fd, " Sample rate: %u\n", mSampleRate);
+ dprintf(fd, " Sample rate: %u Hz\n", mSampleRate);
dprintf(fd, " HAL frame count: %zu\n", mFrameCount);
+ dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize);
- dprintf(fd, " Channel Count: %u\n", mChannelCount);
- dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask,
+ dprintf(fd, " Channel count: %u\n", mChannelCount);
+ dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask,
channelMaskToString(mChannelMask, mType != RECORD).string());
- dprintf(fd, " Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
- dprintf(fd, " Frame size: %zu\n", mFrameSize);
+ dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
+ dprintf(fd, " Frame size: %zu bytes\n", mFrameSize);
dprintf(fd, " Pending config events:");
size_t numConfig = mConfigEvents.size();
if (numConfig) {
@@ -602,6 +765,9 @@
} else {
dprintf(fd, " none\n");
}
+ dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string());
+ dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string());
+ dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource));
if (locked) {
mLock.unlock();
@@ -635,19 +801,19 @@
String16 AudioFlinger::ThreadBase::getWakeLockTag()
{
switch (mType) {
- case MIXER:
- return String16("AudioMix");
- case DIRECT:
- return String16("AudioDirectOut");
- case DUPLICATING:
- return String16("AudioDup");
- case RECORD:
- return String16("AudioIn");
- case OFFLOAD:
- return String16("AudioOffload");
- default:
- ALOG_ASSERT(false);
- return String16("AudioUnknown");
+ case MIXER:
+ return String16("AudioMix");
+ case DIRECT:
+ return String16("AudioDirectOut");
+ case DUPLICATING:
+ return String16("AudioDup");
+ case RECORD:
+ return String16("AudioIn");
+ case OFFLOAD:
+ return String16("AudioOffload");
+ default:
+ ALOG_ASSERT(false);
+ return String16("AudioUnknown");
}
}
@@ -674,7 +840,7 @@
if (status == NO_ERROR) {
mWakeLockToken = binder;
}
- ALOGV("acquireWakeLock_l() %s status %d", mName, status);
+ ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status);
}
}
@@ -687,7 +853,7 @@
void AudioFlinger::ThreadBase::releaseWakeLock_l()
{
if (mWakeLockToken != 0) {
- ALOGV("releaseWakeLock_l() %s", mName);
+ ALOGV("releaseWakeLock_l() %s", mThreadName);
if (mPowerManager != 0) {
mPowerManager->releaseWakeLock(mWakeLockToken, 0,
true /* FIXME force oneway contrary to .aidl */);
@@ -708,7 +874,7 @@
sp<IBinder> binder =
defaultServiceManager()->checkService(String16("power"));
if (binder == 0) {
- ALOGW("Thread %s cannot connect to the power manager service", mName);
+ ALOGW("Thread %s cannot connect to the power manager service", mThreadName);
} else {
mPowerManager = interface_cast<IPowerManager>(binder);
binder->linkToDeath(mDeathRecipient);
@@ -728,7 +894,7 @@
status_t status;
status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(),
true /* FIXME force oneway contrary to .aidl */);
- ALOGV("acquireWakeLock_l() %s status %d", mName, status);
+ ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status);
}
}
@@ -912,7 +1078,7 @@
// mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo).
if (mType == DIRECT) {
ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s",
- desc->name, mName);
+ desc->name, mThreadName);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -936,7 +1102,8 @@
case DUPLICATING:
case RECORD:
default:
- ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName);
+ ALOGW("createEffect_l() Cannot add global effect %s on thread %s",
+ desc->name, mThreadName);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -1201,8 +1368,8 @@
// mLatchD, mLatchQ,
mLatchDValid(false), mLatchQValid(false)
{
- snprintf(mName, kNameLength, "AudioOut_%X", id);
- mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
+ snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id);
+ mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
// Assumes constructor is called by AudioFlinger with it's mLock held, but
// it would be safer to explicitly pass initial masterVolume/masterMute as
@@ -1315,7 +1482,10 @@
void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
- dprintf(fd, "\nOutput thread %p:\n", this);
+ dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type()));
+
+ dumpBase(fd, args);
+
dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
dprintf(fd, " Total writes: %d\n", mNumWrites);
@@ -1326,15 +1496,17 @@
dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer);
dprintf(fd, " Effect buffer: %p\n", mEffectBuffer);
dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
-
- dumpBase(fd, args);
+ AudioStreamOut *output = mOutput;
+ audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE;
+ String8 flagsAsString = outputFlagsToString(flags);
+ dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string());
}
// Thread virtuals
void AudioFlinger::PlaybackThread::onFirstRef()
{
- run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
+ run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
}
// ThreadBase virtuals
@@ -1378,9 +1550,10 @@
(
(sharedBuffer != 0)
) ||
- // use case 2: callback handler and frame count is default or at least as large as HAL
+ // use case 2: frame count is default or at least as large as HAL
(
- (tid != -1) &&
+ // we formerly checked for a callback handler (non-0 tid),
+ // but that is no longer required for TRANSFER_OBTAIN mode
((frameCount == 0) ||
(frameCount >= mFrameCount))
)
@@ -1420,20 +1593,25 @@
audio_is_linear_pcm(format),
channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
*flags &= ~IAudioFlinger::TRACK_FAST;
- // For compatibility with AudioTrack calculation, buffer depth is forced
- // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
- // This is probably too conservative, but legacy application code may depend on it.
- // If you change this calculation, also review the start threshold which is related.
+ }
+ }
+ // For normal PCM streaming tracks, update minimum frame count.
+ // For compatibility with AudioTrack calculation, buffer depth is forced
+ // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
+ // This is probably too conservative, but legacy application code may depend on it.
+ // If you change this calculation, also review the start threshold which is related.
+ if (!(*flags & IAudioFlinger::TRACK_FAST)
+ && audio_is_linear_pcm(format) && sharedBuffer == 0) {
uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
if (minBufCount < 2) {
minBufCount = 2;
}
- size_t minFrameCount = mNormalFrameCount * minBufCount;
- if (frameCount < minFrameCount) {
+ size_t minFrameCount =
+ minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate);
+ if (frameCount < minFrameCount) { // including frameCount == 0
frameCount = minFrameCount;
}
- }
}
*pFrameCount = frameCount;
@@ -1831,7 +2009,7 @@
LOG_FATAL("HAL format %#x not supported for mixed output",
mFormat);
}
- mFrameSize = audio_stream_out_frame_size(mOutput->stream);
+ mFrameSize = mOutput->getFrameSize();
mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
mFrameCount = mBufferSize / mFrameSize;
if (mFrameCount & 15) {
@@ -1861,6 +2039,22 @@
}
}
+ if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) {
+ // For best precision, we use float instead of the associated output
+ // device format (typically PCM 16 bit).
+
+ mFormat = AUDIO_FORMAT_PCM_FLOAT;
+ mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
+ mBufferSize = mFrameSize * mFrameCount;
+
+ // TODO: We currently use the associated output device channel mask and sample rate.
+ // (1) Perhaps use the ORed channel mask of all downstream MixerThreads
+ // (if a valid mask) to avoid premature downmix.
+ // (2) Perhaps use the maximum sample rate of all downstream MixerThreads
+ // instead of the output device sample rate to avoid loss of high frequency information.
+ // This may need to be updated as MixerThread/OutputTracks are added and not here.
+ }
+
// Calculate size of normal sink buffer relative to the HAL output buffer size
double multiplier = 1.0;
if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
@@ -1966,7 +2160,7 @@
} else {
status_t status;
uint32_t frames;
- status = mOutput->stream->get_render_position(mOutput->stream, &frames);
+ status = mOutput->getRenderPosition(&frames);
*dspFrames = (size_t)frames;
return status;
}
@@ -2008,13 +2202,13 @@
}
-AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
+AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
{
Mutex::Autolock _l(mLock);
return mOutput;
}
-AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
+AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
{
Mutex::Autolock _l(mLock);
AudioStreamOut *output = mOutput;
@@ -2137,6 +2331,7 @@
} else {
bytesWritten = framesWritten;
}
+ mLatchDValid = false;
status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp);
if (status == NO_ERROR) {
size_t totalFramesWritten = mNormalSink->framesWritten();
@@ -2159,8 +2354,7 @@
}
// FIXME We should have an implementation of timestamps for direct output threads.
// They are used e.g for multichannel PCM playback over HDMI.
- bytesWritten = mOutput->stream->write(mOutput->stream,
- (char *)mSinkBuffer + offset, mBytesRemaining);
+ bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
// do not wait for async callback in case of error of full write
@@ -2640,7 +2834,9 @@
}
} else {
+ ATRACE_BEGIN("sleep");
usleep(sleepTime);
+ ATRACE_END();
}
}
@@ -2711,8 +2907,7 @@
if ((mType == OFFLOAD || mType == DIRECT)
&& mOutput != NULL && mOutput->stream->get_presentation_position) {
uint64_t position64;
- int ret = mOutput->stream->get_presentation_position(
- mOutput->stream, &position64, ×tamp.mTime);
+ int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime);
if (ret == 0) {
timestamp.mPosition = (uint32_t)position64;
return NO_ERROR;
@@ -2800,6 +2995,12 @@
mNormalFrameCount);
mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
+ if (type == DUPLICATING) {
+ // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks
+ // (downstream MixerThreads) in DuplicatingThread::threadLoop_write().
+ // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink.
+ return;
+ }
// create an NBAIO sink for the HAL output stream, and negotiate
mOutputSink = new AudioStreamOutSink(output->stream);
size_t numCounterOffers = 0;
@@ -2841,6 +3042,7 @@
NBAIO_Format format = mOutputSink->format();
NBAIO_Format origformat = format;
// adjust format to match that of the Fast Mixer
+ ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat);
format.mFormat = fastMixerFormat;
format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;
@@ -3020,8 +3222,10 @@
#endif
}
state->mCommand = FastMixerState::MIX_WRITE;
+#ifdef FAST_THREAD_STATISTICS
mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
- FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
+ FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN);
+#endif
sq->end();
sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
if (kUseFastMixer == FastMixer_Dynamic) {
@@ -3083,7 +3287,7 @@
void AudioFlinger::PlaybackThread::threadLoop_standby()
{
ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
- mOutput->stream->common.standby(&mOutput->stream->common);
+ mOutput->standby();
if (mUseAsyncWrite != 0) {
// discard any pending drain or write ack by incrementing sequence
mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
@@ -3386,8 +3590,7 @@
if (sr == mSampleRate) {
desiredFrames = mNormalFrameCount;
} else {
- // +1 for rounding and +1 for additional sample needed for interpolation
- desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
+ desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate);
// add frames already consumed but not yet released by the resampler
// because mAudioTrackServerProxy->framesReady() will include these frames
desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
@@ -3405,6 +3608,23 @@
}
size_t framesReady = track->framesReady();
+ if (ATRACE_ENABLED()) {
+ // I wish we had formatted trace names
+ char traceName[16];
+ strcpy(traceName, "nRdy");
+ int name = track->name();
+ if (AudioMixer::TRACK0 <= name &&
+ name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) {
+ name -= AudioMixer::TRACK0;
+ traceName[4] = (name / 10) + '0';
+ traceName[5] = (name % 10) + '0';
+ } else {
+ traceName[4] = '?';
+ traceName[5] = '?';
+ }
+ traceName[6] = '\0';
+ ATRACE_INT(traceName, framesReady);
+ }
if ((framesReady >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
@@ -3836,7 +4056,7 @@
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
- mOutput->stream->common.standby(&mOutput->stream->common);
+ mOutput->standby();
mStandby = true;
mBytesWritten = 0;
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
@@ -4096,6 +4316,10 @@
}
if (track->isStopping_1()) {
track->mState = TrackBase::STOPPING_2;
+ if (last && mHwPaused) {
+ doHwResume = true;
+ mHwPaused = false;
+ }
}
if ((track->sharedBuffer() != 0) || track->isStopped() ||
track->isStopping_2() || track->isPaused()) {
@@ -4178,8 +4402,8 @@
while (frameCount) {
AudioBufferProvider::Buffer buffer;
buffer.frameCount = frameCount;
- mActiveTrack->getNextBuffer(&buffer);
- if (buffer.raw == NULL) {
+ status_t status = mActiveTrack->getNextBuffer(&buffer);
+ if (status != NO_ERROR || buffer.raw == NULL) {
memset(curBuf, 0, frameCount * mFrameSize);
break;
}
@@ -4235,14 +4459,17 @@
bool AudioFlinger::DirectOutputThread::shouldStandby_l()
{
bool trackPaused = false;
+ bool trackStopped = false;
// do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
// after a timeout and we will enter standby then.
if (mTracks.size() > 0) {
trackPaused = mTracks[mTracks.size() - 1]->isPaused();
+ trackStopped = mTracks[mTracks.size() - 1]->isStopped() ||
+ mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE;
}
- return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused));
+ return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused && !trackStopped));
}
// getTrackName_l() must be called with ThreadBase::mLock held
@@ -4291,7 +4518,7 @@
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
- mOutput->stream->common.standby(&mOutput->stream->common);
+ mOutput->standby();
mStandby = true;
mBytesWritten = 0;
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
@@ -4345,7 +4572,10 @@
// use shorter standby delay as on normal output to release
// hardware resources as soon as possible
- if (audio_is_linear_pcm(mFormat)) {
+ // no delay on outputs with HW A/V sync
+ if (usesHwAvSync()) {
+ standbyDelay = 0;
+ } else if (audio_is_linear_pcm(mFormat)) {
standbyDelay = microseconds(activeSleepTime*2);
} else {
standbyDelay = kOffloadStandbyDelayNs;
@@ -4354,9 +4584,7 @@
void AudioFlinger::DirectOutputThread::flushHw_l()
{
- if (mOutput->stream->flush != NULL) {
- mOutput->stream->flush(mOutput->stream);
- }
+ mOutput->flush();
mHwPaused = false;
}
@@ -4646,7 +4874,7 @@
size_t audioHALFrames =
(mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
size_t framesWritten =
- mBytesWritten / audio_stream_out_frame_size(mOutput->stream);
+ mBytesWritten / mOutput->getFrameSize();
track->presentationComplete(framesWritten, audioHALFrames);
track->reset();
tracksToRemove->add(track);
@@ -4797,16 +5025,8 @@
ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
{
- // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT
- // for delivery downstream as needed. This in-place conversion is safe as
- // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format
- // (AUDIO_FORMAT_PCM_8_BIT is not allowed here).
- if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
- memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT,
- mSinkBuffer, mFormat, writeFrames * mChannelCount);
- }
for (size_t i = 0; i < outputTracks.size(); i++) {
- outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames);
+ outputTracks[i]->write(mSinkBuffer, writeFrames);
}
mStandby = false;
return (ssize_t)mSinkBufferSize;
@@ -4833,25 +5053,26 @@
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
Mutex::Autolock _l(mLock);
- // FIXME explain this formula
- size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
- // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat
- // due to current usage case and restrictions on the AudioBufferProvider.
- // Actual buffer conversion is done in threadLoop_write().
- //
- // TODO: This may change in the future, depending on multichannel
- // (and non int16_t*) support on AF::PlaybackThread::OutputTrack
- OutputTrack *outputTrack = new OutputTrack(thread,
+ // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass.
+ // Adjust for thread->sampleRate() to determine minimum buffer frame count.
+ // Then triple buffer because Threads do not run synchronously and may not be clock locked.
+ const size_t frameCount =
+ 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate());
+ // TODO: Consider asynchronous sample rate conversion to handle clock disparity
+ // from different OutputTracks and their associated MixerThreads (e.g. one may
+ // nearly empty and the other may be dropping data).
+
+ sp<OutputTrack> outputTrack = new OutputTrack(thread,
this,
mSampleRate,
- AUDIO_FORMAT_PCM_16_BIT,
+ mFormat,
mChannelMask,
frameCount,
IPCThreadState::self()->getCallingUid());
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
mOutputTracks.add(outputTrack);
- ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
+ ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread);
updateWaitTime_l();
}
}
@@ -4952,8 +5173,8 @@
// mFastCaptureNBLogWriter
, mFastTrackAvail(false)
{
- snprintf(mName, kNameLength, "AudioIn_%X", id);
- mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
+ snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id);
+ mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
readInputParameters_l();
@@ -4993,7 +5214,7 @@
}
if (initFastCapture) {
- // create a Pipe for FastMixer to write to, and for us and fast tracks to read from
+ // create a Pipe for FastCapture to write to, and for us and fast tracks to read from
NBAIO_Format format = mInputSource->format();
size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each
size_t pipeSize = pipeFramesP2 * Format_frameSize(format);
@@ -5094,7 +5315,7 @@
void AudioFlinger::RecordThread::onFirstRef()
{
- run(mName, PRIORITY_URGENT_AUDIO);
+ run(mThreadName, PRIORITY_URGENT_AUDIO);
}
bool AudioFlinger::RecordThread::threadLoop()
@@ -5135,7 +5356,9 @@
// sleep with mutex unlocked
if (sleepUs > 0) {
+ ATRACE_BEGIN("sleep");
usleep(sleepUs);
+ ATRACE_END();
sleepUs = 0;
}
@@ -5279,7 +5502,8 @@
state->mCommand = FastCaptureState::READ_WRITE;
#if 0 // FIXME
mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
- FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
+ FastThreadDumpState::kSamplingNforLowRamDevice :
+ FastThreadDumpState::kSamplingN);
#endif
didModify = true;
}
@@ -5427,8 +5651,8 @@
upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src,
part1);
} else {
- downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src,
- part1);
+ downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
+ (const int16_t *)src, part1);
}
dst += part1 * activeTrack->mFrameSize;
front += part1;
@@ -5649,8 +5873,9 @@
// client expresses a preference for FAST, but we get the final say
if (*flags & IAudioFlinger::TRACK_FAST) {
if (
- // use case: callback handler
- (tid != -1) &&
+ // we formerly checked for a callback handler (non-0 tid),
+ // but that is no longer required for TRANSFER_OBTAIN mode
+ //
// frame count is not specified, or is exactly the pipe depth
((frameCount == 0) || (frameCount == mPipeFramesP2)) &&
// PCM data
@@ -5939,15 +6164,17 @@
{
dprintf(fd, "\nInput thread %p:\n", this);
- if (mActiveTracks.size() > 0) {
- dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize);
- } else {
+ dumpBase(fd, args);
+
+ if (mActiveTracks.size() == 0) {
dprintf(fd, " No active record clients\n");
}
dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no");
dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no");
- dumpBase(fd, args);
+ // Make a non-atomic copy of fast capture dump state so it won't change underneath us
+ const FastCaptureDumpState copy(mFastCaptureDumpState);
+ copy.dump(fd);
}
void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
@@ -6412,4 +6639,4 @@
config->ext.mix.usecase.source = mAudioSource;
}
-}; // namespace android
+} // namespace android
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 1088843..d600ea9 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -32,6 +32,8 @@
OFFLOAD // Thread class is OffloadThread
};
+ static const char *threadTypeToString(type_t type);
+
ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
virtual ~ThreadBase();
@@ -406,6 +408,7 @@
audio_channel_mask_t mChannelMask;
uint32_t mChannelCount;
size_t mFrameSize;
+ // not HAL frame size, this is for output sink (to pipe to fast mixer)
audio_format_t mFormat; // Source format for Recording and
// Sink format for Playback.
// Sink format may be different than
@@ -424,13 +427,13 @@
bool mStandby; // Whether thread is currently in standby.
audio_devices_t mOutDevice; // output device
audio_devices_t mInDevice; // input device
- audio_source_t mAudioSource; // (see audio.h, audio_source_t)
+ audio_source_t mAudioSource;
const audio_io_handle_t mId;
Vector< sp<EffectChain> > mEffectChains;
- static const int kNameLength = 16; // prctl(PR_SET_NAME) limit
- char mName[kNameLength];
+ static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit
+ char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated
sp<IPowerManager> mPowerManager;
sp<IBinder> mWakeLockToken;
const sp<PMDeathRecipient> mDeathRecipient;
@@ -1167,7 +1170,8 @@
const sp<MemoryDealer> mReadOnlyHeap;
// one-time initialization, no locks required
- sp<FastCapture> mFastCapture; // non-0 if there is also a fast capture
+ sp<FastCapture> mFastCapture; // non-0 if there is also
+ // a fast capture
// FIXME audio watchdog thread
// contents are not guaranteed to be consistent, no locks required
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index e970036..dc9f249 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -20,6 +20,7 @@
//#define LOG_NDEBUG 0
#include "Configuration.h"
+#include <linux/futex.h>
#include <math.h>
#include <sys/syscall.h>
#include <utils/Log.h>
@@ -404,9 +405,7 @@
mAudioTrackServerProxy(NULL),
mResumeToStopping(false),
mFlushHwPending(false),
- mPreviousValid(false),
- mPreviousFramesWritten(0)
- // mPreviousTimestamp
+ mPreviousTimestampValid(false)
{
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
@@ -443,8 +442,6 @@
// this means we are potentially denying other more important fast tracks from
// being created. It would be better to allocate the index dynamically.
mFastIndex = i;
- // Read the initial underruns because this field is never cleared by the fast mixer
- mObservedUnderruns = thread->getFastTrackUnderruns(i);
thread->mFastTrackAvailMask &= ~(1 << i);
}
}
@@ -693,6 +690,12 @@
}
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ if (isFastTrack()) {
+ // refresh fast track underruns on start because that field is never cleared
+ // by the fast mixer; furthermore, the same track can be recycled, i.e. start
+ // after stop.
+ mObservedUnderruns = playbackThread->getFastTrackUnderruns(mFastIndex);
+ }
status = playbackThread->addTrack_l(this);
if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
@@ -742,6 +745,7 @@
// move to STOPPING_2 when drain completes and then STOPPED
mState = STOPPING_1;
}
+ playbackThread->broadcast_l();
ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
playbackThread);
}
@@ -859,6 +863,7 @@
if (mState == FLUSHED) {
mState = IDLE;
}
+ mPreviousTimestampValid = false;
}
}
@@ -880,19 +885,22 @@
{
// Client should implement this using SSQ; the unpresented frame count in latch is irrelevant
if (isFastTrack()) {
- // FIXME no lock held to set mPreviousValid = false
+ // FIXME no lock held to set mPreviousTimestampValid = false
return INVALID_OPERATION;
}
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
- // FIXME no lock held to set mPreviousValid = false
+ // FIXME no lock held to set mPreviousTimestampValid = false
return INVALID_OPERATION;
}
+
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+
+ status_t result = INVALID_OPERATION;
if (!isOffloaded() && !isDirect()) {
if (!playbackThread->mLatchQValid) {
- mPreviousValid = false;
+ mPreviousTimestampValid = false;
return INVALID_OPERATION;
}
uint32_t unpresentedFrames =
@@ -908,36 +916,54 @@
uint32_t framesWritten = i >= 0 ?
playbackThread->mLatchQ.mFramesReleased[i] :
mAudioTrackServerProxy->framesReleased();
- bool checkPreviousTimestamp = mPreviousValid && framesWritten >= mPreviousFramesWritten;
if (framesWritten < unpresentedFrames) {
- mPreviousValid = false;
- return INVALID_OPERATION;
+ mPreviousTimestampValid = false;
+ // return invalid result
+ } else {
+ timestamp.mPosition = framesWritten - unpresentedFrames;
+ timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime;
+ result = NO_ERROR;
}
- mPreviousFramesWritten = framesWritten;
- uint32_t position = framesWritten - unpresentedFrames;
- struct timespec time = playbackThread->mLatchQ.mTimestamp.mTime;
- if (checkPreviousTimestamp) {
- if (time.tv_sec < mPreviousTimestamp.mTime.tv_sec ||
- (time.tv_sec == mPreviousTimestamp.mTime.tv_sec &&
- time.tv_nsec < mPreviousTimestamp.mTime.tv_nsec)) {
- ALOGW("Time is going backwards");
- }
- // position can bobble slightly as an artifact; this hides the bobble
- static const uint32_t MINIMUM_POSITION_DELTA = 8u;
- if ((position <= mPreviousTimestamp.mPosition) ||
- (position - mPreviousTimestamp.mPosition) < MINIMUM_POSITION_DELTA) {
- position = mPreviousTimestamp.mPosition;
- time = mPreviousTimestamp.mTime;
- }
- }
- timestamp.mPosition = position;
- timestamp.mTime = time;
- mPreviousTimestamp = timestamp;
- mPreviousValid = true;
- return NO_ERROR;
+ } else { // offloaded or direct
+ result = playbackThread->getTimestamp_l(timestamp);
}
- return playbackThread->getTimestamp_l(timestamp);
+ // Prevent retrograde motion in timestamp.
+ if (result == NO_ERROR) {
+ if (mPreviousTimestampValid) {
+ if (timestamp.mTime.tv_sec < mPreviousTimestamp.mTime.tv_sec ||
+ (timestamp.mTime.tv_sec == mPreviousTimestamp.mTime.tv_sec &&
+ timestamp.mTime.tv_nsec < mPreviousTimestamp.mTime.tv_nsec)) {
+ ALOGW("WARNING - retrograde timestamp time");
+ // FIXME Consider blocking this from propagating upwards.
+ }
+
+ // Looking at signed delta will work even when the timestamps
+ // are wrapping around.
+ int32_t deltaPosition = static_cast<int32_t>(timestamp.mPosition
+ - mPreviousTimestamp.mPosition);
+ // position can bobble slightly as an artifact; this hides the bobble
+ static const int32_t MINIMUM_POSITION_DELTA = 8;
+ if (deltaPosition < 0) {
+#define TIME_TO_NANOS(time) ((uint64_t)time.tv_sec * 1000000000 + time.tv_nsec)
+ ALOGW("WARNING - retrograde timestamp position corrected,"
+ " %d = %u - %u, (at %llu, %llu nanos)",
+ deltaPosition,
+ timestamp.mPosition,
+ mPreviousTimestamp.mPosition,
+ TIME_TO_NANOS(timestamp.mTime),
+ TIME_TO_NANOS(mPreviousTimestamp.mTime));
+#undef TIME_TO_NANOS
+ }
+ if (deltaPosition < MINIMUM_POSITION_DELTA) {
+ // Current timestamp is bad. Use last valid timestamp.
+ timestamp = mPreviousTimestamp;
+ }
+ }
+ mPreviousTimestamp = timestamp;
+ mPreviousTimestampValid = true;
+ }
+ return result;
}
status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
@@ -1709,36 +1735,18 @@
mActive = false;
}
-bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
+bool AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
- uint32_t channelCount = mChannelCount;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
- inBuffer.i16 = data;
+ inBuffer.raw = data;
uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
if (!mActive && frames != 0) {
- start();
- sp<ThreadBase> thread = mThread.promote();
- if (thread != 0) {
- MixerThread *mixerThread = (MixerThread *)thread.get();
- if (mFrameCount > frames) {
- if (mBufferQueue.size() < kMaxOverFlowBuffers) {
- uint32_t startFrames = (mFrameCount - frames);
- pInBuffer = new Buffer;
- pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
- pInBuffer->frameCount = startFrames;
- pInBuffer->i16 = pInBuffer->mBuffer;
- memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
- mBufferQueue.add(pInBuffer);
- } else {
- ALOGW("OutputTrack::write() %p no more buffers in queue", this);
- }
- }
- }
+ (void) start();
}
while (waitTimeLeftMs) {
@@ -1773,20 +1781,20 @@
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
pInBuffer->frameCount;
- memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
+ memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * mFrameSize);
Proxy::Buffer buf;
buf.mFrameCount = outFrames;
buf.mRaw = NULL;
mClientProxy->releaseBuffer(&buf);
pInBuffer->frameCount -= outFrames;
- pInBuffer->i16 += outFrames * channelCount;
+ pInBuffer->raw = (int8_t *)pInBuffer->raw + outFrames * mFrameSize;
mOutBuffer.frameCount -= outFrames;
- mOutBuffer.i16 += outFrames * channelCount;
+ mOutBuffer.raw = (int8_t *)mOutBuffer.raw + outFrames * mFrameSize;
if (pInBuffer->frameCount == 0) {
if (mBufferQueue.size()) {
mBufferQueue.removeAt(0);
- delete [] pInBuffer->mBuffer;
+ free(pInBuffer->mBuffer);
delete pInBuffer;
ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
mThread.unsafe_get(), mBufferQueue.size());
@@ -1802,11 +1810,10 @@
if (thread != 0 && !thread->standby()) {
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
pInBuffer = new Buffer;
- pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
+ pInBuffer->mBuffer = malloc(inBuffer.frameCount * mFrameSize);
pInBuffer->frameCount = inBuffer.frameCount;
- pInBuffer->i16 = pInBuffer->mBuffer;
- memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
- sizeof(int16_t));
+ pInBuffer->raw = pInBuffer->mBuffer;
+ memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize);
mBufferQueue.add(pInBuffer);
ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
mThread.unsafe_get(), mBufferQueue.size());
@@ -1817,23 +1824,10 @@
}
}
- // Calling write() with a 0 length buffer, means that no more data will be written:
- // If no more buffers are pending, fill output track buffer to make sure it is started
- // by output mixer.
- if (frames == 0 && mBufferQueue.size() == 0) {
- // FIXME borken, replace by getting framesReady() from proxy
- size_t user = 0; // was mCblk->user
- if (user < mFrameCount) {
- frames = mFrameCount - user;
- pInBuffer = new Buffer;
- pInBuffer->mBuffer = new int16_t[frames * channelCount];
- pInBuffer->frameCount = frames;
- pInBuffer->i16 = pInBuffer->mBuffer;
- memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
- mBufferQueue.add(pInBuffer);
- } else if (mActive) {
- stop();
- }
+ // Calling write() with a 0 length buffer means that no more data will be written:
+ // We rely on stop() to set the appropriate flags to allow the remaining frames to play out.
+ if (frames == 0 && mBufferQueue.size() == 0 && mActive) {
+ stop();
}
return outputBufferFull;
@@ -1859,7 +1853,7 @@
for (size_t i = 0; i < size; i++) {
Buffer *pBuffer = mBufferQueue.itemAt(i);
- delete [] pBuffer->mBuffer;
+ free(pBuffer->mBuffer);
delete pBuffer;
}
mBufferQueue.clear();
@@ -2212,4 +2206,4 @@
mProxy->releaseBuffer(buffer);
}
-}; // namespace android
+} // namespace android
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 84a655a..7893778 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -427,6 +427,14 @@
printf("quality: %d channels: %d msec: %" PRId64 " Mfrms/s: %.2lf\n",
quality, channels, time/1000000, output_frames * looplimit / (time / 1e9) / 1e6);
resampler->reset();
+
+ // TODO fix legacy bug: reset does not clear buffers.
+ // delete and recreate resampler here.
+ delete resampler;
+ resampler = AudioResampler::create(format, channels,
+ output_freq, quality);
+ resampler->setSampleRate(input_freq);
+ resampler->setVolume(AudioResampler::UNITY_GAIN_FLOAT, AudioResampler::UNITY_GAIN_FLOAT);
}
memset(output_vaddr, 0, output_size);
diff --git a/services/audioflinger/tests/build_and_run_all_unit_tests.sh b/services/audioflinger/tests/build_and_run_all_unit_tests.sh
index 2c453b0..7f4d456 100755
--- a/services/audioflinger/tests/build_and_run_all_unit_tests.sh
+++ b/services/audioflinger/tests/build_and_run_all_unit_tests.sh
@@ -15,7 +15,7 @@
echo "waiting for device"
adb root && adb wait-for-device remount
adb push $OUT/system/lib/libaudioresampler.so /system/lib
-adb push $OUT/system/bin/resampler_tests /system/bin
+adb push $OUT/data/nativetest/resampler_tests /system/bin
sh $ANDROID_BUILD_TOP/frameworks/av/services/audioflinger/tests/run_all_unit_tests.sh
diff --git a/services/audioflinger/tests/mixer_to_wav_tests.sh b/services/audioflinger/tests/mixer_to_wav_tests.sh
index 9b39e77..d0482a1 100755
--- a/services/audioflinger/tests/mixer_to_wav_tests.sh
+++ b/services/audioflinger/tests/mixer_to_wav_tests.sh
@@ -60,11 +60,21 @@
fi
# Test:
+# process__genericResampling with mixed integer and float track input
+# track__Resample / track__genericResample
+ adb shell test-mixer $1 -s 48000 \
+ -o /sdcard/tm48000grif.wav \
+ sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \
+ sine:f,6,6000,19000 chirp:i,4,30000
+ adb pull /sdcard/tm48000grif.wav $2
+
+# Test:
# process__genericResampling
# track__Resample / track__genericResample
adb shell test-mixer $1 -s 48000 \
-o /sdcard/tm48000gr.wav \
- sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000
+ sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \
+ sine:6,6000,19000
adb pull /sdcard/tm48000gr.wav $2
# Test:
diff --git a/services/audioflinger/tests/test-mixer.cpp b/services/audioflinger/tests/test-mixer.cpp
index 9a4fad6..8da6245 100644
--- a/services/audioflinger/tests/test-mixer.cpp
+++ b/services/audioflinger/tests/test-mixer.cpp
@@ -39,7 +39,7 @@
fprintf(stderr, "Usage: %s [-f] [-m] [-c channels]"
" [-s sample-rate] [-o <output-file>] [-a <aux-buffer-file>] [-P csv]"
" (<input-file> | <command>)+\n", name);
- fprintf(stderr, " -f enable floating point input track\n");
+ fprintf(stderr, " -f enable floating point input track by default\n");
fprintf(stderr, " -m enable floating point mixer output\n");
fprintf(stderr, " -c number of mixer output channels\n");
fprintf(stderr, " -s mixer sample-rate\n");
@@ -47,8 +47,8 @@
fprintf(stderr, " -a <aux-buffer-file>\n");
fprintf(stderr, " -P # frames provided per call to resample() in CSV format\n");
fprintf(stderr, " <input-file> is a WAV file\n");
- fprintf(stderr, " <command> can be 'sine:<channels>,<frequency>,<samplerate>'\n");
- fprintf(stderr, " 'chirp:<channels>,<samplerate>'\n");
+ fprintf(stderr, " <command> can be 'sine:[(i|f),]<channels>,<frequency>,<samplerate>'\n");
+ fprintf(stderr, " 'chirp:[(i|f),]<channels>,<samplerate>'\n");
}
static int writeFile(const char *filename, const void *buffer,
@@ -78,6 +78,18 @@
return EXIT_SUCCESS;
}
+const char *parseFormat(const char *s, bool *useFloat) {
+ if (!strncmp(s, "f,", 2)) {
+ *useFloat = true;
+ return s + 2;
+ }
+ if (!strncmp(s, "i,", 2)) {
+ *useFloat = false;
+ return s + 2;
+ }
+ return s;
+}
+
int main(int argc, char* argv[]) {
const char* const progname = argv[0];
bool useInputFloat = false;
@@ -88,8 +100,9 @@
std::vector<int> Pvalues;
const char* outputFilename = NULL;
const char* auxFilename = NULL;
- std::vector<int32_t> Names;
- std::vector<SignalProvider> Providers;
+ std::vector<int32_t> names;
+ std::vector<SignalProvider> providers;
+ std::vector<audio_format_t> formats;
for (int ch; (ch = getopt(argc, argv, "fmc:s:o:a:P:")) != -1;) {
switch (ch) {
@@ -138,54 +151,65 @@
size_t outputFrames = 0;
// create providers for each track
- Providers.resize(argc);
+ names.resize(argc);
+ providers.resize(argc);
+ formats.resize(argc);
for (int i = 0; i < argc; ++i) {
static const char chirp[] = "chirp:";
static const char sine[] = "sine:";
static const double kSeconds = 1;
+ bool useFloat = useInputFloat;
if (!strncmp(argv[i], chirp, strlen(chirp))) {
std::vector<int> v;
+ const char *s = parseFormat(argv[i] + strlen(chirp), &useFloat);
- parseCSV(argv[i] + strlen(chirp), v);
+ parseCSV(s, v);
if (v.size() == 2) {
printf("creating chirp(%d %d)\n", v[0], v[1]);
- if (useInputFloat) {
- Providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
+ if (useFloat) {
+ providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
- Providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
+ providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
- Providers[i].setIncr(Pvalues);
+ providers[i].setIncr(Pvalues);
} else {
fprintf(stderr, "malformed input '%s'\n", argv[i]);
}
} else if (!strncmp(argv[i], sine, strlen(sine))) {
std::vector<int> v;
+ const char *s = parseFormat(argv[i] + strlen(sine), &useFloat);
- parseCSV(argv[i] + strlen(sine), v);
+ parseCSV(s, v);
if (v.size() == 3) {
printf("creating sine(%d %d %d)\n", v[0], v[1], v[2]);
- if (useInputFloat) {
- Providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
+ if (useFloat) {
+ providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
- Providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
+ providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
- Providers[i].setIncr(Pvalues);
+ providers[i].setIncr(Pvalues);
} else {
fprintf(stderr, "malformed input '%s'\n", argv[i]);
}
} else {
printf("creating filename(%s)\n", argv[i]);
if (useInputFloat) {
- Providers[i].setFile<float>(argv[i]);
+ providers[i].setFile<float>(argv[i]);
+ formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
- Providers[i].setFile<short>(argv[i]);
+ providers[i].setFile<short>(argv[i]);
+ formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
- Providers[i].setIncr(Pvalues);
+ providers[i].setIncr(Pvalues);
}
// calculate the number of output frames
- size_t nframes = (int64_t) Providers[i].getNumFrames() * outputSampleRate
- / Providers[i].getSampleRate();
+ size_t nframes = (int64_t) providers[i].getNumFrames() * outputSampleRate
+ / providers[i].getSampleRate();
if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames
outputFrames = nframes;
}
@@ -213,22 +237,20 @@
// create the mixer.
const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960
AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate);
- audio_format_t inputFormat = useInputFloat
- ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
audio_format_t mixerFormat = useMixerFloat
? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
- float f = AudioMixer::UNITY_GAIN_FLOAT / Providers.size(); // normalize volume by # tracks
+ float f = AudioMixer::UNITY_GAIN_FLOAT / providers.size(); // normalize volume by # tracks
static float f0; // zero
// set up the tracks.
- 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());
+ 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,
- inputFormat, AUDIO_SESSION_OUTPUT_MIX);
+ formats[i], AUDIO_SESSION_OUTPUT_MIX);
ALOG_ASSERT(name >= 0);
- Names.push_back(name);
- mixer->setBufferProvider(name, &Providers[i]);
+ names[i] = name;
+ mixer->setBufferProvider(name, &providers[i]);
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
(void *)outputAddr);
mixer->setParameter(
@@ -240,7 +262,7 @@
name,
AudioMixer::TRACK,
AudioMixer::FORMAT,
- (void *)(uintptr_t)inputFormat);
+ (void *)(uintptr_t)formats[i]);
mixer->setParameter(
name,
AudioMixer::TRACK,
@@ -255,7 +277,7 @@
name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
- (void *)(uintptr_t)Providers[i].getSampleRate());
+ (void *)(uintptr_t)providers[i].getSampleRate());
if (useRamp) {
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0);
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0);
@@ -277,11 +299,11 @@
// pump the mixer to process data.
size_t i;
for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) {
- for (size_t j = 0; j < Names.size(); ++j) {
- mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
+ for (size_t j = 0; j < names.size(); ++j) {
+ mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
(char *) outputAddr + i * outputFrameSize);
if (auxFilename) {
- mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
+ mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
(char *) auxAddr + i * auxFrameSize);
}
}
diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk
index 188fc89..d4ce86a 100644
--- a/services/audiopolicy/Android.mk
+++ b/services/audiopolicy/Android.mk
@@ -3,25 +3,27 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- AudioPolicyService.cpp \
- AudioPolicyEffects.cpp
+ service/AudioPolicyService.cpp \
+ service/AudioPolicyEffects.cpp
ifeq ($(USE_LEGACY_AUDIO_POLICY), 1)
LOCAL_SRC_FILES += \
- AudioPolicyInterfaceImplLegacy.cpp \
- AudioPolicyClientImplLegacy.cpp
+ service/AudioPolicyInterfaceImplLegacy.cpp \
+ service/AudioPolicyClientImplLegacy.cpp
LOCAL_CFLAGS += -DUSE_LEGACY_AUDIO_POLICY
else
LOCAL_SRC_FILES += \
- AudioPolicyInterfaceImpl.cpp \
- AudioPolicyClientImpl.cpp
+ service/AudioPolicyInterfaceImpl.cpp \
+ service/AudioPolicyClientImpl.cpp
endif
LOCAL_C_INCLUDES := \
$(TOPDIR)frameworks/av/services/audioflinger \
$(call include-path-for, audio-effects) \
- $(call include-path-for, audio-utils)
+ $(call include-path-for, audio-utils) \
+ $(TOPDIR)frameworks/av/services/audiopolicy/common/include \
+ $(TOPDIR)frameworks/av/services/audiopolicy/engine/interface \
LOCAL_SHARED_LIBRARIES := \
libcutils \
@@ -39,7 +41,8 @@
endif
LOCAL_STATIC_LIBRARIES := \
- libmedia_helper
+ libmedia_helper \
+ libaudiopolicycomponents
LOCAL_MODULE:= libaudiopolicyservice
@@ -53,7 +56,7 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- AudioPolicyManager.cpp
+ managerdefault/AudioPolicyManager.cpp \
LOCAL_SHARED_LIBRARIES := \
libcutils \
@@ -61,8 +64,15 @@
liblog \
libsoundtrigger
+LOCAL_SHARED_LIBRARIES += libaudiopolicyenginedefault
+
+LOCAL_C_INCLUDES += \
+ $(TOPDIR)frameworks/av/services/audiopolicy/common/include \
+ $(TOPDIR)frameworks/av/services/audiopolicy/engine/interface \
+
LOCAL_STATIC_LIBRARIES := \
- libmedia_helper
+ libmedia_helper \
+ libaudiopolicycomponents
LOCAL_MODULE:= libaudiopolicymanagerdefault
@@ -73,14 +83,26 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- AudioPolicyFactory.cpp
+ manager/AudioPolicyFactory.cpp
LOCAL_SHARED_LIBRARIES := \
libaudiopolicymanagerdefault
+LOCAL_STATIC_LIBRARIES := \
+ libaudiopolicycomponents
+
+LOCAL_C_INCLUDES += \
+ $(TOPDIR)frameworks/av/services/audiopolicy/common/include \
+ $(TOPDIR)frameworks/av/services/audiopolicy/engine/interface \
+
LOCAL_MODULE:= libaudiopolicymanager
include $(BUILD_SHARED_LIBRARY)
endif
endif
+
+#######################################################################
+# Recursive call sub-folder Android.mk
+#
+include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 4508fa7..116d0d6 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -75,7 +75,8 @@
// indicate a change in device connection status
virtual status_t setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address) = 0;
+ const char *device_address,
+ const char *device_name) = 0;
// retrieve a device connection status
virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address) = 0;
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
deleted file mode 100644
index 6ebd0ed..0000000
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ /dev/null
@@ -1,8098 +0,0 @@
-/*
- * 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 "AudioPolicyManager"
-//#define LOG_NDEBUG 0
-
-//#define VERY_VERBOSE_LOGGING
-#ifdef VERY_VERBOSE_LOGGING
-#define ALOGVV ALOGV
-#else
-#define ALOGVV(a...) do { } while(0)
-#endif
-
-// A device mask for all audio input devices that are considered "virtual" when evaluating
-// active inputs in getActiveInput()
-#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_FM_TUNER)
-// A device mask for all audio output devices that are considered "remote" when evaluating
-// active output devices in isStreamActiveRemotely()
-#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
-// A device mask for all audio input and output devices where matching inputs/outputs on device
-// type alone is not enough: the address must match too
-#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \
- AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
-
-#include <inttypes.h>
-#include <math.h>
-
-#include <cutils/properties.h>
-#include <utils/Log.h>
-#include <hardware/audio.h>
-#include <hardware/audio_effect.h>
-#include <media/AudioParameter.h>
-#include <media/AudioPolicyHelper.h>
-#include <soundtrigger/SoundTrigger.h>
-#include "AudioPolicyManager.h"
-#include "audio_policy_conf.h"
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-// Definitions for audio_policy.conf file parsing
-// ----------------------------------------------------------------------------
-
-struct StringToEnum {
- const char *name;
- uint32_t value;
-};
-
-#define STRING_TO_ENUM(string) { #string, string }
-#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
-
-const StringToEnum sDeviceNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
-};
-
-const StringToEnum sOutputFlagNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
-};
-
-const StringToEnum sInputFlagNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
- STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
-};
-
-const StringToEnum sFormatNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
- STRING_TO_ENUM(AUDIO_FORMAT_MP3),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
- STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
- STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
- STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
- STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
- STRING_TO_ENUM(AUDIO_FORMAT_AC3),
- STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
-};
-
-const StringToEnum sOutChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
-};
-
-const StringToEnum sInChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
-};
-
-const StringToEnum sGainModeNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
- STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
- STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
-};
-
-
-uint32_t AudioPolicyManager::stringToEnum(const struct StringToEnum *table,
- size_t size,
- const char *name)
-{
- for (size_t i = 0; i < size; i++) {
- if (strcmp(table[i].name, name) == 0) {
- ALOGV("stringToEnum() found %s", table[i].name);
- return table[i].value;
- }
- }
- return 0;
-}
-
-const char *AudioPolicyManager::enumToString(const struct StringToEnum *table,
- size_t size,
- uint32_t value)
-{
- for (size_t i = 0; i < size; i++) {
- if (table[i].value == value) {
- return table[i].name;
- }
- }
- return "";
-}
-
-bool AudioPolicyManager::stringToBool(const char *value)
-{
- return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
-}
-
-
-// ----------------------------------------------------------------------------
-// AudioPolicyInterface implementation
-// ----------------------------------------------------------------------------
-
-status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address)
-{
- return setDeviceConnectionStateInt(device, state, device_address);
-}
-
-status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address)
-{
- ALOGV("setDeviceConnectionState() device: %x, state %d, address %s",
- device, state, device_address != NULL ? device_address : "");
-
- // connect/disconnect only 1 device at a time
- if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
-
- sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address);
-
- // handle output devices
- if (audio_is_output_device(device)) {
- SortedVector <audio_io_handle_t> outputs;
-
- ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
-
- // save a copy of the opened output descriptors before any output is opened or closed
- // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
- mPreviousOutputs = mOutputs;
- switch (state)
- {
- // handle output device connection
- case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
- if (index >= 0) {
- ALOGW("setDeviceConnectionState() device already connected: %x", device);
- return INVALID_OPERATION;
- }
- ALOGV("setDeviceConnectionState() connecting device %x", device);
-
- // register new device as available
- index = mAvailableOutputDevices.add(devDesc);
- if (index >= 0) {
- sp<HwModule> module = getModuleForDevice(device);
- if (module == 0) {
- ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
- device);
- mAvailableOutputDevices.remove(devDesc);
- return INVALID_OPERATION;
- }
- mAvailableOutputDevices[index]->mId = nextUniqueId();
- mAvailableOutputDevices[index]->mModule = module;
- } else {
- return NO_MEMORY;
- }
-
- if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
- mAvailableOutputDevices.remove(devDesc);
- return INVALID_OPERATION;
- }
- // outputs should never be empty here
- ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
- "checkOutputsForDevice() returned no outputs but status OK");
- ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
- outputs.size());
-
-
- // Set connect to HALs
- AudioParameter param = AudioParameter(devDesc->mAddress);
- param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
- mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
-
- } break;
- // handle output device disconnection
- case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
- if (index < 0) {
- ALOGW("setDeviceConnectionState() device not connected: %x", device);
- return INVALID_OPERATION;
- }
-
- ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
-
- // Set Disconnect to HALs
- AudioParameter param = AudioParameter(devDesc->mAddress);
- param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
- mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
-
- // remove device from available output devices
- mAvailableOutputDevices.remove(devDesc);
-
- checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
- } break;
-
- default:
- ALOGE("setDeviceConnectionState() invalid state: %x", state);
- return BAD_VALUE;
- }
-
- // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
- // output is suspended before any tracks are moved to it
- checkA2dpSuspend();
- checkOutputForAllStrategies();
- // outputs must be closed after checkOutputForAllStrategies() is executed
- if (!outputs.isEmpty()) {
- for (size_t i = 0; i < outputs.size(); i++) {
- sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
- // close unused outputs after device disconnection or direct outputs that have been
- // opened by checkOutputsForDevice() to query dynamic parameters
- if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
- (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
- (desc->mDirectOpenCount == 0))) {
- closeOutput(outputs[i]);
- }
- }
- // check again after closing A2DP output to reset mA2dpSuspended if needed
- checkA2dpSuspend();
- }
-
- updateDevicesAndOutputs();
- if (mPhoneState == AUDIO_MODE_IN_CALL) {
- audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
- updateCallRouting(newDevice);
- }
- for (size_t i = 0; i < mOutputs.size(); i++) {
- audio_io_handle_t output = mOutputs.keyAt(i);
- if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
- audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i),
- true /*fromCache*/);
- // do not force device change on duplicated output because if device is 0, it will
- // also force a device 0 for the two outputs it is duplicated to which may override
- // a valid device selection on those outputs.
- bool force = !mOutputs.valueAt(i)->isDuplicated()
- && (!deviceDistinguishesOnAddress(device)
- // always force when disconnecting (a non-duplicated device)
- || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
- setOutputDevice(output, newDevice, force, 0);
- }
- }
-
- mpClientInterface->onAudioPortListUpdate();
- return NO_ERROR;
- } // end if is output device
-
- // handle input devices
- if (audio_is_input_device(device)) {
- SortedVector <audio_io_handle_t> inputs;
-
- ssize_t index = mAvailableInputDevices.indexOf(devDesc);
- switch (state)
- {
- // handle input device connection
- case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
- if (index >= 0) {
- ALOGW("setDeviceConnectionState() device already connected: %d", device);
- return INVALID_OPERATION;
- }
- sp<HwModule> module = getModuleForDevice(device);
- if (module == NULL) {
- ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
- device);
- return INVALID_OPERATION;
- }
- if (checkInputsForDevice(device, state, inputs, devDesc->mAddress) != NO_ERROR) {
- return INVALID_OPERATION;
- }
-
- index = mAvailableInputDevices.add(devDesc);
- if (index >= 0) {
- mAvailableInputDevices[index]->mId = nextUniqueId();
- mAvailableInputDevices[index]->mModule = module;
- } else {
- return NO_MEMORY;
- }
-
- // Set connect to HALs
- AudioParameter param = AudioParameter(devDesc->mAddress);
- param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
- mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
-
- } break;
-
- // handle input device disconnection
- case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
- if (index < 0) {
- ALOGW("setDeviceConnectionState() device not connected: %d", device);
- return INVALID_OPERATION;
- }
-
- ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
-
- // Set Disconnect to HALs
- AudioParameter param = AudioParameter(devDesc->mAddress);
- param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
- mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
-
- checkInputsForDevice(device, state, inputs, devDesc->mAddress);
- mAvailableInputDevices.remove(devDesc);
-
- } break;
-
- default:
- ALOGE("setDeviceConnectionState() invalid state: %x", state);
- return BAD_VALUE;
- }
-
- closeAllInputs();
-
- if (mPhoneState == AUDIO_MODE_IN_CALL) {
- audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
- updateCallRouting(newDevice);
- }
-
- mpClientInterface->onAudioPortListUpdate();
- return NO_ERROR;
- } // end if is input device
-
- ALOGW("setDeviceConnectionState() invalid device: %x", device);
- return BAD_VALUE;
-}
-
-audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
- const char *device_address)
-{
- sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address);
- DeviceVector *deviceVector;
-
- if (audio_is_output_device(device)) {
- deviceVector = &mAvailableOutputDevices;
- } else if (audio_is_input_device(device)) {
- deviceVector = &mAvailableInputDevices;
- } else {
- ALOGW("getDeviceConnectionState() invalid device type %08x", device);
- return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
- }
-
- ssize_t index = deviceVector->indexOf(devDesc);
- if (index >= 0) {
- return AUDIO_POLICY_DEVICE_STATE_AVAILABLE;
- } else {
- return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
- }
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::getDeviceDescriptor(
- const audio_devices_t device,
- const char *device_address)
-{
- String8 address = (device_address == NULL) ? String8("") : String8(device_address);
- // handle legacy remote submix case where the address was not always specified
- if (deviceDistinguishesOnAddress(device) && (address.length() == 0)) {
- address = String8("0");
- }
-
- for (size_t i = 0; i < mHwModules.size(); i++) {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- DeviceVector deviceList =
- mHwModules[i]->mDeclaredDevices.getDevicesFromTypeAddr(device, address);
- if (!deviceList.isEmpty()) {
- return deviceList.itemAt(0);
- }
- deviceList = mHwModules[i]->mDeclaredDevices.getDevicesFromType(device);
- if (!deviceList.isEmpty()) {
- return deviceList.itemAt(0);
- }
- }
-
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
- devDesc->mAddress = address;
- return devDesc;
-}
-
-void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs)
-{
- bool createTxPatch = false;
- struct audio_patch patch;
- patch.num_sources = 1;
- patch.num_sinks = 1;
- status_t status;
- audio_patch_handle_t afPatchHandle;
- DeviceVector deviceList;
-
- audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
- ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice);
-
- // release existing RX patch if any
- if (mCallRxPatch != 0) {
- mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
- mCallRxPatch.clear();
- }
- // release TX patch if any
- if (mCallTxPatch != 0) {
- mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
- mCallTxPatch.clear();
- }
-
- // If the RX device is on the primary HW module, then use legacy routing method for voice calls
- // via setOutputDevice() on primary output.
- // Otherwise, create two audio patches for TX and RX path.
- if (availablePrimaryOutputDevices() & rxDevice) {
- setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs);
- // If the TX device is also on the primary HW module, setOutputDevice() will take care
- // of it due to legacy implementation. If not, create a patch.
- if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN)
- == AUDIO_DEVICE_NONE) {
- createTxPatch = true;
- }
- } else {
- // create RX path audio patch
- deviceList = mAvailableOutputDevices.getDevicesFromType(rxDevice);
- ALOG_ASSERT(!deviceList.isEmpty(),
- "updateCallRouting() selected device not in output device list");
- sp<DeviceDescriptor> rxSinkDeviceDesc = deviceList.itemAt(0);
- deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX);
- ALOG_ASSERT(!deviceList.isEmpty(),
- "updateCallRouting() no telephony RX device");
- sp<DeviceDescriptor> rxSourceDeviceDesc = deviceList.itemAt(0);
-
- rxSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
- rxSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
-
- // request to reuse existing output stream if one is already opened to reach the RX device
- SortedVector<audio_io_handle_t> outputs =
- getOutputsForDevice(rxDevice, mOutputs);
- audio_io_handle_t output = selectOutput(outputs,
- AUDIO_OUTPUT_FLAG_NONE,
- AUDIO_FORMAT_INVALID);
- if (output != AUDIO_IO_HANDLE_NONE) {
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- ALOG_ASSERT(!outputDesc->isDuplicated(),
- "updateCallRouting() RX device output is duplicated");
- outputDesc->toAudioPortConfig(&patch.sources[1]);
- patch.num_sources = 2;
- }
-
- afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0);
- ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch",
- status);
- if (status == NO_ERROR) {
- mCallRxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &patch, mUidCached);
- mCallRxPatch->mAfPatchHandle = afPatchHandle;
- mCallRxPatch->mUid = mUidCached;
- }
- createTxPatch = true;
- }
- if (createTxPatch) {
-
- struct audio_patch patch;
- patch.num_sources = 1;
- patch.num_sinks = 1;
- deviceList = mAvailableInputDevices.getDevicesFromType(txDevice);
- ALOG_ASSERT(!deviceList.isEmpty(),
- "updateCallRouting() selected device not in input device list");
- sp<DeviceDescriptor> txSourceDeviceDesc = deviceList.itemAt(0);
- txSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
- deviceList = mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX);
- ALOG_ASSERT(!deviceList.isEmpty(),
- "updateCallRouting() no telephony TX device");
- sp<DeviceDescriptor> txSinkDeviceDesc = deviceList.itemAt(0);
- txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
-
- SortedVector<audio_io_handle_t> outputs =
- getOutputsForDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, mOutputs);
- audio_io_handle_t output = selectOutput(outputs,
- AUDIO_OUTPUT_FLAG_NONE,
- AUDIO_FORMAT_INVALID);
- // request to reuse existing output stream if one is already opened to reach the TX
- // path output device
- if (output != AUDIO_IO_HANDLE_NONE) {
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- ALOG_ASSERT(!outputDesc->isDuplicated(),
- "updateCallRouting() RX device output is duplicated");
- outputDesc->toAudioPortConfig(&patch.sources[1]);
- patch.num_sources = 2;
- }
-
- afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0);
- ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch",
- status);
- if (status == NO_ERROR) {
- mCallTxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &patch, mUidCached);
- mCallTxPatch->mAfPatchHandle = afPatchHandle;
- mCallTxPatch->mUid = mUidCached;
- }
- }
-}
-
-void AudioPolicyManager::setPhoneState(audio_mode_t state)
-{
- ALOGV("setPhoneState() state %d", state);
- if (state < 0 || state >= AUDIO_MODE_CNT) {
- ALOGW("setPhoneState() invalid state %d", state);
- return;
- }
-
- if (state == mPhoneState ) {
- ALOGW("setPhoneState() setting same state %d", state);
- return;
- }
-
- // if leaving call state, handle special case of active streams
- // pertaining to sonification strategy see handleIncallSonification()
- if (isInCall()) {
- ALOGV("setPhoneState() in call state management: new state is %d", state);
- for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
- if (stream == AUDIO_STREAM_PATCH) {
- continue;
- }
- handleIncallSonification((audio_stream_type_t)stream, false, true);
- }
-
- // force reevaluating accessibility routing when call starts
- mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
- }
-
- // store previous phone state for management of sonification strategy below
- int oldState = mPhoneState;
- mPhoneState = state;
- bool force = false;
-
- // are we entering or starting a call
- if (!isStateInCall(oldState) && isStateInCall(state)) {
- ALOGV(" Entering call in setPhoneState()");
- // force routing command to audio hardware when starting a call
- // even if no device change is needed
- force = true;
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
- mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
- sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
- }
- } else if (isStateInCall(oldState) && !isStateInCall(state)) {
- ALOGV(" Exiting call in setPhoneState()");
- // force routing command to audio hardware when exiting a call
- // even if no device change is needed
- force = true;
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
- mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
- sVolumeProfiles[AUDIO_STREAM_DTMF][j];
- }
- } else if (isStateInCall(state) && (state != oldState)) {
- ALOGV(" Switching between telephony and VoIP in setPhoneState()");
- // force routing command to audio hardware when switching between telephony and VoIP
- // even if no device change is needed
- force = true;
- }
-
- // check for device and output changes triggered by new phone state
- checkA2dpSuspend();
- checkOutputForAllStrategies();
- updateDevicesAndOutputs();
-
- sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
-
- int delayMs = 0;
- if (isStateInCall(state)) {
- nsecs_t sysTime = systemTime();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
- // mute media and sonification strategies and delay device switch by the largest
- // latency of any output where either strategy is active.
- // This avoid sending the ring tone or music tail into the earpiece or headset.
- if ((desc->isStrategyActive(STRATEGY_MEDIA,
- SONIFICATION_HEADSET_MUSIC_DELAY,
- sysTime) ||
- desc->isStrategyActive(STRATEGY_SONIFICATION,
- SONIFICATION_HEADSET_MUSIC_DELAY,
- sysTime)) &&
- (delayMs < (int)desc->mLatency*2)) {
- delayMs = desc->mLatency*2;
- }
- setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
- setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
- getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
- setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
- setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
- getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
- }
- }
-
- // Note that despite the fact that getNewOutputDevice() is called on the primary output,
- // the device returned is not necessarily reachable via this output
- audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
- // force routing command to audio hardware when ending call
- // even if no device change is needed
- if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
- rxDevice = hwOutputDesc->device();
- }
-
- if (state == AUDIO_MODE_IN_CALL) {
- updateCallRouting(rxDevice, delayMs);
- } else if (oldState == AUDIO_MODE_IN_CALL) {
- if (mCallRxPatch != 0) {
- mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
- mCallRxPatch.clear();
- }
- if (mCallTxPatch != 0) {
- mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
- mCallTxPatch.clear();
- }
- setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
- } else {
- setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
- }
- // if entering in call state, handle special case of active streams
- // pertaining to sonification strategy see handleIncallSonification()
- if (isStateInCall(state)) {
- ALOGV("setPhoneState() in call state management: new state is %d", state);
- for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
- if (stream == AUDIO_STREAM_PATCH) {
- continue;
- }
- handleIncallSonification((audio_stream_type_t)stream, true, true);
- }
- }
-
- // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
- if (state == AUDIO_MODE_RINGTONE &&
- isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
- mLimitRingtoneVolume = true;
- } else {
- mLimitRingtoneVolume = false;
- }
-}
-
-void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage,
- audio_policy_forced_cfg_t config)
-{
- ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
-
- bool forceVolumeReeval = false;
- switch(usage) {
- case AUDIO_POLICY_FORCE_FOR_COMMUNICATION:
- if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO &&
- config != AUDIO_POLICY_FORCE_NONE) {
- ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
- return;
- }
- forceVolumeReeval = true;
- mForceUse[usage] = config;
- break;
- case AUDIO_POLICY_FORCE_FOR_MEDIA:
- if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP &&
- config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
- config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
- config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE &&
- config != AUDIO_POLICY_FORCE_NO_BT_A2DP && config != AUDIO_POLICY_FORCE_SPEAKER ) {
- ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
- return;
- }
- mForceUse[usage] = config;
- break;
- case AUDIO_POLICY_FORCE_FOR_RECORD:
- if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
- config != AUDIO_POLICY_FORCE_NONE) {
- ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
- return;
- }
- mForceUse[usage] = config;
- break;
- case AUDIO_POLICY_FORCE_FOR_DOCK:
- if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK &&
- config != AUDIO_POLICY_FORCE_BT_DESK_DOCK &&
- config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
- config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
- config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) {
- ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
- }
- forceVolumeReeval = true;
- mForceUse[usage] = config;
- break;
- case AUDIO_POLICY_FORCE_FOR_SYSTEM:
- if (config != AUDIO_POLICY_FORCE_NONE &&
- config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
- ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
- }
- forceVolumeReeval = true;
- mForceUse[usage] = config;
- break;
- case AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO:
- if (config != AUDIO_POLICY_FORCE_NONE &&
- config != AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED) {
- ALOGW("setForceUse() invalid config %d forHDMI_SYSTEM_AUDIO", config);
- }
- mForceUse[usage] = config;
- break;
- default:
- ALOGW("setForceUse() invalid usage %d", usage);
- break;
- }
-
- // check for device and output changes triggered by new force usage
- checkA2dpSuspend();
- checkOutputForAllStrategies();
- updateDevicesAndOutputs();
- if (mPhoneState == AUDIO_MODE_IN_CALL) {
- audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
- updateCallRouting(newDevice);
- }
- for (size_t i = 0; i < mOutputs.size(); i++) {
- audio_io_handle_t output = mOutputs.keyAt(i);
- audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/);
- if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
- setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
- }
- if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
- applyStreamVolumes(output, newDevice, 0, true);
- }
- }
-
- audio_io_handle_t activeInput = getActiveInput();
- if (activeInput != 0) {
- setInputDevice(activeInput, getNewInputDevice(activeInput));
- }
-
-}
-
-audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage)
-{
- return mForceUse[usage];
-}
-
-void AudioPolicyManager::setSystemProperty(const char* property, const char* value)
-{
- ALOGV("setSystemProperty() property %s, value %s", property, value);
-}
-
-// Find a direct output profile compatible with the parameters passed, even if the input flags do
-// not explicitly request a direct output
-sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getProfileForDirectOutput(
- audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags)
-{
- for (size_t i = 0; i < mHwModules.size(); i++) {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
- sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
- bool found = profile->isCompatibleProfile(device, String8(""), samplingRate,
- NULL /*updatedSamplingRate*/, format, channelMask,
- flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ?
- AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT);
- if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) {
- return profile;
- }
- }
- }
- return 0;
-}
-
-audio_io_handle_t AudioPolicyManager::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)
-{
- routing_strategy strategy = getStrategy(stream);
- audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
- ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
- device, stream, samplingRate, format, channelMask, flags);
-
- return getOutputForDevice(device, AUDIO_SESSION_ALLOCATE,
- stream, samplingRate,format, channelMask,
- flags, offloadInfo);
-}
-
-status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
- audio_io_handle_t *output,
- audio_session_t session,
- 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_attributes_t attributes;
- if (attr != NULL) {
- if (!isValidAttributes(attr)) {
- ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
- attr->usage, attr->content_type, attr->flags,
- attr->tags);
- return BAD_VALUE;
- }
- attributes = *attr;
- } else {
- if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) {
- ALOGE("getOutputForAttr(): invalid stream type");
- return BAD_VALUE;
- }
- stream_type_to_audio_attributes(*stream, &attributes);
- }
-
- for (size_t i = 0; i < mPolicyMixes.size(); i++) {
- sp<AudioOutputDescriptor> desc;
- if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_PLAYERS) {
- for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) {
- if ((RULE_MATCH_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
- mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage == attributes.usage) ||
- (RULE_EXCLUDE_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
- mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage != attributes.usage)) {
- desc = mPolicyMixes[i]->mOutput;
- break;
- }
- if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
- strncmp(attributes.tags + strlen("addr="),
- mPolicyMixes[i]->mMix.mRegistrationId.string(),
- AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
- desc = mPolicyMixes[i]->mOutput;
- break;
- }
- }
- } else if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_RECORDERS) {
- if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE &&
- strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
- strncmp(attributes.tags + strlen("addr="),
- mPolicyMixes[i]->mMix.mRegistrationId.string(),
- AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
- desc = mPolicyMixes[i]->mOutput;
- }
- }
- if (desc != 0) {
- if (!audio_is_linear_pcm(format)) {
- return BAD_VALUE;
- }
- desc->mPolicyMix = &mPolicyMixes[i]->mMix;
- *stream = streamTypefromAttributesInt(&attributes);
- *output = desc->mIoHandle;
- ALOGV("getOutputForAttr() returns output %d", *output);
- return NO_ERROR;
- }
- }
- if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
- ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
- return BAD_VALUE;
- }
-
- ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x",
- attributes.usage, attributes.content_type, attributes.tags, attributes.flags);
-
- routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes);
- audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
-
- if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
- flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
- }
-
- ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x",
- device, samplingRate, format, channelMask, flags);
-
- *stream = streamTypefromAttributesInt(&attributes);
- *output = getOutputForDevice(device, session, *stream,
- samplingRate, format, channelMask,
- flags, offloadInfo);
- if (*output == AUDIO_IO_HANDLE_NONE) {
- return INVALID_OPERATION;
- }
- return NO_ERROR;
-}
-
-audio_io_handle_t AudioPolicyManager::getOutputForDevice(
- audio_devices_t device,
- audio_session_t session __unused,
- 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 output = AUDIO_IO_HANDLE_NONE;
- uint32_t latency = 0;
- status_t status;
-
-#ifdef AUDIO_POLICY_TEST
- if (mCurOutput != 0) {
- ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
- mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
-
- if (mTestOutputs[mCurOutput] == 0) {
- ALOGV("getOutput() opening test output");
- sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
- outputDesc->mDevice = mTestDevice;
- outputDesc->mLatency = mTestLatencyMs;
- outputDesc->mFlags =
- (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
- outputDesc->mRefCount[stream] = 0;
- audio_config_t config = AUDIO_CONFIG_INITIALIZER;
- config.sample_rate = mTestSamplingRate;
- config.channel_mask = mTestChannels;
- config.format = mTestFormat;
- if (offloadInfo != NULL) {
- config.offload_info = *offloadInfo;
- }
- status = mpClientInterface->openOutput(0,
- &mTestOutputs[mCurOutput],
- &config,
- &outputDesc->mDevice,
- String8(""),
- &outputDesc->mLatency,
- outputDesc->mFlags);
- if (status == NO_ERROR) {
- outputDesc->mSamplingRate = config.sample_rate;
- outputDesc->mFormat = config.format;
- outputDesc->mChannelMask = config.channel_mask;
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"),mCurOutput);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
- addOutput(mTestOutputs[mCurOutput], outputDesc);
- }
- }
- return mTestOutputs[mCurOutput];
- }
-#endif //AUDIO_POLICY_TEST
-
- // open a direct output if required by specified parameters
- //force direct flag if offload flag is set: offloading implies a direct output stream
- // and all common behaviors are driven by checking only the direct flag
- // this should normally be set appropriately in the policy configuration file
- if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
- }
- if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
- flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
- }
- // only allow deep buffering for music stream type
- if (stream != AUDIO_STREAM_MUSIC) {
- flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
- }
-
- sp<IOProfile> profile;
-
- // skip direct output selection if the request can obviously be attached to a mixed output
- // and not explicitly requested
- if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
- audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE &&
- audio_channel_count_from_out_mask(channelMask) <= 2) {
- goto non_direct_output;
- }
-
- // Do not allow offloading if one non offloadable effect is enabled. This prevents from
- // creating an offloaded track and tearing it down immediately after start when audioflinger
- // detects there is an active non offloadable effect.
- // FIXME: We should check the audio session here but we do not have it in this context.
- // This may prevent offloading in rare situations where effects are left active by apps
- // in the background.
-
- if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
- !isNonOffloadableEffectEnabled()) {
- profile = getProfileForDirectOutput(device,
- samplingRate,
- format,
- channelMask,
- (audio_output_flags_t)flags);
- }
-
- if (profile != 0) {
- sp<AudioOutputDescriptor> outputDesc = NULL;
-
- for (size_t i = 0; i < mOutputs.size(); i++) {
- sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
- if (!desc->isDuplicated() && (profile == desc->mProfile)) {
- outputDesc = desc;
- // reuse direct output if currently open and configured with same parameters
- if ((samplingRate == outputDesc->mSamplingRate) &&
- (format == outputDesc->mFormat) &&
- (channelMask == outputDesc->mChannelMask)) {
- outputDesc->mDirectOpenCount++;
- ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
- return mOutputs.keyAt(i);
- }
- }
- }
- // close direct output if currently open and configured with different parameters
- if (outputDesc != NULL) {
- closeOutput(outputDesc->mIoHandle);
- }
- outputDesc = new AudioOutputDescriptor(profile);
- outputDesc->mDevice = device;
- outputDesc->mLatency = 0;
- outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
- audio_config_t config = AUDIO_CONFIG_INITIALIZER;
- config.sample_rate = samplingRate;
- config.channel_mask = channelMask;
- config.format = format;
- if (offloadInfo != NULL) {
- config.offload_info = *offloadInfo;
- }
- status = mpClientInterface->openOutput(profile->mModule->mHandle,
- &output,
- &config,
- &outputDesc->mDevice,
- String8(""),
- &outputDesc->mLatency,
- outputDesc->mFlags);
-
- // only accept an output with the requested parameters
- if (status != NO_ERROR ||
- (samplingRate != 0 && samplingRate != config.sample_rate) ||
- (format != AUDIO_FORMAT_DEFAULT && format != config.format) ||
- (channelMask != 0 && channelMask != config.channel_mask)) {
- ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
- "format %d %d, channelMask %04x %04x", output, samplingRate,
- outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
- outputDesc->mChannelMask);
- if (output != AUDIO_IO_HANDLE_NONE) {
- mpClientInterface->closeOutput(output);
- }
- // fall back to mixer output if possible when the direct output could not be open
- if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
- goto non_direct_output;
- }
- return AUDIO_IO_HANDLE_NONE;
- }
- outputDesc->mSamplingRate = config.sample_rate;
- outputDesc->mChannelMask = config.channel_mask;
- outputDesc->mFormat = config.format;
- outputDesc->mRefCount[stream] = 0;
- outputDesc->mStopTime[stream] = 0;
- outputDesc->mDirectOpenCount = 1;
-
- audio_io_handle_t srcOutput = getOutputForEffect();
- addOutput(output, outputDesc);
- audio_io_handle_t dstOutput = getOutputForEffect();
- if (dstOutput == output) {
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
- }
- mPreviousOutputs = mOutputs;
- ALOGV("getOutput() returns new direct output %d", output);
- mpClientInterface->onAudioPortListUpdate();
- return output;
- }
-
-non_direct_output:
-
- // ignoring channel mask due to downmix capability in mixer
-
- // open a non direct output
-
- // for non direct outputs, only PCM is supported
- if (audio_is_linear_pcm(format)) {
- // get which output is suitable for the specified stream. The actual
- // routing change will happen when startOutput() will be called
- SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
-
- // at this stage we should ignore the DIRECT flag as no direct output could be found earlier
- flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
- output = selectOutput(outputs, flags, format);
- }
- ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
- "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
-
- ALOGV("getOutput() returns output %d", output);
-
- return output;
-}
-
-audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
- audio_output_flags_t flags,
- audio_format_t format)
-{
- // select one output among several that provide a path to a particular device or set of
- // devices (the list was previously build by getOutputsForDevice()).
- // The priority is as follows:
- // 1: the output with the highest number of requested policy flags
- // 2: the primary output
- // 3: the first output in the list
-
- if (outputs.size() == 0) {
- return 0;
- }
- if (outputs.size() == 1) {
- return outputs[0];
- }
-
- int maxCommonFlags = 0;
- audio_io_handle_t outputFlags = 0;
- audio_io_handle_t outputPrimary = 0;
-
- for (size_t i = 0; i < outputs.size(); i++) {
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
- if (!outputDesc->isDuplicated()) {
- // if a valid format is specified, skip output if not compatible
- if (format != AUDIO_FORMAT_INVALID) {
- if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
- if (format != outputDesc->mFormat) {
- continue;
- }
- } else if (!audio_is_linear_pcm(format)) {
- continue;
- }
- }
-
- int commonFlags = popcount(outputDesc->mProfile->mFlags & flags);
- if (commonFlags > maxCommonFlags) {
- outputFlags = outputs[i];
- maxCommonFlags = commonFlags;
- ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
- }
- if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
- outputPrimary = outputs[i];
- }
- }
- }
-
- if (outputFlags != 0) {
- return outputFlags;
- }
- if (outputPrimary != 0) {
- return outputPrimary;
- }
-
- return outputs[0];
-}
-
-status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
-{
- ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- ALOGW("startOutput() unknown output %d", output);
- return BAD_VALUE;
- }
-
- // cannot start playback of STREAM_TTS if any other output is being used
- uint32_t beaconMuteLatency = 0;
- if (stream == AUDIO_STREAM_TTS) {
- ALOGV("\t found BEACON stream");
- if (isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {
- return INVALID_OPERATION;
- } else {
- beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
- }
- } else {
- // some playback other than beacon starts
- beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
- }
-
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
-
- // increment usage count for this stream on the requested output:
- // NOTE that the usage count is the same for duplicated output and hardware output which is
- // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
- outputDesc->changeRefCount(stream, 1);
-
- if (outputDesc->mRefCount[stream] == 1) {
- // starting an output being rerouted?
- audio_devices_t newDevice;
- if (outputDesc->mPolicyMix != NULL) {
- newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
- } else {
- newDevice = getNewOutputDevice(output, false /*fromCache*/);
- }
- routing_strategy strategy = getStrategy(stream);
- bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
- (strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||
- (beaconMuteLatency > 0);
- uint32_t waitMs = beaconMuteLatency;
- bool force = false;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
- if (desc != outputDesc) {
- // force a device change if any other output is managed by the same hw
- // module and has a current device selection that differs from selected device.
- // In this case, the audio HAL must receive the new device selection so that it can
- // change the device currently selected by the other active output.
- if (outputDesc->sharesHwModuleWith(desc) &&
- desc->device() != newDevice) {
- force = true;
- }
- // wait for audio on other active outputs to be presented when starting
- // a notification so that audio focus effect can propagate, or that a mute/unmute
- // event occurred for beacon
- uint32_t latency = desc->latency();
- if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
- waitMs = latency;
- }
- }
- }
- uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
-
- // handle special case for sonification while in call
- if (isInCall()) {
- handleIncallSonification(stream, true, false);
- }
-
- // apply volume rules for current stream and device if necessary
- checkAndSetVolume(stream,
- mStreams[stream].getVolumeIndex(newDevice),
- output,
- newDevice);
-
- // update the outputs if starting an output with a stream that can affect notification
- // routing
- handleNotificationRoutingForStream(stream);
-
- // Automatically enable the remote submix input when output is started on a re routing mix
- // of type MIX_TYPE_RECORDERS
- if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL &&
- outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
- setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- outputDesc->mPolicyMix->mRegistrationId);
- }
-
- // force reevaluating accessibility routing when ringtone or alarm starts
- if (strategy == STRATEGY_SONIFICATION) {
- mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
- }
-
- if (waitMs > muteWaitMs) {
- usleep((waitMs - muteWaitMs) * 2 * 1000);
- }
- }
- return NO_ERROR;
-}
-
-
-status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
-{
- ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- ALOGW("stopOutput() unknown output %d", output);
- return BAD_VALUE;
- }
-
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
-
- // always handle stream stop, check which stream type is stopping
- handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
-
- // handle special case for sonification while in call
- if (isInCall()) {
- handleIncallSonification(stream, false, false);
- }
-
- if (outputDesc->mRefCount[stream] > 0) {
- // decrement usage count of this stream on the output
- outputDesc->changeRefCount(stream, -1);
- // store time at which the stream was stopped - see isStreamActive()
- if (outputDesc->mRefCount[stream] == 0) {
- // Automatically disable the remote submix input when output is stopped on a
- // re routing mix of type MIX_TYPE_RECORDERS
- if (audio_is_remote_submix_device(outputDesc->mDevice) &&
- outputDesc->mPolicyMix != NULL &&
- outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
- setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- outputDesc->mPolicyMix->mRegistrationId);
- }
-
- outputDesc->mStopTime[stream] = systemTime();
- audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
- // delay the device switch by twice the latency because stopOutput() is executed when
- // the track stop() command is received and at that time the audio track buffer can
- // still contain data that needs to be drained. The latency only covers the audio HAL
- // and kernel buffers. Also the latency does not always include additional delay in the
- // audio path (audio DSP, CODEC ...)
- setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
-
- // force restoring the device selection on other active outputs if it differs from the
- // one being selected for this output
- for (size_t i = 0; i < mOutputs.size(); i++) {
- audio_io_handle_t curOutput = mOutputs.keyAt(i);
- sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
- if (curOutput != output &&
- desc->isActive() &&
- outputDesc->sharesHwModuleWith(desc) &&
- (newDevice != desc->device())) {
- setOutputDevice(curOutput,
- getNewOutputDevice(curOutput, false /*fromCache*/),
- true,
- outputDesc->mLatency*2);
- }
- }
- // update the outputs if stopping one with a stream that can affect notification routing
- handleNotificationRoutingForStream(stream);
- }
- return NO_ERROR;
- } else {
- ALOGW("stopOutput() refcount is already 0 for output %d", output);
- return INVALID_OPERATION;
- }
-}
-
-void AudioPolicyManager::releaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream __unused,
- audio_session_t session __unused)
-{
- ALOGV("releaseOutput() %d", output);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- ALOGW("releaseOutput() releasing unknown output %d", output);
- return;
- }
-
-#ifdef AUDIO_POLICY_TEST
- int testIndex = testOutputIndex(output);
- if (testIndex != 0) {
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
- if (outputDesc->isActive()) {
- mpClientInterface->closeOutput(output);
- mOutputs.removeItem(output);
- mTestOutputs[testIndex] = 0;
- }
- return;
- }
-#endif //AUDIO_POLICY_TEST
-
- sp<AudioOutputDescriptor> desc = mOutputs.valueAt(index);
- if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
- if (desc->mDirectOpenCount <= 0) {
- ALOGW("releaseOutput() invalid open count %d for output %d",
- desc->mDirectOpenCount, output);
- return;
- }
- if (--desc->mDirectOpenCount == 0) {
- closeOutput(output);
- // If effects where present on the output, audioflinger moved them to the primary
- // output by default: move them back to the appropriate output.
- audio_io_handle_t dstOutput = getOutputForEffect();
- if (dstOutput != mPrimaryOutput) {
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput);
- }
- mpClientInterface->onAudioPortListUpdate();
- }
- }
-}
-
-
-status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr,
- audio_io_handle_t *input,
- audio_session_t session,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_input_flags_t flags,
- input_type_t *inputType)
-{
- ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x,"
- "session %d, flags %#x",
- attr->source, samplingRate, format, channelMask, session, flags);
-
- *input = AUDIO_IO_HANDLE_NONE;
- *inputType = API_INPUT_INVALID;
- audio_devices_t device;
- // handle legacy remote submix case where the address was not always specified
- String8 address = String8("");
- bool isSoundTrigger = false;
- audio_source_t inputSource = attr->source;
- audio_source_t halInputSource;
- AudioMix *policyMix = NULL;
-
- if (inputSource == AUDIO_SOURCE_DEFAULT) {
- inputSource = AUDIO_SOURCE_MIC;
- }
- halInputSource = inputSource;
-
- if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX &&
- strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {
- device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
- address = String8(attr->tags + strlen("addr="));
- ssize_t index = mPolicyMixes.indexOfKey(address);
- if (index < 0) {
- ALOGW("getInputForAttr() no policy for address %s", address.string());
- return BAD_VALUE;
- }
- if (mPolicyMixes[index]->mMix.mMixType != MIX_TYPE_PLAYERS) {
- ALOGW("getInputForAttr() bad policy mix type for address %s", address.string());
- return BAD_VALUE;
- }
- policyMix = &mPolicyMixes[index]->mMix;
- *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
- } else {
- device = getDeviceAndMixForInputSource(inputSource, &policyMix);
- if (device == AUDIO_DEVICE_NONE) {
- ALOGW("getInputForAttr() could not find device for source %d", inputSource);
- return BAD_VALUE;
- }
- if (policyMix != NULL) {
- address = policyMix->mRegistrationId;
- if (policyMix->mMixType == MIX_TYPE_RECORDERS) {
- // there is an external policy, but this input is attached to a mix of recorders,
- // meaning it receives audio injected into the framework, so the recorder doesn't
- // know about it and is therefore considered "legacy"
- *inputType = API_INPUT_LEGACY;
- } else {
- // recording a mix of players defined by an external policy, we're rerouting for
- // an external policy
- *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
- }
- } else if (audio_is_remote_submix_device(device)) {
- address = String8("0");
- *inputType = API_INPUT_MIX_CAPTURE;
- } else {
- *inputType = API_INPUT_LEGACY;
- }
- // adapt channel selection to input source
- switch (inputSource) {
- case AUDIO_SOURCE_VOICE_UPLINK:
- channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
- break;
- case AUDIO_SOURCE_VOICE_DOWNLINK:
- channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
- break;
- case AUDIO_SOURCE_VOICE_CALL:
- channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
- break;
- default:
- break;
- }
- if (inputSource == AUDIO_SOURCE_HOTWORD) {
- ssize_t index = mSoundTriggerSessions.indexOfKey(session);
- if (index >= 0) {
- *input = mSoundTriggerSessions.valueFor(session);
- isSoundTrigger = true;
- flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD);
- ALOGV("SoundTrigger capture on session %d input %d", session, *input);
- } else {
- halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION;
- }
- }
- }
-
- sp<IOProfile> profile = getInputProfile(device, address,
- samplingRate, format, channelMask,
- flags);
- if (profile == 0) {
- //retry without flags
- audio_input_flags_t log_flags = flags;
- flags = AUDIO_INPUT_FLAG_NONE;
- profile = getInputProfile(device, address,
- samplingRate, format, channelMask,
- flags);
- if (profile == 0) {
- ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u,"
- "format %#x, channelMask 0x%X, flags %#x",
- device, samplingRate, format, channelMask, log_flags);
- return BAD_VALUE;
- }
- }
-
- if (profile->mModule->mHandle == 0) {
- ALOGE("getInputForAttr(): HW module %s not opened", profile->mModule->mName);
- return NO_INIT;
- }
-
- audio_config_t config = AUDIO_CONFIG_INITIALIZER;
- config.sample_rate = samplingRate;
- config.channel_mask = channelMask;
- config.format = format;
-
- status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
- input,
- &config,
- &device,
- address,
- halInputSource,
- flags);
-
- // only accept input with the exact requested set of parameters
- if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE ||
- (samplingRate != config.sample_rate) ||
- (format != config.format) ||
- (channelMask != config.channel_mask)) {
- ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d, channelMask %x",
- samplingRate, format, channelMask);
- if (*input != AUDIO_IO_HANDLE_NONE) {
- mpClientInterface->closeInput(*input);
- }
- return BAD_VALUE;
- }
-
- sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile);
- inputDesc->mInputSource = inputSource;
- inputDesc->mRefCount = 0;
- inputDesc->mOpenRefCount = 1;
- inputDesc->mSamplingRate = samplingRate;
- inputDesc->mFormat = format;
- inputDesc->mChannelMask = channelMask;
- inputDesc->mDevice = device;
- inputDesc->mSessions.add(session);
- inputDesc->mIsSoundTrigger = isSoundTrigger;
- inputDesc->mPolicyMix = policyMix;
-
- ALOGV("getInputForAttr() returns input type = %d", inputType);
-
- addInput(*input, inputDesc);
- mpClientInterface->onAudioPortListUpdate();
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::startInput(audio_io_handle_t input,
- audio_session_t session)
-{
- ALOGV("startInput() input %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- ALOGW("startInput() unknown input %d", input);
- return BAD_VALUE;
- }
- sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
-
- index = inputDesc->mSessions.indexOf(session);
- if (index < 0) {
- ALOGW("startInput() unknown session %d on input %d", session, input);
- return BAD_VALUE;
- }
-
- // virtual input devices are compatible with other input devices
- if (!isVirtualInputDevice(inputDesc->mDevice)) {
-
- // for a non-virtual input device, check if there is another (non-virtual) active input
- audio_io_handle_t activeInput = getActiveInput();
- if (activeInput != 0 && activeInput != input) {
-
- // If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed,
- // otherwise the active input continues and the new input cannot be started.
- sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
- if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
- ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput);
- stopInput(activeInput, activeDesc->mSessions.itemAt(0));
- releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
- } else {
- ALOGE("startInput(%d) failed: other input %d already started", input, activeInput);
- return INVALID_OPERATION;
- }
- }
- }
-
- if (inputDesc->mRefCount == 0) {
- if (activeInputsCount() == 0) {
- SoundTrigger::setCaptureState(true);
- }
- setInputDevice(input, getNewInputDevice(input), true /* force */);
-
- // automatically enable the remote submix output when input is started if not
- // used by a policy mix of type MIX_TYPE_RECORDERS
- // For remote submix (a virtual device), we open only one input per capture request.
- if (audio_is_remote_submix_device(inputDesc->mDevice)) {
- String8 address = String8("");
- if (inputDesc->mPolicyMix == NULL) {
- address = String8("0");
- } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
- address = inputDesc->mPolicyMix->mRegistrationId;
- }
- if (address != "") {
- setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- address);
- }
- }
- }
-
- ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
-
- inputDesc->mRefCount++;
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::stopInput(audio_io_handle_t input,
- audio_session_t session)
-{
- ALOGV("stopInput() input %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- ALOGW("stopInput() unknown input %d", input);
- return BAD_VALUE;
- }
- sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
-
- index = inputDesc->mSessions.indexOf(session);
- if (index < 0) {
- ALOGW("stopInput() unknown session %d on input %d", session, input);
- return BAD_VALUE;
- }
-
- if (inputDesc->mRefCount == 0) {
- ALOGW("stopInput() input %d already stopped", input);
- return INVALID_OPERATION;
- }
-
- inputDesc->mRefCount--;
- if (inputDesc->mRefCount == 0) {
-
- // automatically disable the remote submix output when input is stopped if not
- // used by a policy mix of type MIX_TYPE_RECORDERS
- if (audio_is_remote_submix_device(inputDesc->mDevice)) {
- String8 address = String8("");
- if (inputDesc->mPolicyMix == NULL) {
- address = String8("0");
- } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
- address = inputDesc->mPolicyMix->mRegistrationId;
- }
- if (address != "") {
- setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- address);
- }
- }
-
- resetInputDevice(input);
-
- if (activeInputsCount() == 0) {
- SoundTrigger::setCaptureState(false);
- }
- }
- return NO_ERROR;
-}
-
-void AudioPolicyManager::releaseInput(audio_io_handle_t input,
- audio_session_t session)
-{
- ALOGV("releaseInput() %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- ALOGW("releaseInput() releasing unknown input %d", input);
- return;
- }
- sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
- ALOG_ASSERT(inputDesc != 0);
-
- index = inputDesc->mSessions.indexOf(session);
- if (index < 0) {
- ALOGW("releaseInput() unknown session %d on input %d", session, input);
- return;
- }
- inputDesc->mSessions.remove(session);
- if (inputDesc->mOpenRefCount == 0) {
- ALOGW("releaseInput() invalid open ref count %d", inputDesc->mOpenRefCount);
- return;
- }
- inputDesc->mOpenRefCount--;
- if (inputDesc->mOpenRefCount > 0) {
- ALOGV("releaseInput() exit > 0");
- return;
- }
-
- closeInput(input);
- mpClientInterface->onAudioPortListUpdate();
- ALOGV("releaseInput() exit");
-}
-
-void AudioPolicyManager::closeAllInputs() {
- bool patchRemoved = false;
-
- for(size_t input_index = 0; input_index < mInputs.size(); input_index++) {
- sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index);
- ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
- if (patch_index >= 0) {
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patch_index);
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
- mAudioPatches.removeItemsAt(patch_index);
- patchRemoved = true;
- }
- mpClientInterface->closeInput(mInputs.keyAt(input_index));
- }
- mInputs.clear();
- nextAudioPortGeneration();
-
- if (patchRemoved) {
- mpClientInterface->onAudioPatchListUpdate();
- }
-}
-
-void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream,
- int indexMin,
- int indexMax)
-{
- ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
- if (indexMin < 0 || indexMin >= indexMax) {
- ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
- return;
- }
- mStreams[stream].mIndexMin = indexMin;
- mStreams[stream].mIndexMax = indexMax;
- //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now
- if (stream == AUDIO_STREAM_MUSIC) {
- mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMin = indexMin;
- mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMax = indexMax;
- }
-}
-
-status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream,
- int index,
- audio_devices_t device)
-{
-
- if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
- return BAD_VALUE;
- }
- if (!audio_is_output_device(device)) {
- return BAD_VALUE;
- }
-
- // Force max volume if stream cannot be muted
- if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
-
- ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
- stream, device, index);
-
- // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
- // clear all device specific values
- if (device == AUDIO_DEVICE_OUT_DEFAULT) {
- mStreams[stream].mIndexCur.clear();
- }
- mStreams[stream].mIndexCur.add(device, index);
-
- // update volume on all outputs whose current device is also selected by the same
- // strategy as the device specified by the caller
- audio_devices_t strategyDevice = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
-
-
- //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now
- audio_devices_t accessibilityDevice = AUDIO_DEVICE_NONE;
- if (stream == AUDIO_STREAM_MUSIC) {
- mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexCur.add(device, index);
- accessibilityDevice = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, true /*fromCache*/);
- }
- if ((device != AUDIO_DEVICE_OUT_DEFAULT) &&
- (device & (strategyDevice | accessibilityDevice)) == 0) {
- return NO_ERROR;
- }
- status_t status = NO_ERROR;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- audio_devices_t curDevice =
- getDeviceForVolume(mOutputs.valueAt(i)->device());
- if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) {
- status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
- if (volStatus != NO_ERROR) {
- status = volStatus;
- }
- }
- if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) {
- status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY,
- index, mOutputs.keyAt(i), curDevice);
- }
- }
- return status;
-}
-
-status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream,
- int *index,
- audio_devices_t device)
-{
- if (index == NULL) {
- return BAD_VALUE;
- }
- if (!audio_is_output_device(device)) {
- return BAD_VALUE;
- }
- // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to
- // the strategy the stream belongs to.
- if (device == AUDIO_DEVICE_OUT_DEFAULT) {
- device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
- }
- device = getDeviceForVolume(device);
-
- *index = mStreams[stream].getVolumeIndex(device);
- ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
- return NO_ERROR;
-}
-
-audio_io_handle_t AudioPolicyManager::selectOutputForEffects(
- const SortedVector<audio_io_handle_t>& outputs)
-{
- // select one output among several suitable for global effects.
- // The priority is as follows:
- // 1: An offloaded output. If the effect ends up not being offloadable,
- // AudioFlinger will invalidate the track and the offloaded output
- // will be closed causing the effect to be moved to a PCM output.
- // 2: A deep buffer output
- // 3: the first output in the list
-
- if (outputs.size() == 0) {
- return 0;
- }
-
- audio_io_handle_t outputOffloaded = 0;
- audio_io_handle_t outputDeepBuffer = 0;
-
- for (size_t i = 0; i < outputs.size(); i++) {
- sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
- ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags);
- if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- outputOffloaded = outputs[i];
- }
- if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
- outputDeepBuffer = outputs[i];
- }
- }
-
- ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d",
- outputOffloaded, outputDeepBuffer);
- if (outputOffloaded != 0) {
- return outputOffloaded;
- }
- if (outputDeepBuffer != 0) {
- return outputDeepBuffer;
- }
-
- return outputs[0];
-}
-
-audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc)
-{
- // apply simple rule where global effects are attached to the same output as MUSIC streams
-
- routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC);
- audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
- SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
-
- audio_io_handle_t output = selectOutputForEffects(dstOutputs);
- ALOGV("getOutputForEffect() got output %d for fx %s flags %x",
- output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags);
-
- return output;
-}
-
-status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc,
- audio_io_handle_t io,
- uint32_t strategy,
- int session,
- int id)
-{
- ssize_t index = mOutputs.indexOfKey(io);
- if (index < 0) {
- index = mInputs.indexOfKey(io);
- if (index < 0) {
- ALOGW("registerEffect() unknown io %d", io);
- return INVALID_OPERATION;
- }
- }
-
- if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
- ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
- desc->name, desc->memoryUsage);
- return INVALID_OPERATION;
- }
- mTotalEffectsMemory += desc->memoryUsage;
- ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
- desc->name, io, strategy, session, id);
- ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
-
- sp<EffectDescriptor> effectDesc = new EffectDescriptor();
- memcpy (&effectDesc->mDesc, desc, sizeof(effect_descriptor_t));
- effectDesc->mIo = io;
- effectDesc->mStrategy = (routing_strategy)strategy;
- effectDesc->mSession = session;
- effectDesc->mEnabled = false;
-
- mEffects.add(id, effectDesc);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::unregisterEffect(int id)
-{
- ssize_t index = mEffects.indexOfKey(id);
- if (index < 0) {
- ALOGW("unregisterEffect() unknown effect ID %d", id);
- return INVALID_OPERATION;
- }
-
- sp<EffectDescriptor> effectDesc = mEffects.valueAt(index);
-
- setEffectEnabled(effectDesc, false);
-
- if (mTotalEffectsMemory < effectDesc->mDesc.memoryUsage) {
- ALOGW("unregisterEffect() memory %d too big for total %d",
- effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
- effectDesc->mDesc.memoryUsage = mTotalEffectsMemory;
- }
- mTotalEffectsMemory -= effectDesc->mDesc.memoryUsage;
- ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
- effectDesc->mDesc.name, id, effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
-
- mEffects.removeItem(id);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled)
-{
- ssize_t index = mEffects.indexOfKey(id);
- if (index < 0) {
- ALOGW("unregisterEffect() unknown effect ID %d", id);
- return INVALID_OPERATION;
- }
-
- return setEffectEnabled(mEffects.valueAt(index), enabled);
-}
-
-status_t AudioPolicyManager::setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled)
-{
- if (enabled == effectDesc->mEnabled) {
- ALOGV("setEffectEnabled(%s) effect already %s",
- enabled?"true":"false", enabled?"enabled":"disabled");
- return INVALID_OPERATION;
- }
-
- if (enabled) {
- if (mTotalEffectsCpuLoad + effectDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
- ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
- effectDesc->mDesc.name, (float)effectDesc->mDesc.cpuLoad/10);
- return INVALID_OPERATION;
- }
- mTotalEffectsCpuLoad += effectDesc->mDesc.cpuLoad;
- ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
- } else {
- if (mTotalEffectsCpuLoad < effectDesc->mDesc.cpuLoad) {
- ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
- effectDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
- effectDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
- }
- mTotalEffectsCpuLoad -= effectDesc->mDesc.cpuLoad;
- ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
- }
- effectDesc->mEnabled = enabled;
- return NO_ERROR;
-}
-
-bool AudioPolicyManager::isNonOffloadableEffectEnabled()
-{
- for (size_t i = 0; i < mEffects.size(); i++) {
- sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
- if (effectDesc->mEnabled && (effectDesc->mStrategy == STRATEGY_MEDIA) &&
- ((effectDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
- ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
- effectDesc->mDesc.name, effectDesc->mSession);
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
-{
- nsecs_t sysTime = systemTime();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
- if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream,
- uint32_t inPastMs) const
-{
- nsecs_t sysTime = systemTime();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
- if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
- outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
- // do not consider re routing (when the output is going to a dynamic policy)
- // as "remote playback"
- if (outputDesc->mPolicyMix == NULL) {
- return true;
- }
- }
- }
- return false;
-}
-
-bool AudioPolicyManager::isSourceActive(audio_source_t source) const
-{
- for (size_t i = 0; i < mInputs.size(); i++) {
- const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
- if (inputDescriptor->mRefCount == 0) {
- continue;
- }
- if (inputDescriptor->mInputSource == (int)source) {
- return true;
- }
- // AUDIO_SOURCE_HOTWORD is equivalent to AUDIO_SOURCE_VOICE_RECOGNITION only if it
- // corresponds to an active capture triggered by a hardware hotword recognition
- if ((source == AUDIO_SOURCE_VOICE_RECOGNITION) &&
- (inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) {
- // FIXME: we should not assume that the first session is the active one and keep
- // activity count per session. Same in startInput().
- ssize_t index = mSoundTriggerSessions.indexOfKey(inputDescriptor->mSessions.itemAt(0));
- if (index >= 0) {
- return true;
- }
- }
- }
- return false;
-}
-
-// Register a list of custom mixes with their attributes and format.
-// When a mix is registered, corresponding input and output profiles are
-// added to the remote submix hw module. The profile contains only the
-// parameters (sampling rate, format...) specified by the mix.
-// The corresponding input remote submix device is also connected.
-//
-// When a remote submix device is connected, the address is checked to select the
-// appropriate profile and the corresponding input or output stream is opened.
-//
-// When capture starts, getInputForAttr() will:
-// - 1 look for a mix matching the address passed in attribtutes tags if any
-// - 2 if none found, getDeviceForInputSource() will:
-// - 2.1 look for a mix matching the attributes source
-// - 2.2 if none found, default to device selection by policy rules
-// At this time, the corresponding output remote submix device is also connected
-// and active playback use cases can be transferred to this mix if needed when reconnecting
-// after AudioTracks are invalidated
-//
-// When playback starts, getOutputForAttr() will:
-// - 1 look for a mix matching the address passed in attribtutes tags if any
-// - 2 if none found, look for a mix matching the attributes usage
-// - 3 if none found, default to device and output selection by policy rules.
-
-status_t AudioPolicyManager::registerPolicyMixes(Vector<AudioMix> mixes)
-{
- sp<HwModule> module;
- for (size_t i = 0; i < mHwModules.size(); i++) {
- if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
- mHwModules[i]->mHandle != 0) {
- module = mHwModules[i];
- break;
- }
- }
-
- if (module == 0) {
- return INVALID_OPERATION;
- }
-
- ALOGV("registerPolicyMixes() num mixes %d", mixes.size());
-
- for (size_t i = 0; i < mixes.size(); i++) {
- String8 address = mixes[i].mRegistrationId;
- ssize_t index = mPolicyMixes.indexOfKey(address);
- if (index >= 0) {
- ALOGE("registerPolicyMixes(): mix for address %s already registered", address.string());
- continue;
- }
- audio_config_t outputConfig = mixes[i].mFormat;
- audio_config_t inputConfig = mixes[i].mFormat;
- // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in
- // stereo and let audio flinger do the channel conversion if needed.
- outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
- inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
- module->addOutputProfile(address, &outputConfig,
- AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address);
- module->addInputProfile(address, &inputConfig,
- AUDIO_DEVICE_IN_REMOTE_SUBMIX, address);
- sp<AudioPolicyMix> policyMix = new AudioPolicyMix();
- policyMix->mMix = mixes[i];
- mPolicyMixes.add(address, policyMix);
- if (mixes[i].mMixType == MIX_TYPE_PLAYERS) {
- setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- address.string());
- } else {
- setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- address.string());
- }
- }
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes)
-{
- sp<HwModule> module;
- for (size_t i = 0; i < mHwModules.size(); i++) {
- if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
- mHwModules[i]->mHandle != 0) {
- module = mHwModules[i];
- break;
- }
- }
-
- if (module == 0) {
- return INVALID_OPERATION;
- }
-
- ALOGV("unregisterPolicyMixes() num mixes %d", mixes.size());
-
- for (size_t i = 0; i < mixes.size(); i++) {
- String8 address = mixes[i].mRegistrationId;
- ssize_t index = mPolicyMixes.indexOfKey(address);
- if (index < 0) {
- ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string());
- continue;
- }
-
- mPolicyMixes.removeItemsAt(index);
-
- if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) ==
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
- {
- setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- address.string());
- }
-
- if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
- {
- setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- address.string());
- }
- module->removeOutputProfile(address);
- module->removeInputProfile(address);
- }
- return NO_ERROR;
-}
-
-
-status_t AudioPolicyManager::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
- result.append(buffer);
-
- snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput);
- result.append(buffer);
- snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for communications %d\n",
- mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n",
- mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO]);
- result.append(buffer);
-
- snprintf(buffer, SIZE, " Available output devices:\n");
- result.append(buffer);
- write(fd, result.string(), result.size());
- for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) {
- mAvailableOutputDevices[i]->dump(fd, 2, i);
- }
- snprintf(buffer, SIZE, "\n Available input devices:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
- mAvailableInputDevices[i]->dump(fd, 2, i);
- }
-
- snprintf(buffer, SIZE, "\nHW Modules dump:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mHwModules.size(); i++) {
- snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1);
- write(fd, buffer, strlen(buffer));
- mHwModules[i]->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nOutputs dump:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mOutputs.size(); i++) {
- snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mOutputs.valueAt(i)->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nInputs dump:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mInputs.size(); i++) {
- snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mInputs.valueAt(i)->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nStreams dump:\n");
- write(fd, buffer, strlen(buffer));
- snprintf(buffer, SIZE,
- " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < AUDIO_STREAM_CNT; i++) {
- snprintf(buffer, SIZE, " %02zu ", i);
- write(fd, buffer, strlen(buffer));
- mStreams[i].dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
- (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
- write(fd, buffer, strlen(buffer));
-
- snprintf(buffer, SIZE, "Registered effects:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mEffects.size(); i++) {
- snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mEffects.valueAt(i)->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nAudio Patches:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mAudioPatches.size(); i++) {
- mAudioPatches[i]->dump(fd, 2, i);
- }
-
- return NO_ERROR;
-}
-
-// This function checks for the parameters which can be offloaded.
-// This can be enhanced depending on the capability of the DSP and policy
-// of the system.
-bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
-{
- ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
- " BitRate=%u, duration=%" PRId64 " us, has_video=%d",
- offloadInfo.sample_rate, offloadInfo.channel_mask,
- offloadInfo.format,
- offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
- offloadInfo.has_video);
-
- // Check if offload has been disabled
- char propValue[PROPERTY_VALUE_MAX];
- if (property_get("audio.offload.disable", propValue, "0")) {
- if (atoi(propValue) != 0) {
- ALOGV("offload disabled by audio.offload.disable=%s", propValue );
- return false;
- }
- }
-
- // Check if stream type is music, then only allow offload as of now.
- if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
- {
- ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
- return false;
- }
-
- //TODO: enable audio offloading with video when ready
- if (offloadInfo.has_video)
- {
- ALOGV("isOffloadSupported: has_video == true, returning false");
- return false;
- }
-
- //If duration is less than minimum value defined in property, return false
- if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
- if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
- ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
- return false;
- }
- } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
- ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
- return false;
- }
-
- // Do not allow offloading if one non offloadable effect is enabled. This prevents from
- // creating an offloaded track and tearing it down immediately after start when audioflinger
- // detects there is an active non offloadable effect.
- // FIXME: We should check the audio session here but we do not have it in this context.
- // This may prevent offloading in rare situations where effects are left active by apps
- // in the background.
- if (isNonOffloadableEffectEnabled()) {
- return false;
- }
-
- // See if there is a profile to support this.
- // AUDIO_DEVICE_NONE
- sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
- offloadInfo.sample_rate,
- offloadInfo.format,
- offloadInfo.channel_mask,
- AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
- ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
- return (profile != 0);
-}
-
-status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role,
- audio_port_type_t type,
- unsigned int *num_ports,
- struct audio_port *ports,
- unsigned int *generation)
-{
- if (num_ports == NULL || (*num_ports != 0 && ports == NULL) ||
- generation == NULL) {
- return BAD_VALUE;
- }
- ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports);
- if (ports == NULL) {
- *num_ports = 0;
- }
-
- size_t portsWritten = 0;
- size_t portsMax = *num_ports;
- *num_ports = 0;
- if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) {
- if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
- for (size_t i = 0;
- i < mAvailableOutputDevices.size() && portsWritten < portsMax; i++) {
- mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]);
- }
- *num_ports += mAvailableOutputDevices.size();
- }
- if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
- for (size_t i = 0;
- i < mAvailableInputDevices.size() && portsWritten < portsMax; i++) {
- mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]);
- }
- *num_ports += mAvailableInputDevices.size();
- }
- }
- if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) {
- if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
- for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) {
- mInputs[i]->toAudioPort(&ports[portsWritten++]);
- }
- *num_ports += mInputs.size();
- }
- if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
- size_t numOutputs = 0;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- if (!mOutputs[i]->isDuplicated()) {
- numOutputs++;
- if (portsWritten < portsMax) {
- mOutputs[i]->toAudioPort(&ports[portsWritten++]);
- }
- }
- }
- *num_ports += numOutputs;
- }
- }
- *generation = curAudioPortGeneration();
- ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports);
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused)
-{
- return NO_ERROR;
-}
-
-sp<AudioPolicyManager::AudioOutputDescriptor> AudioPolicyManager::getOutputFromId(
- audio_port_handle_t id) const
-{
- sp<AudioOutputDescriptor> outputDesc = NULL;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- outputDesc = mOutputs.valueAt(i);
- if (outputDesc->mId == id) {
- break;
- }
- }
- return outputDesc;
-}
-
-sp<AudioPolicyManager::AudioInputDescriptor> AudioPolicyManager::getInputFromId(
- audio_port_handle_t id) const
-{
- sp<AudioInputDescriptor> inputDesc = NULL;
- for (size_t i = 0; i < mInputs.size(); i++) {
- inputDesc = mInputs.valueAt(i);
- if (inputDesc->mId == id) {
- break;
- }
- }
- return inputDesc;
-}
-
-sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleForDevice(
- audio_devices_t device) const
-{
- sp <HwModule> module;
-
- for (size_t i = 0; i < mHwModules.size(); i++) {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- if (audio_is_output_device(device)) {
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
- {
- if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) {
- return mHwModules[i];
- }
- }
- } else {
- for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) {
- if (mHwModules[i]->mInputProfiles[j]->mSupportedDevices.types() &
- device & ~AUDIO_DEVICE_BIT_IN) {
- return mHwModules[i];
- }
- }
- }
- }
- return module;
-}
-
-sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleFromName(const char *name) const
-{
- sp <HwModule> module;
-
- for (size_t i = 0; i < mHwModules.size(); i++)
- {
- if (strcmp(mHwModules[i]->mName, name) == 0) {
- return mHwModules[i];
- }
- }
- return module;
-}
-
-audio_devices_t AudioPolicyManager::availablePrimaryOutputDevices()
-{
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
- audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types();
- return devices & mAvailableOutputDevices.types();
-}
-
-audio_devices_t AudioPolicyManager::availablePrimaryInputDevices()
-{
- audio_module_handle_t primaryHandle =
- mOutputs.valueFor(mPrimaryOutput)->mProfile->mModule->mHandle;
- audio_devices_t devices = AUDIO_DEVICE_NONE;
- for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
- if (mAvailableInputDevices[i]->mModule->mHandle == primaryHandle) {
- devices |= mAvailableInputDevices[i]->mDeviceType;
- }
- }
- return devices;
-}
-
-status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch,
- audio_patch_handle_t *handle,
- uid_t uid)
-{
- ALOGV("createAudioPatch()");
-
- if (handle == NULL || patch == NULL) {
- return BAD_VALUE;
- }
- ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks);
-
- if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX ||
- patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) {
- return BAD_VALUE;
- }
- // only one source per audio patch supported for now
- if (patch->num_sources > 1) {
- return INVALID_OPERATION;
- }
-
- if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE) {
- return INVALID_OPERATION;
- }
- for (size_t i = 0; i < patch->num_sinks; i++) {
- if (patch->sinks[i].role != AUDIO_PORT_ROLE_SINK) {
- return INVALID_OPERATION;
- }
- }
-
- sp<AudioPatch> patchDesc;
- ssize_t index = mAudioPatches.indexOfKey(*handle);
-
- ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id,
- patch->sources[0].role,
- patch->sources[0].type);
-#if LOG_NDEBUG == 0
- for (size_t i = 0; i < patch->num_sinks; i++) {
- ALOGV("createAudioPatch sink %d: id %d role %d type %d", i, patch->sinks[i].id,
- patch->sinks[i].role,
- patch->sinks[i].type);
- }
-#endif
-
- if (index >= 0) {
- patchDesc = mAudioPatches.valueAt(index);
- ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
- mUidCached, patchDesc->mUid, uid);
- if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
- return INVALID_OPERATION;
- }
- } else {
- *handle = 0;
- }
-
- if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
- sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id);
- if (outputDesc == NULL) {
- ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
- return BAD_VALUE;
- }
- ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports",
- outputDesc->mIoHandle);
- if (patchDesc != 0) {
- if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
- ALOGV("createAudioPatch() source id differs for patch current id %d new id %d",
- patchDesc->mPatch.sources[0].id, patch->sources[0].id);
- return BAD_VALUE;
- }
- }
- DeviceVector devices;
- for (size_t i = 0; i < patch->num_sinks; i++) {
- // Only support mix to devices connection
- // TODO add support for mix to mix connection
- if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
- ALOGV("createAudioPatch() source mix but sink is not a device");
- return INVALID_OPERATION;
- }
- sp<DeviceDescriptor> devDesc =
- mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
- if (devDesc == 0) {
- ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id);
- return BAD_VALUE;
- }
-
- if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
- devDesc->mAddress,
- patch->sources[0].sample_rate,
- NULL, // updatedSamplingRate
- patch->sources[0].format,
- patch->sources[0].channel_mask,
- AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) {
- ALOGV("createAudioPatch() profile not supported for device %08x",
- devDesc->mDeviceType);
- return INVALID_OPERATION;
- }
- devices.add(devDesc);
- }
- if (devices.size() == 0) {
- return INVALID_OPERATION;
- }
-
- // TODO: reconfigure output format and channels here
- ALOGV("createAudioPatch() setting device %08x on output %d",
- devices.types(), outputDesc->mIoHandle);
- setOutputDevice(outputDesc->mIoHandle, devices.types(), true, 0, handle);
- index = mAudioPatches.indexOfKey(*handle);
- if (index >= 0) {
- if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
- ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided");
- }
- patchDesc = mAudioPatches.valueAt(index);
- patchDesc->mUid = uid;
- ALOGV("createAudioPatch() success");
- } else {
- ALOGW("createAudioPatch() setOutputDevice() failed to create a patch");
- return INVALID_OPERATION;
- }
- } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
- if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
- // input device to input mix connection
- // only one sink supported when connecting an input device to a mix
- if (patch->num_sinks > 1) {
- return INVALID_OPERATION;
- }
- sp<AudioInputDescriptor> inputDesc = getInputFromId(patch->sinks[0].id);
- if (inputDesc == NULL) {
- return BAD_VALUE;
- }
- if (patchDesc != 0) {
- if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) {
- return BAD_VALUE;
- }
- }
- sp<DeviceDescriptor> devDesc =
- mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
- if (devDesc == 0) {
- return BAD_VALUE;
- }
-
- if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
- devDesc->mAddress,
- patch->sinks[0].sample_rate,
- NULL, /*updatedSampleRate*/
- patch->sinks[0].format,
- patch->sinks[0].channel_mask,
- // FIXME for the parameter type,
- // and the NONE
- (audio_output_flags_t)
- AUDIO_INPUT_FLAG_NONE)) {
- return INVALID_OPERATION;
- }
- // TODO: reconfigure output format and channels here
- ALOGV("createAudioPatch() setting device %08x on output %d",
- devDesc->mDeviceType, inputDesc->mIoHandle);
- setInputDevice(inputDesc->mIoHandle, devDesc->mDeviceType, true, handle);
- index = mAudioPatches.indexOfKey(*handle);
- if (index >= 0) {
- if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
- ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided");
- }
- patchDesc = mAudioPatches.valueAt(index);
- patchDesc->mUid = uid;
- ALOGV("createAudioPatch() success");
- } else {
- ALOGW("createAudioPatch() setInputDevice() failed to create a patch");
- return INVALID_OPERATION;
- }
- } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
- // device to device connection
- if (patchDesc != 0) {
- if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
- return BAD_VALUE;
- }
- }
- sp<DeviceDescriptor> srcDeviceDesc =
- mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
- if (srcDeviceDesc == 0) {
- return BAD_VALUE;
- }
-
- //update source and sink with our own data as the data passed in the patch may
- // be incomplete.
- struct audio_patch newPatch = *patch;
- srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
-
- for (size_t i = 0; i < patch->num_sinks; i++) {
- if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
- ALOGV("createAudioPatch() source device but one sink is not a device");
- return INVALID_OPERATION;
- }
-
- sp<DeviceDescriptor> sinkDeviceDesc =
- mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
- if (sinkDeviceDesc == 0) {
- return BAD_VALUE;
- }
- sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
-
- if (srcDeviceDesc->mModule != sinkDeviceDesc->mModule) {
- // only one sink supported when connected devices across HW modules
- if (patch->num_sinks > 1) {
- return INVALID_OPERATION;
- }
- SortedVector<audio_io_handle_t> outputs =
- getOutputsForDevice(sinkDeviceDesc->mDeviceType,
- mOutputs);
- // if the sink device is reachable via an opened output stream, request to go via
- // this output stream by adding a second source to the patch description
- audio_io_handle_t output = selectOutput(outputs,
- AUDIO_OUTPUT_FLAG_NONE,
- AUDIO_FORMAT_INVALID);
- if (output != AUDIO_IO_HANDLE_NONE) {
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- if (outputDesc->isDuplicated()) {
- return INVALID_OPERATION;
- }
- outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]);
- newPatch.num_sources = 2;
- }
- }
- }
- // TODO: check from routing capabilities in config file and other conflicting patches
-
- audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- if (index >= 0) {
- afPatchHandle = patchDesc->mAfPatchHandle;
- }
-
- status_t status = mpClientInterface->createAudioPatch(&newPatch,
- &afPatchHandle,
- 0);
- ALOGV("createAudioPatch() patch panel returned %d patchHandle %d",
- status, afPatchHandle);
- if (status == NO_ERROR) {
- if (index < 0) {
- patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &newPatch, uid);
- addAudioPatch(patchDesc->mHandle, patchDesc);
- } else {
- patchDesc->mPatch = newPatch;
- }
- patchDesc->mAfPatchHandle = afPatchHandle;
- *handle = patchDesc->mHandle;
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- } else {
- ALOGW("createAudioPatch() patch panel could not connect device patch, error %d",
- status);
- return INVALID_OPERATION;
- }
- } else {
- return BAD_VALUE;
- }
- } else {
- return BAD_VALUE;
- }
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle,
- uid_t uid)
-{
- ALOGV("releaseAudioPatch() patch %d", handle);
-
- ssize_t index = mAudioPatches.indexOfKey(handle);
-
- if (index < 0) {
- return BAD_VALUE;
- }
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
- mUidCached, patchDesc->mUid, uid);
- if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
- return INVALID_OPERATION;
- }
-
- struct audio_patch *patch = &patchDesc->mPatch;
- patchDesc->mUid = mUidCached;
- if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
- sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id);
- if (outputDesc == NULL) {
- ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id);
- return BAD_VALUE;
- }
-
- setOutputDevice(outputDesc->mIoHandle,
- getNewOutputDevice(outputDesc->mIoHandle, true /*fromCache*/),
- true,
- 0,
- NULL);
- } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
- if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
- sp<AudioInputDescriptor> inputDesc = getInputFromId(patch->sinks[0].id);
- if (inputDesc == NULL) {
- ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id);
- return BAD_VALUE;
- }
- setInputDevice(inputDesc->mIoHandle,
- getNewInputDevice(inputDesc->mIoHandle),
- true,
- NULL);
- } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
- audio_patch_handle_t afPatchHandle = patchDesc->mAfPatchHandle;
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
- ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d",
- status, patchDesc->mAfPatchHandle);
- removeAudioPatch(patchDesc->mHandle);
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- } else {
- return BAD_VALUE;
- }
- } else {
- return BAD_VALUE;
- }
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches,
- struct audio_patch *patches,
- unsigned int *generation)
-{
- if (num_patches == NULL || (*num_patches != 0 && patches == NULL) ||
- generation == NULL) {
- return BAD_VALUE;
- }
- ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu",
- *num_patches, patches, mAudioPatches.size());
- if (patches == NULL) {
- *num_patches = 0;
- }
-
- size_t patchesWritten = 0;
- size_t patchesMax = *num_patches;
- for (size_t i = 0;
- i < mAudioPatches.size() && patchesWritten < patchesMax; i++) {
- patches[patchesWritten] = mAudioPatches[i]->mPatch;
- patches[patchesWritten++].id = mAudioPatches[i]->mHandle;
- ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d",
- i, mAudioPatches[i]->mPatch.num_sources, mAudioPatches[i]->mPatch.num_sinks);
- }
- *num_patches = mAudioPatches.size();
-
- *generation = curAudioPortGeneration();
- ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches);
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config)
-{
- ALOGV("setAudioPortConfig()");
-
- if (config == NULL) {
- return BAD_VALUE;
- }
- ALOGV("setAudioPortConfig() on port handle %d", config->id);
- // Only support gain configuration for now
- if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) {
- return INVALID_OPERATION;
- }
-
- sp<AudioPortConfig> audioPortConfig;
- if (config->type == AUDIO_PORT_TYPE_MIX) {
- if (config->role == AUDIO_PORT_ROLE_SOURCE) {
- sp<AudioOutputDescriptor> outputDesc = getOutputFromId(config->id);
- if (outputDesc == NULL) {
- return BAD_VALUE;
- }
- ALOG_ASSERT(!outputDesc->isDuplicated(),
- "setAudioPortConfig() called on duplicated output %d",
- outputDesc->mIoHandle);
- audioPortConfig = outputDesc;
- } else if (config->role == AUDIO_PORT_ROLE_SINK) {
- sp<AudioInputDescriptor> inputDesc = getInputFromId(config->id);
- if (inputDesc == NULL) {
- return BAD_VALUE;
- }
- audioPortConfig = inputDesc;
- } else {
- return BAD_VALUE;
- }
- } else if (config->type == AUDIO_PORT_TYPE_DEVICE) {
- sp<DeviceDescriptor> deviceDesc;
- if (config->role == AUDIO_PORT_ROLE_SOURCE) {
- deviceDesc = mAvailableInputDevices.getDeviceFromId(config->id);
- } else if (config->role == AUDIO_PORT_ROLE_SINK) {
- deviceDesc = mAvailableOutputDevices.getDeviceFromId(config->id);
- } else {
- return BAD_VALUE;
- }
- if (deviceDesc == NULL) {
- return BAD_VALUE;
- }
- audioPortConfig = deviceDesc;
- } else {
- return BAD_VALUE;
- }
-
- struct audio_port_config backupConfig;
- status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig);
- if (status == NO_ERROR) {
- struct audio_port_config newConfig;
- audioPortConfig->toAudioPortConfig(&newConfig, config);
- status = mpClientInterface->setAudioPortConfig(&newConfig, 0);
- }
- if (status != NO_ERROR) {
- audioPortConfig->applyAudioPortConfig(&backupConfig);
- }
-
- return status;
-}
-
-void AudioPolicyManager::clearAudioPatches(uid_t uid)
-{
- for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) {
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i);
- if (patchDesc->mUid == uid) {
- releaseAudioPatch(mAudioPatches.keyAt(i), uid);
- }
- }
-}
-
-status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session,
- audio_io_handle_t *ioHandle,
- audio_devices_t *device)
-{
- *session = (audio_session_t)mpClientInterface->newAudioUniqueId();
- *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId();
- *device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD);
-
- mSoundTriggerSessions.add(*session, *ioHandle);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::releaseSoundTriggerSession(audio_session_t session)
-{
- ssize_t index = mSoundTriggerSessions.indexOfKey(session);
- if (index < 0) {
- ALOGW("acquireSoundTriggerSession() session %d not registered", session);
- return BAD_VALUE;
- }
-
- mSoundTriggerSessions.removeItem(session);
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::addAudioPatch(audio_patch_handle_t handle,
- const sp<AudioPatch>& patch)
-{
- ssize_t index = mAudioPatches.indexOfKey(handle);
-
- if (index >= 0) {
- ALOGW("addAudioPatch() patch %d already in", handle);
- return ALREADY_EXISTS;
- }
- mAudioPatches.add(handle, patch);
- ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d"
- "sink handle %d",
- handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks,
- patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id);
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::removeAudioPatch(audio_patch_handle_t handle)
-{
- ssize_t index = mAudioPatches.indexOfKey(handle);
-
- if (index < 0) {
- ALOGW("removeAudioPatch() patch %d not in", handle);
- return ALREADY_EXISTS;
- }
- ALOGV("removeAudioPatch() handle %d af handle %d", handle,
- mAudioPatches.valueAt(index)->mAfPatchHandle);
- mAudioPatches.removeItemsAt(index);
- return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManager
-// ----------------------------------------------------------------------------
-
-uint32_t AudioPolicyManager::nextUniqueId()
-{
- return android_atomic_inc(&mNextUniqueId);
-}
-
-uint32_t AudioPolicyManager::nextAudioPortGeneration()
-{
- return android_atomic_inc(&mAudioPortGeneration);
-}
-
-AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
- :
-#ifdef AUDIO_POLICY_TEST
- Thread(false),
-#endif //AUDIO_POLICY_TEST
- mPrimaryOutput((audio_io_handle_t)0),
- mPhoneState(AUDIO_MODE_NORMAL),
- mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
- mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
- mA2dpSuspended(false),
- mSpeakerDrcEnabled(false), mNextUniqueId(1),
- mAudioPortGeneration(1),
- mBeaconMuteRefCount(0),
- mBeaconPlayingRefCount(0),
- mBeaconMuted(false)
-{
- mUidCached = getuid();
- mpClientInterface = clientInterface;
-
- for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) {
- mForceUse[i] = AUDIO_POLICY_FORCE_NONE;
- }
-
- mDefaultOutputDevice = new DeviceDescriptor(String8(""), AUDIO_DEVICE_OUT_SPEAKER);
- if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
- if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
- ALOGE("could not load audio policy configuration file, setting defaults");
- defaultAudioPolicyConfig();
- }
- }
- // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
-
- // must be done after reading the policy
- initializeVolumeCurves();
-
- // open all output streams needed to access attached devices
- audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
- audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
- for (size_t i = 0; i < mHwModules.size(); i++) {
- mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
- if (mHwModules[i]->mHandle == 0) {
- ALOGW("could not open HW module %s", mHwModules[i]->mName);
- continue;
- }
- // open all output streams needed to access attached devices
- // except for direct output streams that are only opened when they are actually
- // required by an app.
- // This also validates mAvailableOutputDevices list
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
- {
- const sp<IOProfile> outProfile = mHwModules[i]->mOutputProfiles[j];
-
- if (outProfile->mSupportedDevices.isEmpty()) {
- ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName);
- continue;
- }
-
- if ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
- continue;
- }
- audio_devices_t profileType = outProfile->mSupportedDevices.types();
- if ((profileType & mDefaultOutputDevice->mDeviceType) != AUDIO_DEVICE_NONE) {
- profileType = mDefaultOutputDevice->mDeviceType;
- } else {
- // chose first device present in mSupportedDevices also part of
- // outputDeviceTypes
- for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
- profileType = outProfile->mSupportedDevices[k]->mDeviceType;
- if ((profileType & outputDeviceTypes) != 0) {
- break;
- }
- }
- }
- if ((profileType & outputDeviceTypes) == 0) {
- continue;
- }
- sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(outProfile);
-
- outputDesc->mDevice = profileType;
- audio_config_t config = AUDIO_CONFIG_INITIALIZER;
- config.sample_rate = outputDesc->mSamplingRate;
- config.channel_mask = outputDesc->mChannelMask;
- config.format = outputDesc->mFormat;
- audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle,
- &output,
- &config,
- &outputDesc->mDevice,
- String8(""),
- &outputDesc->mLatency,
- outputDesc->mFlags);
-
- if (status != NO_ERROR) {
- ALOGW("Cannot open output stream for device %08x on hw module %s",
- outputDesc->mDevice,
- mHwModules[i]->mName);
- } else {
- outputDesc->mSamplingRate = config.sample_rate;
- outputDesc->mChannelMask = config.channel_mask;
- outputDesc->mFormat = config.format;
-
- for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
- audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType;
- ssize_t index =
- mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]);
- // give a valid ID to an attached device once confirmed it is reachable
- if ((index >= 0) && (mAvailableOutputDevices[index]->mId == 0)) {
- mAvailableOutputDevices[index]->mId = nextUniqueId();
- mAvailableOutputDevices[index]->mModule = mHwModules[i];
- }
- }
- if (mPrimaryOutput == 0 &&
- outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
- mPrimaryOutput = output;
- }
- addOutput(output, outputDesc);
- setOutputDevice(output,
- outputDesc->mDevice,
- true);
- }
- }
- // open input streams needed to access attached devices to validate
- // mAvailableInputDevices list
- for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
- {
- const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
-
- if (inProfile->mSupportedDevices.isEmpty()) {
- ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName);
- continue;
- }
- // chose first device present in mSupportedDevices also part of
- // inputDeviceTypes
- audio_devices_t profileType = AUDIO_DEVICE_NONE;
- for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
- profileType = inProfile->mSupportedDevices[k]->mDeviceType;
- if (profileType & inputDeviceTypes) {
- break;
- }
- }
- if ((profileType & inputDeviceTypes) == 0) {
- continue;
- }
- sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(inProfile);
-
- inputDesc->mInputSource = AUDIO_SOURCE_MIC;
- inputDesc->mDevice = profileType;
-
- // find the address
- DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType);
- // the inputs vector must be of size 1, but we don't want to crash here
- String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress
- : String8("");
- ALOGV(" for input device 0x%x using address %s", profileType, address.string());
- ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!");
-
- audio_config_t config = AUDIO_CONFIG_INITIALIZER;
- config.sample_rate = inputDesc->mSamplingRate;
- config.channel_mask = inputDesc->mChannelMask;
- config.format = inputDesc->mFormat;
- audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
- status_t status = mpClientInterface->openInput(inProfile->mModule->mHandle,
- &input,
- &config,
- &inputDesc->mDevice,
- address,
- AUDIO_SOURCE_MIC,
- AUDIO_INPUT_FLAG_NONE);
-
- if (status == NO_ERROR) {
- for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
- audio_devices_t type = inProfile->mSupportedDevices[k]->mDeviceType;
- ssize_t index =
- mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]);
- // give a valid ID to an attached device once confirmed it is reachable
- if ((index >= 0) && (mAvailableInputDevices[index]->mId == 0)) {
- mAvailableInputDevices[index]->mId = nextUniqueId();
- mAvailableInputDevices[index]->mModule = mHwModules[i];
- }
- }
- mpClientInterface->closeInput(input);
- } else {
- ALOGW("Cannot open input stream for device %08x on hw module %s",
- inputDesc->mDevice,
- mHwModules[i]->mName);
- }
- }
- }
- // make sure all attached devices have been allocated a unique ID
- for (size_t i = 0; i < mAvailableOutputDevices.size();) {
- if (mAvailableOutputDevices[i]->mId == 0) {
- ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType);
- mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
- continue;
- }
- i++;
- }
- for (size_t i = 0; i < mAvailableInputDevices.size();) {
- if (mAvailableInputDevices[i]->mId == 0) {
- ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType);
- mAvailableInputDevices.remove(mAvailableInputDevices[i]);
- continue;
- }
- i++;
- }
- // make sure default device is reachable
- if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
- ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mDeviceType);
- }
-
- ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
-
- updateDevicesAndOutputs();
-
-#ifdef AUDIO_POLICY_TEST
- if (mPrimaryOutput != 0) {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"), 0);
- mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
-
- mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
- mTestSamplingRate = 44100;
- mTestFormat = AUDIO_FORMAT_PCM_16_BIT;
- mTestChannels = AUDIO_CHANNEL_OUT_STEREO;
- mTestLatencyMs = 0;
- mCurOutput = 0;
- mDirectOutput = false;
- for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
- mTestOutputs[i] = 0;
- }
-
- const size_t SIZE = 256;
- char buffer[SIZE];
- snprintf(buffer, SIZE, "AudioPolicyManagerTest");
- run(buffer, ANDROID_PRIORITY_AUDIO);
- }
-#endif //AUDIO_POLICY_TEST
-}
-
-AudioPolicyManager::~AudioPolicyManager()
-{
-#ifdef AUDIO_POLICY_TEST
- exit();
-#endif //AUDIO_POLICY_TEST
- for (size_t i = 0; i < mOutputs.size(); i++) {
- mpClientInterface->closeOutput(mOutputs.keyAt(i));
- }
- for (size_t i = 0; i < mInputs.size(); i++) {
- mpClientInterface->closeInput(mInputs.keyAt(i));
- }
- mAvailableOutputDevices.clear();
- mAvailableInputDevices.clear();
- mOutputs.clear();
- mInputs.clear();
- mHwModules.clear();
-}
-
-status_t AudioPolicyManager::initCheck()
-{
- return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
-}
-
-#ifdef AUDIO_POLICY_TEST
-bool AudioPolicyManager::threadLoop()
-{
- ALOGV("entering threadLoop()");
- while (!exitPending())
- {
- String8 command;
- int valueInt;
- String8 value;
-
- Mutex::Autolock _l(mLock);
- mWaitWorkCV.waitRelative(mLock, milliseconds(50));
-
- command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
- AudioParameter param = AudioParameter(command);
-
- if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
- valueInt != 0) {
- ALOGV("Test command %s received", command.string());
- String8 target;
- if (param.get(String8("target"), target) != NO_ERROR) {
- target = "Manager";
- }
- if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_output"));
- mCurOutput = valueInt;
- }
- if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_direct"));
- if (value == "false") {
- mDirectOutput = false;
- } else if (value == "true") {
- mDirectOutput = true;
- }
- }
- if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_input"));
- mTestInput = valueInt;
- }
-
- if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_format"));
- int format = AUDIO_FORMAT_INVALID;
- if (value == "PCM 16 bits") {
- format = AUDIO_FORMAT_PCM_16_BIT;
- } else if (value == "PCM 8 bits") {
- format = AUDIO_FORMAT_PCM_8_BIT;
- } else if (value == "Compressed MP3") {
- format = AUDIO_FORMAT_MP3;
- }
- if (format != AUDIO_FORMAT_INVALID) {
- if (target == "Manager") {
- mTestFormat = format;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("format"), format);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
- if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_channels"));
- int channels = 0;
-
- if (value == "Channels Stereo") {
- channels = AUDIO_CHANNEL_OUT_STEREO;
- } else if (value == "Channels Mono") {
- channels = AUDIO_CHANNEL_OUT_MONO;
- }
- if (channels != 0) {
- if (target == "Manager") {
- mTestChannels = channels;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("channels"), channels);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
- if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_sampleRate"));
- if (valueInt >= 0 && valueInt <= 96000) {
- int samplingRate = valueInt;
- if (target == "Manager") {
- mTestSamplingRate = samplingRate;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("sampling_rate"), samplingRate);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
-
- if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_reopen"));
-
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
- mpClientInterface->closeOutput(mPrimaryOutput);
-
- audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
-
- mOutputs.removeItem(mPrimaryOutput);
-
- sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
- outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
- audio_config_t config = AUDIO_CONFIG_INITIALIZER;
- config.sample_rate = outputDesc->mSamplingRate;
- config.channel_mask = outputDesc->mChannelMask;
- config.format = outputDesc->mFormat;
- status_t status = mpClientInterface->openOutput(moduleHandle,
- &mPrimaryOutput,
- &config,
- &outputDesc->mDevice,
- String8(""),
- &outputDesc->mLatency,
- outputDesc->mFlags);
- if (status != NO_ERROR) {
- ALOGE("Failed to reopen hardware output stream, "
- "samplingRate: %d, format %d, channels %d",
- outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
- } else {
- outputDesc->mSamplingRate = config.sample_rate;
- outputDesc->mChannelMask = config.channel_mask;
- outputDesc->mFormat = config.format;
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"), 0);
- mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
- addOutput(mPrimaryOutput, outputDesc);
- }
- }
-
-
- mpClientInterface->setParameters(0, String8("test_cmd_policy="));
- }
- }
- return false;
-}
-
-void AudioPolicyManager::exit()
-{
- {
- AutoMutex _l(mLock);
- requestExit();
- mWaitWorkCV.signal();
- }
- requestExitAndWait();
-}
-
-int AudioPolicyManager::testOutputIndex(audio_io_handle_t output)
-{
- for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
- if (output == mTestOutputs[i]) return i;
- }
- return 0;
-}
-#endif //AUDIO_POLICY_TEST
-
-// ---
-
-void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc)
-{
- outputDesc->mIoHandle = output;
- outputDesc->mId = nextUniqueId();
- mOutputs.add(output, outputDesc);
- nextAudioPortGeneration();
-}
-
-void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc)
-{
- inputDesc->mIoHandle = input;
- inputDesc->mId = nextUniqueId();
- mInputs.add(input, inputDesc);
- nextAudioPortGeneration();
-}
-
-void AudioPolicyManager::findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
- const audio_devices_t device /*in*/,
- const String8 address /*in*/,
- SortedVector<audio_io_handle_t>& outputs /*out*/) {
- sp<DeviceDescriptor> devDesc =
- desc->mProfile->mSupportedDevices.getDevice(device, address);
- if (devDesc != 0) {
- ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s",
- desc->mIoHandle, address.string());
- outputs.add(desc->mIoHandle);
- }
-}
-
-status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
- audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& outputs,
- const String8 address)
-{
- audio_devices_t device = devDesc->mDeviceType;
- sp<AudioOutputDescriptor> desc;
- // erase all current sample rates, formats and channel masks
- devDesc->clearCapabilities();
-
- if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
- // first list already open outputs that can be routed to this device
- for (size_t i = 0; i < mOutputs.size(); i++) {
- desc = mOutputs.valueAt(i);
- if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices.types() & device)) {
- if (!deviceDistinguishesOnAddress(device)) {
- ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
- outputs.add(mOutputs.keyAt(i));
- } else {
- ALOGV(" checking address match due to device 0x%x", device);
- findIoHandlesByAddress(desc, device, address, outputs);
- }
- }
- }
- // then look for output profiles that can be routed to this device
- SortedVector< sp<IOProfile> > profiles;
- for (size_t i = 0; i < mHwModules.size(); i++)
- {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
- {
- sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
- if (profile->mSupportedDevices.types() & device) {
- if (!deviceDistinguishesOnAddress(device) ||
- address == profile->mSupportedDevices[0]->mAddress) {
- profiles.add(profile);
- ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
- }
- }
- }
- }
-
- ALOGV(" found %d profiles, %d outputs", profiles.size(), outputs.size());
-
- if (profiles.isEmpty() && outputs.isEmpty()) {
- ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
- return BAD_VALUE;
- }
-
- // open outputs for matching profiles if needed. Direct outputs are also opened to
- // query for dynamic parameters and will be closed later by setDeviceConnectionState()
- for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
- sp<IOProfile> profile = profiles[profile_index];
-
- // nothing to do if one output is already opened for this profile
- size_t j;
- for (j = 0; j < outputs.size(); j++) {
- desc = mOutputs.valueFor(outputs.itemAt(j));
- if (!desc->isDuplicated() && desc->mProfile == profile) {
- // matching profile: save the sample rates, format and channel masks supported
- // by the profile in our device descriptor
- devDesc->importAudioPort(profile);
- break;
- }
- }
- if (j != outputs.size()) {
- continue;
- }
-
- ALOGV("opening output for device %08x with params %s profile %p",
- device, address.string(), profile.get());
- desc = new AudioOutputDescriptor(profile);
- desc->mDevice = device;
- audio_config_t config = AUDIO_CONFIG_INITIALIZER;
- config.sample_rate = desc->mSamplingRate;
- config.channel_mask = desc->mChannelMask;
- config.format = desc->mFormat;
- config.offload_info.sample_rate = desc->mSamplingRate;
- config.offload_info.channel_mask = desc->mChannelMask;
- config.offload_info.format = desc->mFormat;
- audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- status_t status = mpClientInterface->openOutput(profile->mModule->mHandle,
- &output,
- &config,
- &desc->mDevice,
- address,
- &desc->mLatency,
- desc->mFlags);
- if (status == NO_ERROR) {
- desc->mSamplingRate = config.sample_rate;
- desc->mChannelMask = config.channel_mask;
- desc->mFormat = config.format;
-
- // Here is where the out_set_parameters() for card & device gets called
- if (!address.isEmpty()) {
- char *param = audio_device_address_to_parameter(device, address);
- mpClientInterface->setParameters(output, String8(param));
- free(param);
- }
-
- // Here is where we step through and resolve any "dynamic" fields
- String8 reply;
- char *value;
- if (profile->mSamplingRates[0] == 0) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
- ALOGV("checkOutputsForDevice() supported sampling rates %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadSamplingRates(value + 1);
- }
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
- ALOGV("checkOutputsForDevice() supported formats %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadFormats(value + 1);
- }
- }
- if (profile->mChannelMasks[0] == 0) {
- reply = mpClientInterface->getParameters(output,
- String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
- ALOGV("checkOutputsForDevice() supported channel masks %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadOutChannels(value + 1);
- }
- }
- if (((profile->mSamplingRates[0] == 0) &&
- (profile->mSamplingRates.size() < 2)) ||
- ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
- (profile->mFormats.size() < 2)) ||
- ((profile->mChannelMasks[0] == 0) &&
- (profile->mChannelMasks.size() < 2))) {
- ALOGW("checkOutputsForDevice() missing param");
- mpClientInterface->closeOutput(output);
- output = AUDIO_IO_HANDLE_NONE;
- } else if (profile->mSamplingRates[0] == 0 || profile->mFormats[0] == 0 ||
- profile->mChannelMasks[0] == 0) {
- mpClientInterface->closeOutput(output);
- config.sample_rate = profile->pickSamplingRate();
- config.channel_mask = profile->pickChannelMask();
- config.format = profile->pickFormat();
- config.offload_info.sample_rate = config.sample_rate;
- config.offload_info.channel_mask = config.channel_mask;
- config.offload_info.format = config.format;
- status = mpClientInterface->openOutput(profile->mModule->mHandle,
- &output,
- &config,
- &desc->mDevice,
- address,
- &desc->mLatency,
- desc->mFlags);
- if (status == NO_ERROR) {
- desc->mSamplingRate = config.sample_rate;
- desc->mChannelMask = config.channel_mask;
- desc->mFormat = config.format;
- } else {
- output = AUDIO_IO_HANDLE_NONE;
- }
- }
-
- if (output != AUDIO_IO_HANDLE_NONE) {
- addOutput(output, desc);
- if (deviceDistinguishesOnAddress(device) && address != "0") {
- ssize_t index = mPolicyMixes.indexOfKey(address);
- if (index >= 0) {
- mPolicyMixes[index]->mOutput = desc;
- desc->mPolicyMix = &mPolicyMixes[index]->mMix;
- } else {
- ALOGE("checkOutputsForDevice() cannot find policy for address %s",
- address.string());
- }
- } else if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) {
- // no duplicated output for direct outputs and
- // outputs used by dynamic policy mixes
- audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE;
-
- // set initial stream volume for device
- applyStreamVolumes(output, device, 0, true);
-
- //TODO: configure audio effect output stage here
-
- // open a duplicating output thread for the new output and the primary output
- duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
- mPrimaryOutput);
- if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) {
- // add duplicated output descriptor
- sp<AudioOutputDescriptor> dupOutputDesc =
- new AudioOutputDescriptor(NULL);
- dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
- dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
- dupOutputDesc->mSamplingRate = desc->mSamplingRate;
- dupOutputDesc->mFormat = desc->mFormat;
- dupOutputDesc->mChannelMask = desc->mChannelMask;
- dupOutputDesc->mLatency = desc->mLatency;
- addOutput(duplicatedOutput, dupOutputDesc);
- applyStreamVolumes(duplicatedOutput, device, 0, true);
- } else {
- ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
- mPrimaryOutput, output);
- mpClientInterface->closeOutput(output);
- mOutputs.removeItem(output);
- nextAudioPortGeneration();
- output = AUDIO_IO_HANDLE_NONE;
- }
- }
- }
- } else {
- output = AUDIO_IO_HANDLE_NONE;
- }
- if (output == AUDIO_IO_HANDLE_NONE) {
- ALOGW("checkOutputsForDevice() could not open output for device %x", device);
- profiles.removeAt(profile_index);
- profile_index--;
- } else {
- outputs.add(output);
- devDesc->importAudioPort(profile);
-
- if (deviceDistinguishesOnAddress(device)) {
- ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)",
- device, address.string());
- setOutputDevice(output, device, true/*force*/, 0/*delay*/,
- NULL/*patch handle*/, address.string());
- }
- ALOGV("checkOutputsForDevice(): adding output %d", output);
- }
- }
-
- if (profiles.isEmpty()) {
- ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
- return BAD_VALUE;
- }
- } else { // Disconnect
- // check if one opened output is not needed any more after disconnecting one device
- for (size_t i = 0; i < mOutputs.size(); i++) {
- desc = mOutputs.valueAt(i);
- if (!desc->isDuplicated()) {
- // exact match on device
- if (deviceDistinguishesOnAddress(device) &&
- (desc->mProfile->mSupportedDevices.types() == device)) {
- findIoHandlesByAddress(desc, device, address, outputs);
- } else if (!(desc->mProfile->mSupportedDevices.types()
- & mAvailableOutputDevices.types())) {
- ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
- mOutputs.keyAt(i));
- outputs.add(mOutputs.keyAt(i));
- }
- }
- }
- // Clear any profiles associated with the disconnected device.
- for (size_t i = 0; i < mHwModules.size(); i++)
- {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
- {
- sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
- if (profile->mSupportedDevices.types() & device) {
- ALOGV("checkOutputsForDevice(): "
- "clearing direct output profile %zu on module %zu", j, i);
- if (profile->mSamplingRates[0] == 0) {
- profile->mSamplingRates.clear();
- profile->mSamplingRates.add(0);
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- profile->mFormats.clear();
- profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
- }
- if (profile->mChannelMasks[0] == 0) {
- profile->mChannelMasks.clear();
- profile->mChannelMasks.add(0);
- }
- }
- }
- }
- }
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device,
- audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& inputs,
- const String8 address)
-{
- sp<AudioInputDescriptor> desc;
- if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
- // first list already open inputs that can be routed to this device
- for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
- desc = mInputs.valueAt(input_index);
- if (desc->mProfile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) {
- ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index));
- inputs.add(mInputs.keyAt(input_index));
- }
- }
-
- // then look for input profiles that can be routed to this device
- SortedVector< sp<IOProfile> > profiles;
- for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++)
- {
- if (mHwModules[module_idx]->mHandle == 0) {
- continue;
- }
- for (size_t profile_index = 0;
- profile_index < mHwModules[module_idx]->mInputProfiles.size();
- profile_index++)
- {
- sp<IOProfile> profile = mHwModules[module_idx]->mInputProfiles[profile_index];
-
- if (profile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) {
- if (!deviceDistinguishesOnAddress(device) ||
- address == profile->mSupportedDevices[0]->mAddress) {
- profiles.add(profile);
- ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
- profile_index, module_idx);
- }
- }
- }
- }
-
- if (profiles.isEmpty() && inputs.isEmpty()) {
- ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
- return BAD_VALUE;
- }
-
- // open inputs for matching profiles if needed. Direct inputs are also opened to
- // query for dynamic parameters and will be closed later by setDeviceConnectionState()
- for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
-
- sp<IOProfile> profile = profiles[profile_index];
- // nothing to do if one input is already opened for this profile
- size_t input_index;
- for (input_index = 0; input_index < mInputs.size(); input_index++) {
- desc = mInputs.valueAt(input_index);
- if (desc->mProfile == profile) {
- break;
- }
- }
- if (input_index != mInputs.size()) {
- continue;
- }
-
- ALOGV("opening input for device 0x%X with params %s", device, address.string());
- desc = new AudioInputDescriptor(profile);
- desc->mDevice = device;
- audio_config_t config = AUDIO_CONFIG_INITIALIZER;
- config.sample_rate = desc->mSamplingRate;
- config.channel_mask = desc->mChannelMask;
- config.format = desc->mFormat;
- audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
- status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
- &input,
- &config,
- &desc->mDevice,
- address,
- AUDIO_SOURCE_MIC,
- AUDIO_INPUT_FLAG_NONE /*FIXME*/);
-
- if (status == NO_ERROR) {
- desc->mSamplingRate = config.sample_rate;
- desc->mChannelMask = config.channel_mask;
- desc->mFormat = config.format;
-
- if (!address.isEmpty()) {
- char *param = audio_device_address_to_parameter(device, address);
- mpClientInterface->setParameters(input, String8(param));
- free(param);
- }
-
- // Here is where we step through and resolve any "dynamic" fields
- String8 reply;
- char *value;
- if (profile->mSamplingRates[0] == 0) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
- ALOGV("checkInputsForDevice() direct input sup sampling rates %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadSamplingRates(value + 1);
- }
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
- ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadFormats(value + 1);
- }
- }
- if (profile->mChannelMasks[0] == 0) {
- reply = mpClientInterface->getParameters(input,
- String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
- ALOGV("checkInputsForDevice() direct input sup channel masks %s",
- reply.string());
- value = strpbrk((char *)reply.string(), "=");
- if (value != NULL) {
- profile->loadInChannels(value + 1);
- }
- }
- if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) ||
- ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) ||
- ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) {
- ALOGW("checkInputsForDevice() direct input missing param");
- mpClientInterface->closeInput(input);
- input = AUDIO_IO_HANDLE_NONE;
- }
-
- if (input != 0) {
- addInput(input, desc);
- }
- } // endif input != 0
-
- if (input == AUDIO_IO_HANDLE_NONE) {
- ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
- profiles.removeAt(profile_index);
- profile_index--;
- } else {
- inputs.add(input);
- ALOGV("checkInputsForDevice(): adding input %d", input);
- }
- } // end scan profiles
-
- if (profiles.isEmpty()) {
- ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
- return BAD_VALUE;
- }
- } else {
- // Disconnect
- // check if one opened input is not needed any more after disconnecting one device
- for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
- desc = mInputs.valueAt(input_index);
- if (!(desc->mProfile->mSupportedDevices.types() & mAvailableInputDevices.types() &
- ~AUDIO_DEVICE_BIT_IN)) {
- ALOGV("checkInputsForDevice(): disconnecting adding input %d",
- mInputs.keyAt(input_index));
- inputs.add(mInputs.keyAt(input_index));
- }
- }
- // Clear any profiles associated with the disconnected device.
- for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) {
- if (mHwModules[module_index]->mHandle == 0) {
- continue;
- }
- for (size_t profile_index = 0;
- profile_index < mHwModules[module_index]->mInputProfiles.size();
- profile_index++) {
- sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index];
- if (profile->mSupportedDevices.types() & device & ~AUDIO_DEVICE_BIT_IN) {
- ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu",
- profile_index, module_index);
- if (profile->mSamplingRates[0] == 0) {
- profile->mSamplingRates.clear();
- profile->mSamplingRates.add(0);
- }
- if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
- profile->mFormats.clear();
- profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
- }
- if (profile->mChannelMasks[0] == 0) {
- profile->mChannelMasks.clear();
- profile->mChannelMasks.add(0);
- }
- }
- }
- }
- } // end disconnect
-
- return NO_ERROR;
-}
-
-
-void AudioPolicyManager::closeOutput(audio_io_handle_t output)
-{
- ALOGV("closeOutput(%d)", output);
-
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- if (outputDesc == NULL) {
- ALOGW("closeOutput() unknown output %d", output);
- return;
- }
-
- for (size_t i = 0; i < mPolicyMixes.size(); i++) {
- if (mPolicyMixes[i]->mOutput == outputDesc) {
- mPolicyMixes[i]->mOutput.clear();
- }
- }
-
- // look for duplicated outputs connected to the output being removed.
- for (size_t i = 0; i < mOutputs.size(); i++) {
- sp<AudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i);
- if (dupOutputDesc->isDuplicated() &&
- (dupOutputDesc->mOutput1 == outputDesc ||
- dupOutputDesc->mOutput2 == outputDesc)) {
- sp<AudioOutputDescriptor> outputDesc2;
- if (dupOutputDesc->mOutput1 == outputDesc) {
- outputDesc2 = dupOutputDesc->mOutput2;
- } else {
- outputDesc2 = dupOutputDesc->mOutput1;
- }
- // As all active tracks on duplicated output will be deleted,
- // and as they were also referenced on the other output, the reference
- // count for their stream type must be adjusted accordingly on
- // the other output.
- for (int j = 0; j < AUDIO_STREAM_CNT; j++) {
- int refCount = dupOutputDesc->mRefCount[j];
- outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount);
- }
- audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
- ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
-
- mpClientInterface->closeOutput(duplicatedOutput);
- mOutputs.removeItem(duplicatedOutput);
- }
- }
-
- nextAudioPortGeneration();
-
- ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
- if (index >= 0) {
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
- mAudioPatches.removeItemsAt(index);
- mpClientInterface->onAudioPatchListUpdate();
- }
-
- AudioParameter param;
- param.add(String8("closing"), String8("true"));
- mpClientInterface->setParameters(output, param.toString());
-
- mpClientInterface->closeOutput(output);
- mOutputs.removeItem(output);
- mPreviousOutputs = mOutputs;
-}
-
-void AudioPolicyManager::closeInput(audio_io_handle_t input)
-{
- ALOGV("closeInput(%d)", input);
-
- sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
- if (inputDesc == NULL) {
- ALOGW("closeInput() unknown input %d", input);
- return;
- }
-
- nextAudioPortGeneration();
-
- ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
- if (index >= 0) {
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
- mAudioPatches.removeItemsAt(index);
- mpClientInterface->onAudioPatchListUpdate();
- }
-
- mpClientInterface->closeInput(input);
- mInputs.removeItem(input);
-}
-
-SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device,
- DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs)
-{
- SortedVector<audio_io_handle_t> outputs;
-
- ALOGVV("getOutputsForDevice() device %04x", device);
- for (size_t i = 0; i < openOutputs.size(); i++) {
- ALOGVV("output %d isDuplicated=%d device=%04x",
- i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
- if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
- ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
- outputs.add(openOutputs.keyAt(i));
- }
- }
- return outputs;
-}
-
-bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
- SortedVector<audio_io_handle_t>& outputs2)
-{
- if (outputs1.size() != outputs2.size()) {
- return false;
- }
- for (size_t i = 0; i < outputs1.size(); i++) {
- if (outputs1[i] != outputs2[i]) {
- return false;
- }
- }
- return true;
-}
-
-void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
-{
- audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
- audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
- SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
- SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
-
- // also take into account external policy-related changes: add all outputs which are
- // associated with policies in the "before" and "after" output vectors
- ALOGVV("checkOutputForStrategy(): policy related outputs");
- for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
- const sp<AudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
- if (desc != 0 && desc->mPolicyMix != NULL) {
- srcOutputs.add(desc->mIoHandle);
- ALOGVV(" previous outputs: adding %d", desc->mIoHandle);
- }
- }
- for (size_t i = 0 ; i < mOutputs.size() ; i++) {
- const sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
- if (desc != 0 && desc->mPolicyMix != NULL) {
- dstOutputs.add(desc->mIoHandle);
- ALOGVV(" new outputs: adding %d", desc->mIoHandle);
- }
- }
-
- if (!vectorsEqual(srcOutputs,dstOutputs)) {
- ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
- strategy, srcOutputs[0], dstOutputs[0]);
- // mute strategy while moving tracks from one output to another
- for (size_t i = 0; i < srcOutputs.size(); i++) {
- sp<AudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]);
- if (desc->isStrategyActive(strategy)) {
- setStrategyMute(strategy, true, srcOutputs[i]);
- setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
- }
- }
-
- // Move effects associated to this strategy from previous output to new output
- if (strategy == STRATEGY_MEDIA) {
- audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
- SortedVector<audio_io_handle_t> moved;
- for (size_t i = 0; i < mEffects.size(); i++) {
- sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
- if (effectDesc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
- effectDesc->mIo != fxOutput) {
- if (moved.indexOf(effectDesc->mIo) < 0) {
- ALOGV("checkOutputForStrategy() moving effect %d to output %d",
- mEffects.keyAt(i), fxOutput);
- mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo,
- fxOutput);
- moved.add(effectDesc->mIo);
- }
- effectDesc->mIo = fxOutput;
- }
- }
- }
- // Move tracks associated to this strategy from previous output to new output
- for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- if (i == AUDIO_STREAM_PATCH) {
- continue;
- }
- if (getStrategy((audio_stream_type_t)i) == strategy) {
- mpClientInterface->invalidateStream((audio_stream_type_t)i);
- }
- }
- }
-}
-
-void AudioPolicyManager::checkOutputForAllStrategies()
-{
- if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
- checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
- checkOutputForStrategy(STRATEGY_PHONE);
- if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
- checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
- checkOutputForStrategy(STRATEGY_SONIFICATION);
- checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
- checkOutputForStrategy(STRATEGY_ACCESSIBILITY);
- checkOutputForStrategy(STRATEGY_MEDIA);
- checkOutputForStrategy(STRATEGY_DTMF);
- checkOutputForStrategy(STRATEGY_REROUTING);
-}
-
-audio_io_handle_t AudioPolicyManager::getA2dpOutput()
-{
- for (size_t i = 0; i < mOutputs.size(); i++) {
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
- if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
- return mOutputs.keyAt(i);
- }
- }
-
- return 0;
-}
-
-void AudioPolicyManager::checkA2dpSuspend()
-{
- audio_io_handle_t a2dpOutput = getA2dpOutput();
- if (a2dpOutput == 0) {
- mA2dpSuspended = false;
- return;
- }
-
- bool isScoConnected =
- ((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET &
- ~AUDIO_DEVICE_BIT_IN) != 0) ||
- ((mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_ALL_SCO) != 0);
- // suspend A2DP output if:
- // (NOT already suspended) &&
- // ((SCO device is connected &&
- // (forced usage for communication || for record is SCO))) ||
- // (phone state is ringing || in call)
- //
- // restore A2DP output if:
- // (Already suspended) &&
- // ((SCO device is NOT connected ||
- // (forced usage NOT for communication && NOT for record is SCO))) &&
- // (phone state is NOT ringing && NOT in call)
- //
- if (mA2dpSuspended) {
- if ((!isScoConnected ||
- ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO) &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] != AUDIO_POLICY_FORCE_BT_SCO))) &&
- ((mPhoneState != AUDIO_MODE_IN_CALL) &&
- (mPhoneState != AUDIO_MODE_RINGTONE))) {
-
- mpClientInterface->restoreOutput(a2dpOutput);
- mA2dpSuspended = false;
- }
- } else {
- if ((isScoConnected &&
- ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
- (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO))) ||
- ((mPhoneState == AUDIO_MODE_IN_CALL) ||
- (mPhoneState == AUDIO_MODE_RINGTONE))) {
-
- mpClientInterface->suspendOutput(a2dpOutput);
- mA2dpSuspended = true;
- }
- }
-}
-
-audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache)
-{
- audio_devices_t device = AUDIO_DEVICE_NONE;
-
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
-
- ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
- if (index >= 0) {
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- if (patchDesc->mUid != mUidCached) {
- ALOGV("getNewOutputDevice() device %08x forced by patch %d",
- outputDesc->device(), outputDesc->mPatchHandle);
- return outputDesc->device();
- }
- }
-
- // check the following by order of priority to request a routing change if necessary:
- // 1: the strategy enforced audible is active and enforced on the output:
- // use device for strategy enforced audible
- // 2: we are in call or the strategy phone is active on the output:
- // use device for strategy phone
- // 3: the strategy for enforced audible is active but not enforced on the output:
- // use the device for strategy enforced audible
- // 4: the strategy sonification is active on the output:
- // use device for strategy sonification
- // 5: the strategy "respectful" sonification is active on the output:
- // use device for strategy "respectful" sonification
- // 6: the strategy accessibility is active on the output:
- // use device for strategy accessibility
- // 7: the strategy media is active on the output:
- // use device for strategy media
- // 8: the strategy DTMF is active on the output:
- // use device for strategy DTMF
- // 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output:
- // use device for strategy t-t-s
- if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE) &&
- mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
- device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
- } else if (isInCall() ||
- outputDesc->isStrategyActive(STRATEGY_PHONE)) {
- device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
- device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_ACCESSIBILITY)) {
- device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
- device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
- device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) {
- device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
- } else if (outputDesc->isStrategyActive(STRATEGY_REROUTING)) {
- device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache);
- }
-
- ALOGV("getNewOutputDevice() selected device %x", device);
- return device;
-}
-
-audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input)
-{
- sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
-
- ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
- if (index >= 0) {
- sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- if (patchDesc->mUid != mUidCached) {
- ALOGV("getNewInputDevice() device %08x forced by patch %d",
- inputDesc->mDevice, inputDesc->mPatchHandle);
- return inputDesc->mDevice;
- }
- }
-
- audio_devices_t device = getDeviceAndMixForInputSource(inputDesc->mInputSource);
-
- ALOGV("getNewInputDevice() selected device %x", device);
- return device;
-}
-
-uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) {
- return (uint32_t)getStrategy(stream);
-}
-
-audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
- // By checking the range of stream before calling getStrategy, we avoid
- // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE
- // and then return STRATEGY_MEDIA, but we want to return the empty set.
- if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) {
- return AUDIO_DEVICE_NONE;
- }
- audio_devices_t devices;
- AudioPolicyManager::routing_strategy strategy = getStrategy(stream);
- devices = getDeviceForStrategy(strategy, true /*fromCache*/);
- SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(devices, mOutputs);
- for (size_t i = 0; i < outputs.size(); i++) {
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
- if (outputDesc->isStrategyActive(strategy)) {
- devices = outputDesc->device();
- break;
- }
- }
-
- /*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it
- and doesn't really need to.*/
- if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
- devices |= AUDIO_DEVICE_OUT_SPEAKER;
- devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE;
- }
-
- return devices;
-}
-
-AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(
- audio_stream_type_t stream) {
-
- ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH");
-
- // stream to strategy mapping
- switch (stream) {
- case AUDIO_STREAM_VOICE_CALL:
- case AUDIO_STREAM_BLUETOOTH_SCO:
- return STRATEGY_PHONE;
- case AUDIO_STREAM_RING:
- case AUDIO_STREAM_ALARM:
- return STRATEGY_SONIFICATION;
- case AUDIO_STREAM_NOTIFICATION:
- return STRATEGY_SONIFICATION_RESPECTFUL;
- case AUDIO_STREAM_DTMF:
- return STRATEGY_DTMF;
- default:
- ALOGE("unknown stream type %d", stream);
- case AUDIO_STREAM_SYSTEM:
- // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
- // while key clicks are played produces a poor result
- case AUDIO_STREAM_MUSIC:
- return STRATEGY_MEDIA;
- case AUDIO_STREAM_ENFORCED_AUDIBLE:
- return STRATEGY_ENFORCED_AUDIBLE;
- case AUDIO_STREAM_TTS:
- return STRATEGY_TRANSMITTED_THROUGH_SPEAKER;
- case AUDIO_STREAM_ACCESSIBILITY:
- return STRATEGY_ACCESSIBILITY;
- case AUDIO_STREAM_REROUTING:
- return STRATEGY_REROUTING;
- }
-}
-
-uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) {
- // flags to strategy mapping
- if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) {
- return (uint32_t) STRATEGY_TRANSMITTED_THROUGH_SPEAKER;
- }
- if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
- return (uint32_t) STRATEGY_ENFORCED_AUDIBLE;
- }
-
- // usage to strategy mapping
- switch (attr->usage) {
- case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
- if (isStreamActive(AUDIO_STREAM_RING) || isStreamActive(AUDIO_STREAM_ALARM)) {
- return (uint32_t) STRATEGY_SONIFICATION;
- }
- if (isInCall()) {
- return (uint32_t) STRATEGY_PHONE;
- }
- return (uint32_t) STRATEGY_ACCESSIBILITY;
-
- case AUDIO_USAGE_MEDIA:
- case AUDIO_USAGE_GAME:
- case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
- case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
- return (uint32_t) STRATEGY_MEDIA;
-
- case AUDIO_USAGE_VOICE_COMMUNICATION:
- return (uint32_t) STRATEGY_PHONE;
-
- case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
- return (uint32_t) STRATEGY_DTMF;
-
- case AUDIO_USAGE_ALARM:
- case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
- return (uint32_t) STRATEGY_SONIFICATION;
-
- case AUDIO_USAGE_NOTIFICATION:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
- case AUDIO_USAGE_NOTIFICATION_EVENT:
- return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL;
-
- case AUDIO_USAGE_UNKNOWN:
- default:
- return (uint32_t) STRATEGY_MEDIA;
- }
-}
-
-void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
- switch(stream) {
- case AUDIO_STREAM_MUSIC:
- checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
- updateDevicesAndOutputs();
- break;
- default:
- break;
- }
-}
-
-bool AudioPolicyManager::isAnyOutputActive(audio_stream_type_t streamToIgnore) {
- for (size_t s = 0 ; s < AUDIO_STREAM_CNT ; s++) {
- if (s == (size_t) streamToIgnore) {
- continue;
- }
- for (size_t i = 0; i < mOutputs.size(); i++) {
- const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
- if (outputDesc->mRefCount[s] != 0) {
- return true;
- }
- }
- }
- return false;
-}
-
-uint32_t AudioPolicyManager::handleEventForBeacon(int event) {
- switch(event) {
- case STARTING_OUTPUT:
- mBeaconMuteRefCount++;
- break;
- case STOPPING_OUTPUT:
- if (mBeaconMuteRefCount > 0) {
- mBeaconMuteRefCount--;
- }
- break;
- case STARTING_BEACON:
- mBeaconPlayingRefCount++;
- break;
- case STOPPING_BEACON:
- if (mBeaconPlayingRefCount > 0) {
- mBeaconPlayingRefCount--;
- }
- break;
- }
-
- if (mBeaconMuteRefCount > 0) {
- // any playback causes beacon to be muted
- return setBeaconMute(true);
- } else {
- // no other playback: unmute when beacon starts playing, mute when it stops
- return setBeaconMute(mBeaconPlayingRefCount == 0);
- }
-}
-
-uint32_t AudioPolicyManager::setBeaconMute(bool mute) {
- ALOGV("setBeaconMute(%d) mBeaconMuteRefCount=%d mBeaconPlayingRefCount=%d",
- mute, mBeaconMuteRefCount, mBeaconPlayingRefCount);
- // keep track of muted state to avoid repeating mute/unmute operations
- if (mBeaconMuted != mute) {
- // mute/unmute AUDIO_STREAM_TTS on all outputs
- ALOGV("\t muting %d", mute);
- uint32_t maxLatency = 0;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
- setStreamMute(AUDIO_STREAM_TTS, mute/*on*/,
- desc->mIoHandle,
- 0 /*delay*/, AUDIO_DEVICE_NONE);
- const uint32_t latency = desc->latency() * 2;
- if (latency > maxLatency) {
- maxLatency = latency;
- }
- }
- mBeaconMuted = mute;
- return maxLatency;
- }
- return 0;
-}
-
-audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
- bool fromCache)
-{
- uint32_t device = AUDIO_DEVICE_NONE;
-
- if (fromCache) {
- ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
- strategy, mDeviceForStrategy[strategy]);
- return mDeviceForStrategy[strategy];
- }
- audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
- switch (strategy) {
-
- case STRATEGY_TRANSMITTED_THROUGH_SPEAKER:
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
- if (!device) {
- ALOGE("getDeviceForStrategy() no device found for "\
- "STRATEGY_TRANSMITTED_THROUGH_SPEAKER");
- }
- break;
-
- case STRATEGY_SONIFICATION_RESPECTFUL:
- if (isInCall()) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
- } else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC,
- SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
- // while media is playing on a remote device, use the the sonification behavior.
- // Note that we test this usecase before testing if media is playing because
- // the isStreamActive() method only informs about the activity of a stream, not
- // if it's for local playback. Note also that we use the same delay between both tests
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
- //user "safe" speaker if available instead of normal speaker to avoid triggering
- //other acoustic safety mechanisms for notification
- if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
- device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
- } else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
- // while media is playing (or has recently played), use the same device
- device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
- } else {
- // when media is not playing anymore, fall back on the sonification behavior
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
- //user "safe" speaker if available instead of normal speaker to avoid triggering
- //other acoustic safety mechanisms for notification
- if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
- device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
- }
-
- break;
-
- case STRATEGY_DTMF:
- if (!isInCall()) {
- // when off call, DTMF strategy follows the same rules as MEDIA strategy
- device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
- break;
- }
- // when in call, DTMF and PHONE strategies follow the same rules
- // FALL THROUGH
-
- case STRATEGY_PHONE:
- // Force use of only devices on primary output if:
- // - in call AND
- // - cannot route from voice call RX OR
- // - audio HAL version is < 3.0 and TX device is on the primary HW module
- if (mPhoneState == AUDIO_MODE_IN_CALL) {
- audio_devices_t txDevice =
- getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
- sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
- if (((mAvailableInputDevices.types() &
- AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) ||
- (((txDevice & availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN) != 0) &&
- (hwOutputDesc->getAudioPort()->mModule->mHalVersion <
- AUDIO_DEVICE_API_VERSION_3_0))) {
- availableOutputDeviceTypes = availablePrimaryOutputDevices();
- }
- }
- // for phone strategy, we first consider the forced use and then the available devices by order
- // of priority
- switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
- case AUDIO_POLICY_FORCE_BT_SCO:
- if (!isInCall() || strategy != STRATEGY_DTMF) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
- if (device) break;
- }
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
- if (device) break;
- // if SCO device is requested but no SCO device is available, fall back to default case
- // FALL THROUGH
-
- default: // FORCE_NONE
- // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
- if (!isInCall() &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0)) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- if (device) break;
- }
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
- if (device) break;
- if (mPhoneState != AUDIO_MODE_IN_CALL) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
- }
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE;
- if (device) break;
- device = mDefaultOutputDevice->mDeviceType;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
- }
- break;
-
- case AUDIO_POLICY_FORCE_SPEAKER:
- // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
- // A2DP speaker when forcing to speaker output
- if (!isInCall() &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0)) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- if (device) break;
- }
- if (!isInCall()) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
- }
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE;
- if (device) break;
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
- if (device) break;
- device = mDefaultOutputDevice->mDeviceType;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
- }
- break;
- }
- break;
-
- case STRATEGY_SONIFICATION:
-
- // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
- // handleIncallSonification().
- if (isInCall()) {
- device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
- break;
- }
- // FALL THROUGH
-
- case STRATEGY_ENFORCED_AUDIBLE:
- // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
- // except:
- // - when in call where it doesn't default to STRATEGY_PHONE behavior
- // - in countries where not enforced in which case it follows STRATEGY_MEDIA
-
- if ((strategy == STRATEGY_SONIFICATION) ||
- (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
- device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
- }
- }
- // The second device used for sonification is the same as the device used by media strategy
- // FALL THROUGH
-
- // FIXME: STRATEGY_ACCESSIBILITY and STRATEGY_REROUTING follow STRATEGY_MEDIA for now
- case STRATEGY_ACCESSIBILITY:
- if (strategy == STRATEGY_ACCESSIBILITY) {
- // do not route accessibility prompts to a digital output currently configured with a
- // compressed format as they would likely not be mixed and dropped.
- for (size_t i = 0; i < mOutputs.size(); i++) {
- sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
- audio_devices_t devices = desc->device() &
- (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC);
- if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) &&
- devices != AUDIO_DEVICE_NONE) {
- availableOutputDeviceTypes = availableOutputDeviceTypes & ~devices;
- }
- }
- }
- // FALL THROUGH
-
- case STRATEGY_REROUTING:
- case STRATEGY_MEDIA: {
- uint32_t device2 = AUDIO_DEVICE_NONE;
- if (strategy != STRATEGY_SONIFICATION) {
- // no sonification on remote submix (e.g. WFD)
- if (mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0")) != 0) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
- }
- }
- if ((device2 == AUDIO_DEVICE_NONE) &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0)) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- }
- }
- if ((device2 == AUDIO_DEVICE_NONE) &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] == AUDIO_POLICY_FORCE_SPEAKER)) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
- }
- if ((device2 == AUDIO_DEVICE_NONE)) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- }
- if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
- // no sonification on aux digital (e.g. HDMI)
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- }
- if ((device2 == AUDIO_DEVICE_NONE) &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
- }
- int device3 = AUDIO_DEVICE_NONE;
- if (strategy == STRATEGY_MEDIA) {
- // ARC, SPDIF and AUX_LINE can co-exist with others.
- device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_HDMI_ARC;
- device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPDIF);
- device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_LINE);
- }
-
- device2 |= device3;
- // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
- // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
- device |= device2;
-
- // If hdmi system audio mode is on, remove speaker out of output list.
- if ((strategy == STRATEGY_MEDIA) &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] ==
- AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) {
- device &= ~AUDIO_DEVICE_OUT_SPEAKER;
- }
-
- if (device) break;
- device = mDefaultOutputDevice->mDeviceType;
- if (device == AUDIO_DEVICE_NONE) {
- ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
- }
- } break;
-
- default:
- ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
- break;
- }
-
- ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
- return device;
-}
-
-void AudioPolicyManager::updateDevicesAndOutputs()
-{
- for (int i = 0; i < NUM_STRATEGIES; i++) {
- mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
- }
- mPreviousOutputs = mOutputs;
-}
-
-uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
- audio_devices_t prevDevice,
- uint32_t delayMs)
-{
- // mute/unmute strategies using an incompatible device combination
- // if muting, wait for the audio in pcm buffer to be drained before proceeding
- // if unmuting, unmute only after the specified delay
- if (outputDesc->isDuplicated()) {
- return 0;
- }
-
- uint32_t muteWaitMs = 0;
- audio_devices_t device = outputDesc->device();
- bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2);
-
- for (size_t i = 0; i < NUM_STRATEGIES; i++) {
- audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
- curDevice = curDevice & outputDesc->mProfile->mSupportedDevices.types();
- bool mute = shouldMute && (curDevice & device) && (curDevice != device);
- bool doMute = false;
-
- if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
- doMute = true;
- outputDesc->mStrategyMutedByDevice[i] = true;
- } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
- doMute = true;
- outputDesc->mStrategyMutedByDevice[i] = false;
- }
- if (doMute) {
- for (size_t j = 0; j < mOutputs.size(); j++) {
- sp<AudioOutputDescriptor> desc = mOutputs.valueAt(j);
- // skip output if it does not share any device with current output
- if ((desc->supportedDevices() & outputDesc->supportedDevices())
- == AUDIO_DEVICE_NONE) {
- continue;
- }
- audio_io_handle_t curOutput = mOutputs.keyAt(j);
- ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
- mute ? "muting" : "unmuting", i, curDevice, curOutput);
- setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
- if (desc->isStrategyActive((routing_strategy)i)) {
- if (mute) {
- // FIXME: should not need to double latency if volume could be applied
- // immediately by the audioflinger mixer. We must account for the delay
- // between now and the next time the audioflinger thread for this output
- // will process a buffer (which corresponds to one buffer size,
- // usually 1/2 or 1/4 of the latency).
- if (muteWaitMs < desc->latency() * 2) {
- muteWaitMs = desc->latency() * 2;
- }
- }
- }
- }
- }
- }
-
- // temporary mute output if device selection changes to avoid volume bursts due to
- // different per device volumes
- if (outputDesc->isActive() && (device != prevDevice)) {
- if (muteWaitMs < outputDesc->latency() * 2) {
- muteWaitMs = outputDesc->latency() * 2;
- }
- for (size_t i = 0; i < NUM_STRATEGIES; i++) {
- if (outputDesc->isStrategyActive((routing_strategy)i)) {
- setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle);
- // do tempMute unmute after twice the mute wait time
- setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle,
- muteWaitMs *2, device);
- }
- }
- }
-
- // wait for the PCM output buffers to empty before proceeding with the rest of the command
- if (muteWaitMs > delayMs) {
- muteWaitMs -= delayMs;
- usleep(muteWaitMs * 1000);
- return muteWaitMs;
- }
- return 0;
-}
-
-uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output,
- audio_devices_t device,
- bool force,
- int delayMs,
- audio_patch_handle_t *patchHandle,
- const char* address)
-{
- ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- AudioParameter param;
- uint32_t muteWaitMs;
-
- if (outputDesc->isDuplicated()) {
- muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs);
- muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs);
- return muteWaitMs;
- }
- // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
- // output profile
- if ((device != AUDIO_DEVICE_NONE) &&
- ((device & outputDesc->mProfile->mSupportedDevices.types()) == 0)) {
- return 0;
- }
-
- // filter devices according to output selected
- device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types());
-
- audio_devices_t prevDevice = outputDesc->mDevice;
-
- ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
-
- if (device != AUDIO_DEVICE_NONE) {
- outputDesc->mDevice = device;
- }
- muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
-
- // Do not change the routing if:
- // the requested device is AUDIO_DEVICE_NONE
- // OR the requested device is the same as current device
- // AND force is not specified
- // AND the output is connected by a valid audio patch.
- // Doing this check here allows the caller to call setOutputDevice() without conditions
- if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force &&
- outputDesc->mPatchHandle != 0) {
- ALOGV("setOutputDevice() setting same device %04x or null device for output %d",
- device, output);
- return muteWaitMs;
- }
-
- ALOGV("setOutputDevice() changing device");
-
- // do the routing
- if (device == AUDIO_DEVICE_NONE) {
- resetOutputDevice(output, delayMs, NULL);
- } else {
- DeviceVector deviceList = (address == NULL) ?
- mAvailableOutputDevices.getDevicesFromType(device)
- : mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address));
- if (!deviceList.isEmpty()) {
- struct audio_patch patch;
- outputDesc->toAudioPortConfig(&patch.sources[0]);
- patch.num_sources = 1;
- patch.num_sinks = 0;
- for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) {
- deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]);
- patch.num_sinks++;
- }
- ssize_t index;
- if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
- index = mAudioPatches.indexOfKey(*patchHandle);
- } else {
- index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
- }
- sp< AudioPatch> patchDesc;
- audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- if (index >= 0) {
- patchDesc = mAudioPatches.valueAt(index);
- afPatchHandle = patchDesc->mAfPatchHandle;
- }
-
- status_t status = mpClientInterface->createAudioPatch(&patch,
- &afPatchHandle,
- delayMs);
- ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d"
- "num_sources %d num_sinks %d",
- status, afPatchHandle, patch.num_sources, patch.num_sinks);
- if (status == NO_ERROR) {
- if (index < 0) {
- patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &patch, mUidCached);
- addAudioPatch(patchDesc->mHandle, patchDesc);
- } else {
- patchDesc->mPatch = patch;
- }
- patchDesc->mAfPatchHandle = afPatchHandle;
- patchDesc->mUid = mUidCached;
- if (patchHandle) {
- *patchHandle = patchDesc->mHandle;
- }
- outputDesc->mPatchHandle = patchDesc->mHandle;
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- }
- }
-
- // inform all input as well
- for (size_t i = 0; i < mInputs.size(); i++) {
- const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
- if (!isVirtualInputDevice(inputDescriptor->mDevice)) {
- AudioParameter inputCmd = AudioParameter();
- ALOGV("%s: inform input %d of device:%d", __func__,
- inputDescriptor->mIoHandle, device);
- inputCmd.addInt(String8(AudioParameter::keyRouting),device);
- mpClientInterface->setParameters(inputDescriptor->mIoHandle,
- inputCmd.toString(),
- delayMs);
- }
- }
- }
-
- // update stream volumes according to new device
- applyStreamVolumes(output, device, delayMs);
-
- return muteWaitMs;
-}
-
-status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output,
- int delayMs,
- audio_patch_handle_t *patchHandle)
-{
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- ssize_t index;
- if (patchHandle) {
- index = mAudioPatches.indexOfKey(*patchHandle);
- } else {
- index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
- }
- if (index < 0) {
- return INVALID_OPERATION;
- }
- sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs);
- ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status);
- outputDesc->mPatchHandle = 0;
- removeAudioPatch(patchDesc->mHandle);
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- return status;
-}
-
-status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input,
- audio_devices_t device,
- bool force,
- audio_patch_handle_t *patchHandle)
-{
- status_t status = NO_ERROR;
-
- sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
- if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
- inputDesc->mDevice = device;
-
- DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device);
- if (!deviceList.isEmpty()) {
- struct audio_patch patch;
- inputDesc->toAudioPortConfig(&patch.sinks[0]);
- // AUDIO_SOURCE_HOTWORD is for internal use only:
- // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL
- if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD &&
- !inputDesc->mIsSoundTrigger) {
- patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION;
- }
- patch.num_sinks = 1;
- //only one input device for now
- deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]);
- patch.num_sources = 1;
- ssize_t index;
- if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
- index = mAudioPatches.indexOfKey(*patchHandle);
- } else {
- index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
- }
- sp< AudioPatch> patchDesc;
- audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- if (index >= 0) {
- patchDesc = mAudioPatches.valueAt(index);
- afPatchHandle = patchDesc->mAfPatchHandle;
- }
-
- status_t status = mpClientInterface->createAudioPatch(&patch,
- &afPatchHandle,
- 0);
- ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d",
- status, afPatchHandle);
- if (status == NO_ERROR) {
- if (index < 0) {
- patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &patch, mUidCached);
- addAudioPatch(patchDesc->mHandle, patchDesc);
- } else {
- patchDesc->mPatch = patch;
- }
- patchDesc->mAfPatchHandle = afPatchHandle;
- patchDesc->mUid = mUidCached;
- if (patchHandle) {
- *patchHandle = patchDesc->mHandle;
- }
- inputDesc->mPatchHandle = patchDesc->mHandle;
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- }
- }
- }
- return status;
-}
-
-status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input,
- audio_patch_handle_t *patchHandle)
-{
- sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
- ssize_t index;
- if (patchHandle) {
- index = mAudioPatches.indexOfKey(*patchHandle);
- } else {
- index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
- }
- if (index < 0) {
- return INVALID_OPERATION;
- }
- sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
- ALOGV("resetInputDevice() releaseAudioPatch returned %d", status);
- inputDesc->mPatchHandle = 0;
- removeAudioPatch(patchDesc->mHandle);
- nextAudioPortGeneration();
- mpClientInterface->onAudioPatchListUpdate();
- return status;
-}
-
-sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
- String8 address,
- uint32_t& samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_input_flags_t flags)
-{
- // Choose an input profile based on the requested capture parameters: select the first available
- // profile supporting all requested parameters.
-
- for (size_t i = 0; i < mHwModules.size(); i++)
- {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
- {
- sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j];
- // profile->log();
- if (profile->isCompatibleProfile(device, address, samplingRate,
- &samplingRate /*updatedSamplingRate*/,
- format, channelMask, (audio_output_flags_t) flags)) {
-
- return profile;
- }
- }
- }
- return NULL;
-}
-
-
-audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource,
- AudioMix **policyMix)
-{
- audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() &
- ~AUDIO_DEVICE_BIT_IN;
-
- for (size_t i = 0; i < mPolicyMixes.size(); i++) {
- if (mPolicyMixes[i]->mMix.mMixType != MIX_TYPE_RECORDERS) {
- continue;
- }
- for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) {
- if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
- mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource == inputSource) ||
- (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
- mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource != inputSource)) {
- if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
- if (policyMix != NULL) {
- *policyMix = &mPolicyMixes[i]->mMix;
- }
- return AUDIO_DEVICE_IN_REMOTE_SUBMIX;
- }
- break;
- }
- }
- }
-
- return getDeviceForInputSource(inputSource);
-}
-
-audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
-{
- uint32_t device = AUDIO_DEVICE_NONE;
- audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() &
- ~AUDIO_DEVICE_BIT_IN;
-
- switch (inputSource) {
- case AUDIO_SOURCE_VOICE_UPLINK:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
- device = AUDIO_DEVICE_IN_VOICE_CALL;
- break;
- }
- break;
-
- case AUDIO_SOURCE_DEFAULT:
- case AUDIO_SOURCE_MIC:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
- } else if ((mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO) &&
- (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
- device = AUDIO_DEVICE_IN_WIRED_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
- device = AUDIO_DEVICE_IN_USB_DEVICE;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
-
- case AUDIO_SOURCE_VOICE_COMMUNICATION:
- // Allow only use of devices on primary input if in call and HAL does not support routing
- // to voice call path.
- if ((mPhoneState == AUDIO_MODE_IN_CALL) &&
- (mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) {
- availableDeviceTypes = availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN;
- }
-
- switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
- case AUDIO_POLICY_FORCE_BT_SCO:
- // if SCO device is requested but no SCO device is available, fall back to default case
- if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
- break;
- }
- // FALL THROUGH
-
- default: // FORCE_NONE
- if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
- device = AUDIO_DEVICE_IN_WIRED_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
- device = AUDIO_DEVICE_IN_USB_DEVICE;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
-
- case AUDIO_POLICY_FORCE_SPEAKER:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
- device = AUDIO_DEVICE_IN_BACK_MIC;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
- }
- break;
-
- case AUDIO_SOURCE_VOICE_RECOGNITION:
- case AUDIO_SOURCE_HOTWORD:
- if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO &&
- availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
- device = AUDIO_DEVICE_IN_WIRED_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
- device = AUDIO_DEVICE_IN_USB_DEVICE;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
- case AUDIO_SOURCE_CAMCORDER:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
- device = AUDIO_DEVICE_IN_BACK_MIC;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
- case AUDIO_SOURCE_VOICE_DOWNLINK:
- case AUDIO_SOURCE_VOICE_CALL:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
- device = AUDIO_DEVICE_IN_VOICE_CALL;
- }
- break;
- case AUDIO_SOURCE_REMOTE_SUBMIX:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
- device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
- }
- break;
- case AUDIO_SOURCE_FM_TUNER:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) {
- device = AUDIO_DEVICE_IN_FM_TUNER;
- }
- break;
- default:
- ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
- break;
- }
- ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
- return device;
-}
-
-bool AudioPolicyManager::isVirtualInputDevice(audio_devices_t device)
-{
- if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
- device &= ~AUDIO_DEVICE_BIT_IN;
- if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0))
- return true;
- }
- return false;
-}
-
-bool AudioPolicyManager::deviceDistinguishesOnAddress(audio_devices_t device) {
- return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL & ~AUDIO_DEVICE_BIT_IN) != 0);
-}
-
-audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs)
-{
- for (size_t i = 0; i < mInputs.size(); i++) {
- const sp<AudioInputDescriptor> input_descriptor = mInputs.valueAt(i);
- if ((input_descriptor->mRefCount > 0)
- && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
- return mInputs.keyAt(i);
- }
- }
- return 0;
-}
-
-uint32_t AudioPolicyManager::activeInputsCount() const
-{
- uint32_t count = 0;
- for (size_t i = 0; i < mInputs.size(); i++) {
- const sp<AudioInputDescriptor> desc = mInputs.valueAt(i);
- if (desc->mRefCount > 0) {
- count++;
- }
- }
- return count;
-}
-
-
-audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device)
-{
- if (device == AUDIO_DEVICE_NONE) {
- // this happens when forcing a route update and no track is active on an output.
- // In this case the returned category is not important.
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else if (popcount(device) > 1) {
- // Multiple device selection is either:
- // - speaker + one other device: give priority to speaker in this case.
- // - one A2DP device + another device: happens with duplicated output. In this case
- // retain the device on the A2DP output as the other must not correspond to an active
- // selection if not the speaker.
- // - HDMI-CEC system audio mode only output: give priority to available item in order.
- if (device & AUDIO_DEVICE_OUT_SPEAKER) {
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
- device = AUDIO_DEVICE_OUT_HDMI_ARC;
- } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
- device = AUDIO_DEVICE_OUT_AUX_LINE;
- } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
- device = AUDIO_DEVICE_OUT_SPDIF;
- } else {
- device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
- }
- }
-
- /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
- if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
- device = AUDIO_DEVICE_OUT_SPEAKER;
-
- ALOGW_IF(popcount(device) != 1,
- "getDeviceForVolume() invalid device combination: %08x",
- device);
-
- return device;
-}
-
-AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device)
-{
- switch(getDeviceForVolume(device)) {
- case AUDIO_DEVICE_OUT_EARPIECE:
- return DEVICE_CATEGORY_EARPIECE;
- case AUDIO_DEVICE_OUT_WIRED_HEADSET:
- case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
- return DEVICE_CATEGORY_HEADSET;
- case AUDIO_DEVICE_OUT_LINE:
- case AUDIO_DEVICE_OUT_AUX_DIGITAL:
- /*USB? Remote submix?*/
- return DEVICE_CATEGORY_EXT_MEDIA;
- case AUDIO_DEVICE_OUT_SPEAKER:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
- case AUDIO_DEVICE_OUT_USB_ACCESSORY:
- case AUDIO_DEVICE_OUT_USB_DEVICE:
- case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
- default:
- return DEVICE_CATEGORY_SPEAKER;
- }
-}
-
-/* static */
-float AudioPolicyManager::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
- int indexInUi)
-{
- device_category deviceCategory = getDeviceCategory(device);
- const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
-
- // the volume index in the UI is relative to the min and max volume indices for this stream type
- int nbSteps = 1 + curve[VOLMAX].mIndex -
- curve[VOLMIN].mIndex;
- int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
- (streamDesc.mIndexMax - streamDesc.mIndexMin);
-
- // find what part of the curve this index volume belongs to, or if it's out of bounds
- int segment = 0;
- if (volIdx < curve[VOLMIN].mIndex) { // out of bounds
- return 0.0f;
- } else if (volIdx < curve[VOLKNEE1].mIndex) {
- segment = 0;
- } else if (volIdx < curve[VOLKNEE2].mIndex) {
- segment = 1;
- } else if (volIdx <= curve[VOLMAX].mIndex) {
- segment = 2;
- } else { // out of bounds
- return 1.0f;
- }
-
- // linear interpolation in the attenuation table in dB
- float decibels = curve[segment].mDBAttenuation +
- ((float)(volIdx - curve[segment].mIndex)) *
- ( (curve[segment+1].mDBAttenuation -
- curve[segment].mDBAttenuation) /
- ((float)(curve[segment+1].mIndex -
- curve[segment].mIndex)) );
-
- float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
-
- ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
- curve[segment].mIndex, volIdx,
- curve[segment+1].mIndex,
- curve[segment].mDBAttenuation,
- decibels,
- curve[segment+1].mDBAttenuation,
- amplification);
-
- return amplification;
-}
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
- {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
- {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
-};
-
-// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
-// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
-// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
-// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
- {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sLinearVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSilentVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sFullScaleVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- *AudioPolicyManager::sVolumeProfiles[AUDIO_STREAM_CNT]
- [AudioPolicyManager::DEVICE_CATEGORY_CNT] = {
- { // AUDIO_STREAM_VOICE_CALL
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_SYSTEM
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_RING
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_MUSIC
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ALARM
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_NOTIFICATION
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_BLUETOOTH_SCO
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ENFORCED_AUDIBLE
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_DTMF
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_TTS
- // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER
- sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ACCESSIBILITY
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_REROUTING
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_PATCH
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
-};
-
-void AudioPolicyManager::initializeVolumeCurves()
-{
- for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
- mStreams[i].mVolumeCurve[j] =
- sVolumeProfiles[i][j];
- }
- }
-
- // Check availability of DRC on speaker path: if available, override some of the speaker curves
- if (mSpeakerDrcEnabled) {
- mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sDefaultSystemVolumeCurveDrc;
- mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerMediaVolumeCurveDrc;
- mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerMediaVolumeCurveDrc;
- }
-}
-
-float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
- int index,
- audio_io_handle_t output,
- audio_devices_t device)
-{
- float volume = 1.0;
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- StreamDescriptor &streamDesc = mStreams[stream];
-
- if (device == AUDIO_DEVICE_NONE) {
- device = outputDesc->device();
- }
-
- volume = volIndexToAmpl(device, streamDesc, index);
-
- // if a headset is connected, apply the following rules to ring tones and notifications
- // to avoid sound level bursts in user's ears:
- // - always attenuate ring tones and notifications volume by 6dB
- // - if music is playing, always limit the volume to current music volume,
- // with a minimum threshold at -36dB so that notification is always perceived.
- const routing_strategy stream_strategy = getStrategy(stream);
- if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
- AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
- AUDIO_DEVICE_OUT_WIRED_HEADSET |
- AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
- ((stream_strategy == STRATEGY_SONIFICATION)
- || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
- || (stream == AUDIO_STREAM_SYSTEM)
- || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
- (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) &&
- streamDesc.mCanBeMuted) {
- volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
- // when the phone is ringing we must consider that music could have been paused just before
- // by the music application and behave as if music was active if the last music track was
- // just stopped
- if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
- mLimitRingtoneVolume) {
- audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
- float musicVol = computeVolume(AUDIO_STREAM_MUSIC,
- mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice),
- output,
- musicDevice);
- float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
- musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
- if (volume > minVol) {
- volume = minVol;
- ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
- }
- }
- }
-
- return volume;
-}
-
-status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream,
- int index,
- audio_io_handle_t output,
- audio_devices_t device,
- int delayMs,
- bool force)
-{
-
- // do not change actual stream volume if the stream is muted
- if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
- ALOGVV("checkAndSetVolume() stream %d muted count %d",
- stream, mOutputs.valueFor(output)->mMuteCount[stream]);
- return NO_ERROR;
- }
-
- // do not change in call volume if bluetooth is connected and vice versa
- if ((stream == AUDIO_STREAM_VOICE_CALL &&
- mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
- (stream == AUDIO_STREAM_BLUETOOTH_SCO &&
- mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) {
- ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
- stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
- return INVALID_OPERATION;
- }
-
- float volume = computeVolume(stream, index, output, device);
- // unit gain if rerouting to external policy
- if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
- ssize_t index = mOutputs.indexOfKey(output);
- if (index >= 0) {
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
- if (outputDesc->mPolicyMix != NULL) {
- ALOGV("max gain when rerouting for output=%d", output);
- volume = 1.0f;
- }
- }
-
- }
- // We actually change the volume if:
- // - the float value returned by computeVolume() changed
- // - the force flag is set
- if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
- force) {
- mOutputs.valueFor(output)->mCurVolume[stream] = volume;
- ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
- // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
- // enabled
- if (stream == AUDIO_STREAM_BLUETOOTH_SCO) {
- mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs);
- }
- mpClientInterface->setStreamVolume(stream, volume, output, delayMs);
- }
-
- if (stream == AUDIO_STREAM_VOICE_CALL ||
- stream == AUDIO_STREAM_BLUETOOTH_SCO) {
- float voiceVolume;
- // Force voice volume to max for bluetooth SCO as volume is managed by the headset
- if (stream == AUDIO_STREAM_VOICE_CALL) {
- voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
- } else {
- voiceVolume = 1.0;
- }
-
- if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
- mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
- mLastVoiceVolume = voiceVolume;
- }
- }
-
- return NO_ERROR;
-}
-
-void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output,
- audio_devices_t device,
- int delayMs,
- bool force)
-{
- ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
-
- for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
- if (stream == AUDIO_STREAM_PATCH) {
- continue;
- }
- checkAndSetVolume((audio_stream_type_t)stream,
- mStreams[stream].getVolumeIndex(device),
- output,
- device,
- delayMs,
- force);
- }
-}
-
-void AudioPolicyManager::setStrategyMute(routing_strategy strategy,
- bool on,
- audio_io_handle_t output,
- int delayMs,
- audio_devices_t device)
-{
- ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
- for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
- if (stream == AUDIO_STREAM_PATCH) {
- continue;
- }
- if (getStrategy((audio_stream_type_t)stream) == strategy) {
- setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device);
- }
- }
-}
-
-void AudioPolicyManager::setStreamMute(audio_stream_type_t stream,
- bool on,
- audio_io_handle_t output,
- int delayMs,
- audio_devices_t device)
-{
- StreamDescriptor &streamDesc = mStreams[stream];
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- if (device == AUDIO_DEVICE_NONE) {
- device = outputDesc->device();
- }
-
- ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
- stream, on, output, outputDesc->mMuteCount[stream], device);
-
- if (on) {
- if (outputDesc->mMuteCount[stream] == 0) {
- if (streamDesc.mCanBeMuted &&
- ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) ||
- (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) {
- checkAndSetVolume(stream, 0, output, device, delayMs);
- }
- }
- // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
- outputDesc->mMuteCount[stream]++;
- } else {
- if (outputDesc->mMuteCount[stream] == 0) {
- ALOGV("setStreamMute() unmuting non muted stream!");
- return;
- }
- if (--outputDesc->mMuteCount[stream] == 0) {
- checkAndSetVolume(stream,
- streamDesc.getVolumeIndex(device),
- output,
- device,
- delayMs);
- }
- }
-}
-
-void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream,
- bool starting, bool stateChange)
-{
- // if the stream pertains to sonification strategy and we are in call we must
- // mute the stream if it is low visibility. If it is high visibility, we must play a tone
- // in the device used for phone strategy and play the tone if the selected device does not
- // interfere with the device used for phone strategy
- // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
- // many times as there are active tracks on the output
- const routing_strategy stream_strategy = getStrategy(stream);
- if ((stream_strategy == STRATEGY_SONIFICATION) ||
- ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
- ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
- stream, starting, outputDesc->mDevice, stateChange);
- if (outputDesc->mRefCount[stream]) {
- int muteCount = 1;
- if (stateChange) {
- muteCount = outputDesc->mRefCount[stream];
- }
- if (audio_is_low_visibility(stream)) {
- ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
- for (int i = 0; i < muteCount; i++) {
- setStreamMute(stream, starting, mPrimaryOutput);
- }
- } else {
- ALOGV("handleIncallSonification() high visibility");
- if (outputDesc->device() &
- getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
- ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
- for (int i = 0; i < muteCount; i++) {
- setStreamMute(stream, starting, mPrimaryOutput);
- }
- }
- if (starting) {
- mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
- AUDIO_STREAM_VOICE_CALL);
- } else {
- mpClientInterface->stopTone();
- }
- }
- }
- }
-}
-
-bool AudioPolicyManager::isInCall()
-{
- return isStateInCall(mPhoneState);
-}
-
-bool AudioPolicyManager::isStateInCall(int state) {
- return ((state == AUDIO_MODE_IN_CALL) ||
- (state == AUDIO_MODE_IN_COMMUNICATION));
-}
-
-uint32_t AudioPolicyManager::getMaxEffectsCpuLoad()
-{
- return MAX_EFFECTS_CPU_LOAD;
-}
-
-uint32_t AudioPolicyManager::getMaxEffectsMemory()
-{
- return MAX_EFFECTS_MEMORY;
-}
-
-
-// --- AudioOutputDescriptor class implementation
-
-AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor(
- const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0), mLatency(0),
- mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
- mPatchHandle(0),
- mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
-{
- // clear usage count for all stream types
- for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- mRefCount[i] = 0;
- mCurVolume[i] = -1.0;
- mMuteCount[i] = 0;
- mStopTime[i] = 0;
- }
- for (int i = 0; i < NUM_STRATEGIES; i++) {
- mStrategyMutedByDevice[i] = false;
- }
- if (profile != NULL) {
- mFlags = (audio_output_flags_t)profile->mFlags;
- mSamplingRate = profile->pickSamplingRate();
- mFormat = profile->pickFormat();
- mChannelMask = profile->pickChannelMask();
- if (profile->mGains.size() > 0) {
- profile->mGains[0]->getDefaultConfig(&mGain);
- }
- }
-}
-
-audio_devices_t AudioPolicyManager::AudioOutputDescriptor::device() const
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
- } else {
- return mDevice;
- }
-}
-
-uint32_t AudioPolicyManager::AudioOutputDescriptor::latency()
-{
- if (isDuplicated()) {
- return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
- } else {
- return mLatency;
- }
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::sharesHwModuleWith(
- const sp<AudioOutputDescriptor> outputDesc)
-{
- if (isDuplicated()) {
- return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
- } else if (outputDesc->isDuplicated()){
- return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
- } else {
- return (mProfile->mModule == outputDesc->mProfile->mModule);
- }
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
- int delta)
-{
- // forward usage count change to attached outputs
- if (isDuplicated()) {
- mOutput1->changeRefCount(stream, delta);
- mOutput2->changeRefCount(stream, delta);
- }
- if ((delta + (int)mRefCount[stream]) < 0) {
- ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
- delta, stream, mRefCount[stream]);
- mRefCount[stream] = 0;
- return;
- }
- mRefCount[stream] += delta;
- ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
-}
-
-audio_devices_t AudioPolicyManager::AudioOutputDescriptor::supportedDevices()
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
- } else {
- return mProfile->mSupportedDevices.types() ;
- }
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
-{
- return isStrategyActive(NUM_STRATEGIES, inPastMs);
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
- uint32_t inPastMs,
- nsecs_t sysTime) const
-{
- if ((sysTime == 0) && (inPastMs != 0)) {
- sysTime = systemTime();
- }
- for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
- if (i == AUDIO_STREAM_PATCH) {
- continue;
- }
- if (((getStrategy((audio_stream_type_t)i) == strategy) ||
- (NUM_STRATEGIES == strategy)) &&
- isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
- uint32_t inPastMs,
- nsecs_t sysTime) const
-{
- if (mRefCount[stream] != 0) {
- return true;
- }
- if (inPastMs == 0) {
- return false;
- }
- if (sysTime == 0) {
- sysTime = systemTime();
- }
- if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
- return true;
- }
- return false;
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
-
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
- dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
- dstConfig->ext.mix.handle = mIoHandle;
- dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::toAudioPort(
- struct audio_port *port) const
-{
- ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
- mProfile->toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.mix.hw_module = mProfile->mModule->mHandle;
- port->ext.mix.handle = mIoHandle;
- port->ext.mix.latency_class =
- mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioPolicyManager::AudioOutputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " ID: %d\n", mId);
- result.append(buffer);
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
- result.append(buffer);
- snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", device());
- result.append(buffer);
- snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
- result.append(buffer);
- for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
- snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n",
- i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- AudioInputDescriptor class implementation
-
-AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0),
- mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
- mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
-{
- if (profile != NULL) {
- mSamplingRate = profile->pickSamplingRate();
- mFormat = profile->pickFormat();
- mChannelMask = profile->pickChannelMask();
- if (profile->mGains.size() > 0) {
- profile->mGains[0]->getDefaultConfig(&mGain);
- }
- }
-}
-
-void AudioPolicyManager::AudioInputDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- ALOG_ASSERT(mProfile != 0,
- "toAudioPortConfig() called on input with null profile %d", mIoHandle);
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
-
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = AUDIO_PORT_ROLE_SINK;
- dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
- dstConfig->ext.mix.handle = mIoHandle;
- dstConfig->ext.mix.usecase.source = mInputSource;
-}
-
-void AudioPolicyManager::AudioInputDescriptor::toAudioPort(
- struct audio_port *port) const
-{
- ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
-
- mProfile->toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.mix.hw_module = mProfile->mModule->mHandle;
- port->ext.mix.handle = mIoHandle;
- port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioPolicyManager::AudioInputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " ID: %d\n", mId);
- result.append(buffer);
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %d\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
- result.append(buffer);
- snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
- result.append(buffer);
- snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
- result.append(buffer);
-
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- StreamDescriptor class implementation
-
-AudioPolicyManager::StreamDescriptor::StreamDescriptor()
- : mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
-{
- mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
-}
-
-int AudioPolicyManager::StreamDescriptor::getVolumeIndex(audio_devices_t device)
-{
- device = AudioPolicyManager::getDeviceForVolume(device);
- // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
- if (mIndexCur.indexOfKey(device) < 0) {
- device = AUDIO_DEVICE_OUT_DEFAULT;
- }
- return mIndexCur.valueFor(device);
-}
-
-void AudioPolicyManager::StreamDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%s %02d %02d ",
- mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
- result.append(buffer);
- for (size_t i = 0; i < mIndexCur.size(); i++) {
- snprintf(buffer, SIZE, "%04x : %02d, ",
- mIndexCur.keyAt(i),
- mIndexCur.valueAt(i));
- result.append(buffer);
- }
- result.append("\n");
-
- write(fd, result.string(), result.size());
-}
-
-// --- EffectDescriptor class implementation
-
-status_t AudioPolicyManager::EffectDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " I/O: %d\n", mIo);
- result.append(buffer);
- snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
- result.append(buffer);
- snprintf(buffer, SIZE, " Session: %d\n", mSession);
- result.append(buffer);
- snprintf(buffer, SIZE, " Name: %s\n", mDesc.name);
- result.append(buffer);
- snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled");
- result.append(buffer);
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- HwModule class implementation
-
-AudioPolicyManager::HwModule::HwModule(const char *name)
- : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
- mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
-{
-}
-
-AudioPolicyManager::HwModule::~HwModule()
-{
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- mOutputProfiles[i]->mSupportedDevices.clear();
- }
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- mInputProfiles[i]->mSupportedDevices.clear();
- }
- free((void *)mName);
-}
-
-status_t AudioPolicyManager::HwModule::loadInput(cnode *root)
-{
- cnode *node = root->first_child;
-
- sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->loadSamplingRates((char *)node->value);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->loadFormats((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->loadInChannels((char *)node->value);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
- mDeclaredDevices);
- } else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->mFlags = parseInputFlagNames((char *)node->value);
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- profile->loadGains(node);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices.isEmpty(),
- "loadInput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadInput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadInput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadInput() invalid supported formats");
- if (!profile->mSupportedDevices.isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadInput() adding input Supported Devices %04x",
- profile->mSupportedDevices.types());
-
- mInputProfiles.add(profile);
- return NO_ERROR;
- } else {
- return BAD_VALUE;
- }
-}
-
-status_t AudioPolicyManager::HwModule::loadOutput(cnode *root)
-{
- cnode *node = root->first_child;
-
- sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->loadSamplingRates((char *)node->value);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->loadFormats((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->loadOutChannels((char *)node->value);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
- mDeclaredDevices);
- } else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->mFlags = parseOutputFlagNames((char *)node->value);
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- profile->loadGains(node);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices.isEmpty(),
- "loadOutput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadOutput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadOutput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadOutput() invalid supported formats");
- if (!profile->mSupportedDevices.isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
- profile->mSupportedDevices.types(), profile->mFlags);
-
- mOutputProfiles.add(profile);
- return NO_ERROR;
- } else {
- return BAD_VALUE;
- }
-}
-
-status_t AudioPolicyManager::HwModule::loadDevice(cnode *root)
-{
- cnode *node = root->first_child;
-
- audio_devices_t type = AUDIO_DEVICE_NONE;
- while (node) {
- if (strcmp(node->name, DEVICE_TYPE) == 0) {
- type = parseDeviceNames((char *)node->value);
- break;
- }
- node = node->next;
- }
- if (type == AUDIO_DEVICE_NONE ||
- (!audio_is_input_device(type) && !audio_is_output_device(type))) {
- ALOGW("loadDevice() bad type %08x", type);
- return BAD_VALUE;
- }
- sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
- deviceDesc->mModule = this;
-
- node = root->first_child;
- while (node) {
- if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
- deviceDesc->mAddress = String8((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- if (audio_is_input_device(type)) {
- deviceDesc->loadInChannels((char *)node->value);
- } else {
- deviceDesc->loadOutChannels((char *)node->value);
- }
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- deviceDesc->loadGains(node);
- }
- node = node->next;
- }
-
- ALOGV("loadDevice() adding device name %s type %08x address %s",
- deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
-
- mDeclaredDevices.add(deviceDesc);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::addOutputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address)
-{
- sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this);
-
- profile->mSamplingRates.add(config->sample_rate);
- profile->mChannelMasks.add(config->channel_mask);
- profile->mFormats.add(config->format);
-
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
- devDesc->mAddress = address;
- profile->mSupportedDevices.add(devDesc);
-
- mOutputProfiles.add(profile);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::removeOutputProfile(String8 name)
-{
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- if (mOutputProfiles[i]->mName == name) {
- mOutputProfiles.removeAt(i);
- break;
- }
- }
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::addInputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address)
-{
- sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this);
-
- profile->mSamplingRates.add(config->sample_rate);
- profile->mChannelMasks.add(config->channel_mask);
- profile->mFormats.add(config->format);
-
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
- devDesc->mAddress = address;
- profile->mSupportedDevices.add(devDesc);
-
- ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
-
- mInputProfiles.add(profile);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::removeInputProfile(String8 name)
-{
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- if (mInputProfiles[i]->mName == name) {
- mInputProfiles.removeAt(i);
- break;
- }
- }
-
- return NO_ERROR;
-}
-
-
-void AudioPolicyManager::HwModule::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " - name: %s\n", mName);
- result.append(buffer);
- snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
- result.append(buffer);
- write(fd, result.string(), result.size());
- if (mOutputProfiles.size()) {
- write(fd, " - outputs:\n", strlen(" - outputs:\n"));
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " output %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mOutputProfiles[i]->dump(fd);
- }
- }
- if (mInputProfiles.size()) {
- write(fd, " - inputs:\n", strlen(" - inputs:\n"));
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " input %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mInputProfiles[i]->dump(fd);
- }
- }
- if (mDeclaredDevices.size()) {
- write(fd, " - devices:\n", strlen(" - devices:\n"));
- for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
- mDeclaredDevices[i]->dump(fd, 4, i);
- }
- }
-}
-
-// --- AudioPort class implementation
-
-
-AudioPolicyManager::AudioPort::AudioPort(const String8& name, audio_port_type_t type,
- audio_port_role_t role, const sp<HwModule>& module) :
- mName(name), mType(type), mRole(role), mModule(module), mFlags(0)
-{
- mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
- ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
-}
-
-void AudioPolicyManager::AudioPort::toAudioPort(struct audio_port *port) const
-{
- port->role = mRole;
- port->type = mType;
- unsigned int i;
- for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
- if (mSamplingRates[i] != 0) {
- port->sample_rates[i] = mSamplingRates[i];
- }
- }
- port->num_sample_rates = i;
- for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
- if (mChannelMasks[i] != 0) {
- port->channel_masks[i] = mChannelMasks[i];
- }
- }
- port->num_channel_masks = i;
- for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
- if (mFormats[i] != 0) {
- port->formats[i] = mFormats[i];
- }
- }
- port->num_formats = i;
-
- ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
-
- for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
- port->gains[i] = mGains[i]->mGain;
- }
- port->num_gains = i;
-}
-
-void AudioPolicyManager::AudioPort::importAudioPort(const sp<AudioPort> port) {
- for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
- const uint32_t rate = port->mSamplingRates.itemAt(k);
- if (rate != 0) { // skip "dynamic" rates
- bool hasRate = false;
- for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
- if (rate == mSamplingRates.itemAt(l)) {
- hasRate = true;
- break;
- }
- }
- if (!hasRate) { // never import a sampling rate twice
- mSamplingRates.add(rate);
- }
- }
- }
- for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
- const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
- if (mask != 0) { // skip "dynamic" masks
- bool hasMask = false;
- for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
- if (mask == mChannelMasks.itemAt(l)) {
- hasMask = true;
- break;
- }
- }
- if (!hasMask) { // never import a channel mask twice
- mChannelMasks.add(mask);
- }
- }
- }
- for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
- const audio_format_t format = port->mFormats.itemAt(k);
- if (format != 0) { // skip "dynamic" formats
- bool hasFormat = false;
- for (size_t l = 0 ; l < mFormats.size() ; l++) {
- if (format == mFormats.itemAt(l)) {
- hasFormat = true;
- break;
- }
- }
- if (!hasFormat) { // never import a channel mask twice
- mFormats.add(format);
- }
- }
- }
- for (size_t k = 0 ; k < port->mGains.size() ; k++) {
- sp<AudioGain> gain = port->mGains.itemAt(k);
- if (gain != 0) {
- bool hasGain = false;
- for (size_t l = 0 ; l < mGains.size() ; l++) {
- if (gain == mGains.itemAt(l)) {
- hasGain = true;
- break;
- }
- }
- if (!hasGain) { // never import a gain twice
- mGains.add(gain);
- }
- }
- }
-}
-
-void AudioPolicyManager::AudioPort::clearCapabilities() {
- mChannelMasks.clear();
- mFormats.clear();
- mSamplingRates.clear();
- mGains.clear();
-}
-
-void AudioPolicyManager::AudioPort::loadSamplingRates(char *name)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
- // rates should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mSamplingRates.add(0);
- return;
- }
-
- while (str != NULL) {
- uint32_t rate = atoi(str);
- if (rate != 0) {
- ALOGV("loadSamplingRates() adding rate %d", rate);
- mSamplingRates.add(rate);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadFormats(char *name)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mFormats indicates the supported formats
- // should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mFormats.add(AUDIO_FORMAT_DEFAULT);
- return;
- }
-
- while (str != NULL) {
- audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- str);
- if (format != AUDIO_FORMAT_DEFAULT) {
- mFormats.add(format);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadInChannels(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadInChannels() %s", name);
-
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
- ARRAY_SIZE(sInChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- ALOGV("loadInChannels() adding channelMask %04x", channelMask);
- mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadOutChannels(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadOutChannels() %s", name);
-
- // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
- // masks should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
- ARRAY_SIZE(sOutChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
- return;
-}
-
-audio_gain_mode_t AudioPolicyManager::AudioPort::loadGainMode(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadGainMode() %s", name);
- audio_gain_mode_t mode = 0;
- while (str != NULL) {
- mode |= (audio_gain_mode_t)stringToEnum(sGainModeNameToEnumTable,
- ARRAY_SIZE(sGainModeNameToEnumTable),
- str);
- str = strtok(NULL, "|");
- }
- return mode;
-}
-
-void AudioPolicyManager::AudioPort::loadGain(cnode *root, int index)
-{
- cnode *node = root->first_child;
-
- sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
-
- while (node) {
- if (strcmp(node->name, GAIN_MODE) == 0) {
- gain->mGain.mode = loadGainMode((char *)node->value);
- } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
- if (mUseInChannelMask) {
- gain->mGain.channel_mask =
- (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
- ARRAY_SIZE(sInChannelsNameToEnumTable),
- (char *)node->value);
- } else {
- gain->mGain.channel_mask =
- (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
- ARRAY_SIZE(sOutChannelsNameToEnumTable),
- (char *)node->value);
- }
- } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
- gain->mGain.min_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
- gain->mGain.max_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
- gain->mGain.default_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
- gain->mGain.step_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
- gain->mGain.min_ramp_ms = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
- gain->mGain.max_ramp_ms = atoi((char *)node->value);
- }
- node = node->next;
- }
-
- ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
- gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
-
- if (gain->mGain.mode == 0) {
- return;
- }
- mGains.add(gain);
-}
-
-void AudioPolicyManager::AudioPort::loadGains(cnode *root)
-{
- cnode *node = root->first_child;
- int index = 0;
- while (node) {
- ALOGV("loadGains() loading gain %s", node->name);
- loadGain(node, index++);
- node = node->next;
- }
-}
-
-status_t AudioPolicyManager::AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
-{
- if (mSamplingRates.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if (mSamplingRates[i] == samplingRate) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t *updatedSamplingRate) const
-{
- if (mSamplingRates.isEmpty()) {
- return NO_ERROR;
- }
-
- // Search for the closest supported sampling rate that is above (preferred)
- // or below (acceptable) the desired sampling rate, within a permitted ratio.
- // The sampling rates do not need to be sorted in ascending order.
- ssize_t maxBelow = -1;
- ssize_t minAbove = -1;
- uint32_t candidate;
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- candidate = mSamplingRates[i];
- if (candidate == samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- // candidate < desired
- if (candidate < samplingRate) {
- if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
- maxBelow = i;
- }
- // candidate > desired
- } else {
- if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
- minAbove = i;
- }
- }
- }
- // This uses hard-coded knowledge about AudioFlinger resampling ratios.
- // TODO Move these assumptions out.
- static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs
- static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur
- // due to approximation by an int32_t of the
- // phase increments
- // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
- if (minAbove >= 0) {
- candidate = mSamplingRates[minAbove];
- if (candidate / kMaxDownSampleRatio <= samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- }
- // But if we have to up-sample from a lower sampling rate, that's OK.
- if (maxBelow >= 0) {
- candidate = mSamplingRates[maxBelow];
- if (candidate * kMaxUpSampleRatio >= samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- }
- // leave updatedSamplingRate unmodified
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
-{
- if (mChannelMasks.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- if (mChannelMasks[i] == channelMask) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
- const
-{
- if (mChannelMasks.isEmpty()) {
- return NO_ERROR;
- }
-
- const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- // FIXME Does not handle multi-channel automatic conversions yet
- audio_channel_mask_t supported = mChannelMasks[i];
- if (supported == channelMask) {
- return NO_ERROR;
- }
- if (isRecordThread) {
- // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
- // FIXME Abstract this out to a table.
- if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
- && channelMask == AUDIO_CHANNEL_IN_MONO) ||
- (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
- || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
- return NO_ERROR;
- }
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkFormat(audio_format_t format) const
-{
- if (mFormats.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mFormats.size(); i ++) {
- if (mFormats[i] == format) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-
-uint32_t AudioPolicyManager::AudioPort::pickSamplingRate() const
-{
- // special case for uninitialized dynamic profile
- if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
- return 0;
- }
-
- // For direct outputs, pick minimum sampling rate: this helps ensuring that the
- // channel count / sampling rate combination chosen will be supported by the connected
- // sink
- if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
- uint32_t samplingRate = UINT_MAX;
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
- samplingRate = mSamplingRates[i];
- }
- }
- return (samplingRate == UINT_MAX) ? 0 : samplingRate;
- }
-
- uint32_t samplingRate = 0;
- uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
-
- // For mixed output and inputs, use max mixer sampling rates. Do not
- // limit sampling rate otherwise
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxRate = UINT_MAX;
- }
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
- samplingRate = mSamplingRates[i];
- }
- }
- return samplingRate;
-}
-
-audio_channel_mask_t AudioPolicyManager::AudioPort::pickChannelMask() const
-{
- // special case for uninitialized dynamic profile
- if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
- return AUDIO_CHANNEL_NONE;
- }
- audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
-
- // For direct outputs, pick minimum channel count: this helps ensuring that the
- // channel count / sampling rate combination chosen will be supported by the connected
- // sink
- if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
- uint32_t channelCount = UINT_MAX;
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- uint32_t cnlCount;
- if (mUseInChannelMask) {
- cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
- } else {
- cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
- }
- if ((cnlCount < channelCount) && (cnlCount > 0)) {
- channelMask = mChannelMasks[i];
- channelCount = cnlCount;
- }
- }
- return channelMask;
- }
-
- uint32_t channelCount = 0;
- uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
-
- // For mixed output and inputs, use max mixer channel count. Do not
- // limit channel count otherwise
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxCount = UINT_MAX;
- }
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- uint32_t cnlCount;
- if (mUseInChannelMask) {
- cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
- } else {
- cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
- }
- if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
- channelMask = mChannelMasks[i];
- channelCount = cnlCount;
- }
- }
- return channelMask;
-}
-
-/* format in order of increasing preference */
-const audio_format_t AudioPolicyManager::AudioPort::sPcmFormatCompareTable[] = {
- AUDIO_FORMAT_DEFAULT,
- AUDIO_FORMAT_PCM_16_BIT,
- AUDIO_FORMAT_PCM_8_24_BIT,
- AUDIO_FORMAT_PCM_24_BIT_PACKED,
- AUDIO_FORMAT_PCM_32_BIT,
- AUDIO_FORMAT_PCM_FLOAT,
-};
-
-int AudioPolicyManager::AudioPort::compareFormats(audio_format_t format1,
- audio_format_t format2)
-{
- // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
- // compressed format and better than any PCM format. This is by design of pickFormat()
- if (!audio_is_linear_pcm(format1)) {
- if (!audio_is_linear_pcm(format2)) {
- return 0;
- }
- return 1;
- }
- if (!audio_is_linear_pcm(format2)) {
- return -1;
- }
-
- int index1 = -1, index2 = -1;
- for (size_t i = 0;
- (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
- i ++) {
- if (sPcmFormatCompareTable[i] == format1) {
- index1 = i;
- }
- if (sPcmFormatCompareTable[i] == format2) {
- index2 = i;
- }
- }
- // format1 not found => index1 < 0 => format2 > format1
- // format2 not found => index2 < 0 => format2 < format1
- return index1 - index2;
-}
-
-audio_format_t AudioPolicyManager::AudioPort::pickFormat() const
-{
- // special case for uninitialized dynamic profile
- if (mFormats.size() == 1 && mFormats[0] == 0) {
- return AUDIO_FORMAT_DEFAULT;
- }
-
- audio_format_t format = AUDIO_FORMAT_DEFAULT;
- audio_format_t bestFormat =
- AudioPolicyManager::AudioPort::sPcmFormatCompareTable[
- ARRAY_SIZE(AudioPolicyManager::AudioPort::sPcmFormatCompareTable) - 1];
- // For mixed output and inputs, use best mixer output format. Do not
- // limit format otherwise
- if ((mType != AUDIO_PORT_TYPE_MIX) ||
- ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
- bestFormat = AUDIO_FORMAT_INVALID;
- }
-
- for (size_t i = 0; i < mFormats.size(); i ++) {
- if ((compareFormats(mFormats[i], format) > 0) &&
- (compareFormats(mFormats[i], bestFormat) <= 0)) {
- format = mFormats[i];
- }
- }
- return format;
-}
-
-status_t AudioPolicyManager::AudioPort::checkGain(const struct audio_gain_config *gainConfig,
- int index) const
-{
- if (index < 0 || (size_t)index >= mGains.size()) {
- return BAD_VALUE;
- }
- return mGains[index]->checkConfig(gainConfig);
-}
-
-void AudioPolicyManager::AudioPort::dump(int fd, int spaces) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- if (mName.size() != 0) {
- snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
- result.append(buffer);
- }
-
- if (mSamplingRates.size() != 0) {
- snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- if (i == 0 && mSamplingRates[i] == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
- }
- result.append(buffer);
- result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mChannelMasks.size() != 0) {
- snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
-
- if (i == 0 && mChannelMasks[i] == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
- }
- result.append(buffer);
- result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mFormats.size() != 0) {
- snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mFormats.size(); i++) {
- const char *formatStr = enumToString(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- mFormats[i]);
- if (i == 0 && strcmp(formatStr, "") == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "%s", formatStr);
- }
- result.append(buffer);
- result.append(i == (mFormats.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
- write(fd, result.string(), result.size());
- if (mGains.size() != 0) {
- snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
- write(fd, buffer, strlen(buffer) + 1);
- result.append(buffer);
- for (size_t i = 0; i < mGains.size(); i++) {
- mGains[i]->dump(fd, spaces + 2, i);
- }
- }
-}
-
-// --- AudioGain class implementation
-
-AudioPolicyManager::AudioGain::AudioGain(int index, bool useInChannelMask)
-{
- mIndex = index;
- mUseInChannelMask = useInChannelMask;
- memset(&mGain, 0, sizeof(struct audio_gain));
-}
-
-void AudioPolicyManager::AudioGain::getDefaultConfig(struct audio_gain_config *config)
-{
- config->index = mIndex;
- config->mode = mGain.mode;
- config->channel_mask = mGain.channel_mask;
- if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- config->values[0] = mGain.default_value;
- } else {
- uint32_t numValues;
- if (mUseInChannelMask) {
- numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
- } else {
- numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
- }
- for (size_t i = 0; i < numValues; i++) {
- config->values[i] = mGain.default_value;
- }
- }
- if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- config->ramp_duration_ms = mGain.min_ramp_ms;
- }
-}
-
-status_t AudioPolicyManager::AudioGain::checkConfig(const struct audio_gain_config *config)
-{
- if ((config->mode & ~mGain.mode) != 0) {
- return BAD_VALUE;
- }
- if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- if ((config->values[0] < mGain.min_value) ||
- (config->values[0] > mGain.max_value)) {
- return BAD_VALUE;
- }
- } else {
- if ((config->channel_mask & ~mGain.channel_mask) != 0) {
- return BAD_VALUE;
- }
- uint32_t numValues;
- if (mUseInChannelMask) {
- numValues = audio_channel_count_from_in_mask(config->channel_mask);
- } else {
- numValues = audio_channel_count_from_out_mask(config->channel_mask);
- }
- for (size_t i = 0; i < numValues; i++) {
- if ((config->values[i] < mGain.min_value) ||
- (config->values[i] > mGain.max_value)) {
- return BAD_VALUE;
- }
- }
- }
- if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
- (config->ramp_duration_ms > mGain.max_ramp_ms)) {
- return BAD_VALUE;
- }
- }
- return NO_ERROR;
-}
-
-void AudioPolicyManager::AudioGain::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
- result.append(buffer);
-
- write(fd, result.string(), result.size());
-}
-
-// --- AudioPortConfig class implementation
-
-AudioPolicyManager::AudioPortConfig::AudioPortConfig()
-{
- mSamplingRate = 0;
- mChannelMask = AUDIO_CHANNEL_NONE;
- mFormat = AUDIO_FORMAT_INVALID;
- mGain.index = -1;
-}
-
-status_t AudioPolicyManager::AudioPortConfig::applyAudioPortConfig(
- const struct audio_port_config *config,
- struct audio_port_config *backupConfig)
-{
- struct audio_port_config localBackupConfig;
- status_t status = NO_ERROR;
-
- localBackupConfig.config_mask = config->config_mask;
- toAudioPortConfig(&localBackupConfig);
-
- sp<AudioPort> audioport = getAudioPort();
- if (audioport == 0) {
- status = NO_INIT;
- goto exit;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- status = audioport->checkExactSamplingRate(config->sample_rate);
- if (status != NO_ERROR) {
- goto exit;
- }
- mSamplingRate = config->sample_rate;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- status = audioport->checkExactChannelMask(config->channel_mask);
- if (status != NO_ERROR) {
- goto exit;
- }
- mChannelMask = config->channel_mask;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- status = audioport->checkFormat(config->format);
- if (status != NO_ERROR) {
- goto exit;
- }
- mFormat = config->format;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- status = audioport->checkGain(&config->gain, config->gain.index);
- if (status != NO_ERROR) {
- goto exit;
- }
- mGain = config->gain;
- }
-
-exit:
- if (status != NO_ERROR) {
- applyAudioPortConfig(&localBackupConfig);
- }
- if (backupConfig != NULL) {
- *backupConfig = localBackupConfig;
- }
- return status;
-}
-
-void AudioPolicyManager::AudioPortConfig::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- dstConfig->sample_rate = mSamplingRate;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
- dstConfig->sample_rate = srcConfig->sample_rate;
- }
- } else {
- dstConfig->sample_rate = 0;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- dstConfig->channel_mask = mChannelMask;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
- dstConfig->channel_mask = srcConfig->channel_mask;
- }
- } else {
- dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- dstConfig->format = mFormat;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
- dstConfig->format = srcConfig->format;
- }
- } else {
- dstConfig->format = AUDIO_FORMAT_INVALID;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- dstConfig->gain = mGain;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
- dstConfig->gain = srcConfig->gain;
- }
- } else {
- dstConfig->gain.index = -1;
- }
- if (dstConfig->gain.index != -1) {
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
- } else {
- dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
- }
-}
-
-// --- IOProfile class implementation
-
-AudioPolicyManager::IOProfile::IOProfile(const String8& name, audio_port_role_t role,
- const sp<HwModule>& module)
- : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
-{
-}
-
-AudioPolicyManager::IOProfile::~IOProfile()
-{
-}
-
-// checks if the IO profile is compatible with specified parameters.
-// Sampling rate, format and channel mask must be specified in order to
-// get a valid a match
-bool AudioPolicyManager::IOProfile::isCompatibleProfile(audio_devices_t device,
- String8 address,
- uint32_t samplingRate,
- uint32_t *updatedSamplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- uint32_t flags) const
-{
- const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
- const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
- ALOG_ASSERT(isPlaybackThread != isRecordThread);
-
- if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) {
- return false;
- }
-
- if (samplingRate == 0) {
- return false;
- }
- uint32_t myUpdatedSamplingRate = samplingRate;
- if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
- return false;
- }
- if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
- NO_ERROR) {
- return false;
- }
-
- if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
- return false;
- }
-
- if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
- checkExactChannelMask(channelMask) != NO_ERROR)) {
- return false;
- }
- if (isRecordThread && (!audio_is_input_channel(channelMask) ||
- checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
- return false;
- }
-
- if (isPlaybackThread && (mFlags & flags) != flags) {
- return false;
- }
- // The only input flag that is allowed to be different is the fast flag.
- // An existing fast stream is compatible with a normal track request.
- // An existing normal stream is compatible with a fast track request,
- // but the fast request will be denied by AudioFlinger and converted to normal track.
- if (isRecordThread && ((mFlags ^ flags) &
- ~AUDIO_INPUT_FLAG_FAST)) {
- return false;
- }
-
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = myUpdatedSamplingRate;
- }
- return true;
-}
-
-void AudioPolicyManager::IOProfile::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- AudioPort::dump(fd, 4);
-
- snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " - devices:\n");
- result.append(buffer);
- write(fd, result.string(), result.size());
- for (size_t i = 0; i < mSupportedDevices.size(); i++) {
- mSupportedDevices[i]->dump(fd, 6, i);
- }
-}
-
-void AudioPolicyManager::IOProfile::log()
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- ALOGV(" - sampling rates: ");
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- ALOGV(" %d", mSamplingRates[i]);
- }
-
- ALOGV(" - channel masks: ");
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV(" 0x%04x", mChannelMasks[i]);
- }
-
- ALOGV(" - formats: ");
- for (size_t i = 0; i < mFormats.size(); i++) {
- ALOGV(" 0x%08x", mFormats[i]);
- }
-
- ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types());
- ALOGV(" - flags: 0x%04x\n", mFlags);
-}
-
-
-// --- DeviceDescriptor implementation
-
-
-AudioPolicyManager::DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) :
- AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
- audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
- AUDIO_PORT_ROLE_SOURCE,
- NULL),
- mDeviceType(type), mAddress(""), mId(0)
-{
-}
-
-bool AudioPolicyManager::DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
-{
- // Devices are considered equal if they:
- // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
- // - have the same address or one device does not specify the address
- // - have the same channel mask or one device does not specify the channel mask
- return (mDeviceType == other->mDeviceType) &&
- (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
- (mChannelMask == 0 || other->mChannelMask == 0 ||
- mChannelMask == other->mChannelMask);
-}
-
-void AudioPolicyManager::DeviceDescriptor::loadGains(cnode *root)
-{
- AudioPort::loadGains(root);
- if (mGains.size() > 0) {
- mGains[0]->getDefaultConfig(&mGain);
- }
-}
-
-
-void AudioPolicyManager::DeviceVector::refreshTypes()
-{
- mDeviceTypes = AUDIO_DEVICE_NONE;
- for(size_t i = 0; i < size(); i++) {
- mDeviceTypes |= itemAt(i)->mDeviceType;
- }
- ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
-}
-
-ssize_t AudioPolicyManager::DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
-{
- for(size_t i = 0; i < size(); i++) {
- if (item->equals(itemAt(i))) {
- return i;
- }
- }
- return -1;
-}
-
-ssize_t AudioPolicyManager::DeviceVector::add(const sp<DeviceDescriptor>& item)
-{
- ssize_t ret = indexOf(item);
-
- if (ret < 0) {
- ret = SortedVector::add(item);
- if (ret >= 0) {
- refreshTypes();
- }
- } else {
- ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
- ret = -1;
- }
- return ret;
-}
-
-ssize_t AudioPolicyManager::DeviceVector::remove(const sp<DeviceDescriptor>& item)
-{
- size_t i;
- ssize_t ret = indexOf(item);
-
- if (ret < 0) {
- ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
- } else {
- ret = SortedVector::removeAt(ret);
- if (ret >= 0) {
- refreshTypes();
- }
- }
- return ret;
-}
-
-void AudioPolicyManager::DeviceVector::loadDevicesFromType(audio_devices_t types)
-{
- DeviceVector deviceList;
-
- uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
- types &= ~role_bit;
-
- while (types) {
- uint32_t i = 31 - __builtin_clz(types);
- uint32_t type = 1 << i;
- types &= ~type;
- add(new DeviceDescriptor(String8(""), type | role_bit));
- }
-}
-
-void AudioPolicyManager::DeviceVector::loadDevicesFromName(char *name,
- const DeviceVector& declaredDevices)
-{
- char *devName = strtok(name, "|");
- while (devName != NULL) {
- if (strlen(devName) != 0) {
- audio_devices_t type = stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- devName);
- if (type != AUDIO_DEVICE_NONE) {
- sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(""), type);
- if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
- type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
- dev->mAddress = String8("0");
- }
- add(dev);
- } else {
- sp<DeviceDescriptor> deviceDesc =
- declaredDevices.getDeviceFromName(String8(devName));
- if (deviceDesc != 0) {
- add(deviceDesc);
- }
- }
- }
- devName = strtok(NULL, "|");
- }
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDevice(
- audio_devices_t type, String8 address) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mDeviceType == type) {
- if (address == "" || itemAt(i)->mAddress == address) {
- device = itemAt(i);
- if (itemAt(i)->mAddress == address) {
- break;
- }
- }
- }
- }
- ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
- type, address.string(), device.get());
- return device;
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromId(
- audio_port_handle_t id) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- ALOGV("DeviceVector::getDeviceFromId(%d) itemAt(%zu)->mId %d", id, i, itemAt(i)->mId);
- if (itemAt(i)->mId == id) {
- device = itemAt(i);
- break;
- }
- }
- return device;
-}
-
-AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromType(
- audio_devices_t type) const
-{
- DeviceVector devices;
- bool isOutput = audio_is_output_devices(type);
- type &= ~AUDIO_DEVICE_BIT_IN;
- for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
- bool curIsOutput = audio_is_output_devices(itemAt(i)->mDeviceType);
- audio_devices_t curType = itemAt(i)->mDeviceType & ~AUDIO_DEVICE_BIT_IN;
- if ((isOutput == curIsOutput) && ((type & curType) != 0)) {
- devices.add(itemAt(i));
- type &= ~curType;
- ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
- itemAt(i)->mDeviceType, itemAt(i).get());
- }
- }
- return devices;
-}
-
-AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromTypeAddr(
- audio_devices_t type, String8 address) const
-{
- DeviceVector devices;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mDeviceType == type) {
- if (itemAt(i)->mAddress == address) {
- devices.add(itemAt(i));
- }
- }
- }
- return devices;
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromName(
- const String8& name) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mName == name) {
- device = itemAt(i);
- break;
- }
- }
- return device;
-}
-
-void AudioPolicyManager::DeviceDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
-
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = audio_is_output_device(mDeviceType) ?
- AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
- dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
- dstConfig->ext.device.type = mDeviceType;
- dstConfig->ext.device.hw_module = mModule->mHandle;
- strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-void AudioPolicyManager::DeviceDescriptor::toAudioPort(struct audio_port *port) const
-{
- ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
- AudioPort::toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.device.type = mDeviceType;
- port->ext.device.hw_module = mModule->mHandle;
- strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-status_t AudioPolicyManager::DeviceDescriptor::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
- result.append(buffer);
- if (mId != 0) {
- snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
- result.append(buffer);
- }
- snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
- enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mDeviceType));
- result.append(buffer);
- if (mAddress.size() != 0) {
- snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
- AudioPort::dump(fd, spaces);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::AudioPatch::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
-
- snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
- result.append(buffer);
- for (size_t i = 0; i < mPatch.num_sources; i++) {
- if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
- snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
- mPatch.sources[i].id, enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mPatch.sources[i].ext.device.type));
- } else {
- snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
- mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
- }
- result.append(buffer);
- }
- snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
- result.append(buffer);
- for (size_t i = 0; i < mPatch.num_sinks; i++) {
- if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
- snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
- mPatch.sinks[i].id, enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mPatch.sinks[i].ext.device.type));
- } else {
- snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
- mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
- }
- result.append(buffer);
- }
-
- write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
-// --- audio_policy.conf file parsing
-
-uint32_t AudioPolicyManager::parseOutputFlagNames(char *name)
-{
- uint32_t flag = 0;
-
- // it is OK to cast name to non const here as we are not going to use it after
- // strtok() modifies it
- char *flagName = strtok(name, "|");
- while (flagName != NULL) {
- if (strlen(flagName) != 0) {
- flag |= stringToEnum(sOutputFlagNameToEnumTable,
- ARRAY_SIZE(sOutputFlagNameToEnumTable),
- flagName);
- }
- flagName = strtok(NULL, "|");
- }
- //force direct flag if offload flag is set: offloading implies a direct output stream
- // and all common behaviors are driven by checking only the direct flag
- // this should normally be set appropriately in the policy configuration file
- if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- flag |= AUDIO_OUTPUT_FLAG_DIRECT;
- }
-
- return flag;
-}
-
-uint32_t AudioPolicyManager::parseInputFlagNames(char *name)
-{
- uint32_t flag = 0;
-
- // it is OK to cast name to non const here as we are not going to use it after
- // strtok() modifies it
- char *flagName = strtok(name, "|");
- while (flagName != NULL) {
- if (strlen(flagName) != 0) {
- flag |= stringToEnum(sInputFlagNameToEnumTable,
- ARRAY_SIZE(sInputFlagNameToEnumTable),
- flagName);
- }
- flagName = strtok(NULL, "|");
- }
- return flag;
-}
-
-audio_devices_t AudioPolicyManager::parseDeviceNames(char *name)
-{
- uint32_t device = 0;
-
- char *devName = strtok(name, "|");
- while (devName != NULL) {
- if (strlen(devName) != 0) {
- device |= stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- devName);
- }
- devName = strtok(NULL, "|");
- }
- return device;
-}
-
-void AudioPolicyManager::loadHwModule(cnode *root)
-{
- status_t status = NAME_NOT_FOUND;
- cnode *node;
- sp<HwModule> module = new HwModule(root->name);
-
- node = config_find(root, DEVICES_TAG);
- if (node != NULL) {
- node = node->first_child;
- while (node) {
- ALOGV("loadHwModule() loading device %s", node->name);
- status_t tmpStatus = module->loadDevice(node);
- if (status == NAME_NOT_FOUND || status == NO_ERROR) {
- status = tmpStatus;
- }
- node = node->next;
- }
- }
- node = config_find(root, OUTPUTS_TAG);
- if (node != NULL) {
- node = node->first_child;
- while (node) {
- ALOGV("loadHwModule() loading output %s", node->name);
- status_t tmpStatus = module->loadOutput(node);
- if (status == NAME_NOT_FOUND || status == NO_ERROR) {
- status = tmpStatus;
- }
- node = node->next;
- }
- }
- node = config_find(root, INPUTS_TAG);
- if (node != NULL) {
- node = node->first_child;
- while (node) {
- ALOGV("loadHwModule() loading input %s", node->name);
- status_t tmpStatus = module->loadInput(node);
- if (status == NAME_NOT_FOUND || status == NO_ERROR) {
- status = tmpStatus;
- }
- node = node->next;
- }
- }
- loadGlobalConfig(root, module);
-
- if (status == NO_ERROR) {
- mHwModules.add(module);
- }
-}
-
-void AudioPolicyManager::loadHwModules(cnode *root)
-{
- cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
- if (node == NULL) {
- return;
- }
-
- node = node->first_child;
- while (node) {
- ALOGV("loadHwModules() loading module %s", node->name);
- loadHwModule(node);
- node = node->next;
- }
-}
-
-void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp<HwModule>& module)
-{
- cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
-
- if (node == NULL) {
- return;
- }
- DeviceVector declaredDevices;
- if (module != NULL) {
- declaredDevices = module->mDeclaredDevices;
- }
-
- node = node->first_child;
- while (node) {
- if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
- mAvailableOutputDevices.loadDevicesFromName((char *)node->value,
- declaredDevices);
- ALOGV("loadGlobalConfig() Attached Output Devices %08x",
- mAvailableOutputDevices.types());
- } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
- audio_devices_t device = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- (char *)node->value);
- if (device != AUDIO_DEVICE_NONE) {
- mDefaultOutputDevice = new DeviceDescriptor(String8(""), device);
- } else {
- ALOGW("loadGlobalConfig() default device not specified");
- }
- ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", mDefaultOutputDevice->mDeviceType);
- } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
- mAvailableInputDevices.loadDevicesFromName((char *)node->value,
- declaredDevices);
- ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types());
- } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
- mSpeakerDrcEnabled = stringToBool((char *)node->value);
- ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
- } else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) {
- uint32_t major, minor;
- sscanf((char *)node->value, "%u.%u", &major, &minor);
- module->mHalVersion = HARDWARE_DEVICE_API_VERSION(major, minor);
- ALOGV("loadGlobalConfig() mHalVersion = %04x major %u minor %u",
- module->mHalVersion, major, minor);
- }
- node = node->next;
- }
-}
-
-status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path)
-{
- cnode *root;
- char *data;
-
- data = (char *)load_file(path, NULL);
- if (data == NULL) {
- return -ENODEV;
- }
- root = config_node("", "");
- config_load(root, data);
-
- loadHwModules(root);
- // legacy audio_policy.conf files have one global_configuration section
- loadGlobalConfig(root, getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY));
- config_free(root);
- free(root);
- free(data);
-
- ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
-
- return NO_ERROR;
-}
-
-void AudioPolicyManager::defaultAudioPolicyConfig(void)
-{
- sp<HwModule> module;
- sp<IOProfile> profile;
- sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(String8(""),
- AUDIO_DEVICE_IN_BUILTIN_MIC);
- mAvailableOutputDevices.add(mDefaultOutputDevice);
- mAvailableInputDevices.add(defaultInputDevice);
-
- module = new HwModule("primary");
-
- profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE, module);
- profile->mSamplingRates.add(44100);
- profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
- profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
- profile->mSupportedDevices.add(mDefaultOutputDevice);
- profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
- module->mOutputProfiles.add(profile);
-
- profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK, module);
- profile->mSamplingRates.add(8000);
- profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
- profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
- profile->mSupportedDevices.add(defaultInputDevice);
- module->mInputProfiles.add(profile);
-
- mHwModules.add(module);
-}
-
-audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr)
-{
- // flags to stream type mapping
- if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
- return AUDIO_STREAM_ENFORCED_AUDIBLE;
- }
- if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
- return AUDIO_STREAM_BLUETOOTH_SCO;
- }
- if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) {
- return AUDIO_STREAM_TTS;
- }
-
- // usage to stream type mapping
- switch (attr->usage) {
- case AUDIO_USAGE_MEDIA:
- case AUDIO_USAGE_GAME:
- case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
- return AUDIO_STREAM_MUSIC;
- case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
- if (isStreamActive(AUDIO_STREAM_ALARM)) {
- return AUDIO_STREAM_ALARM;
- }
- if (isStreamActive(AUDIO_STREAM_RING)) {
- return AUDIO_STREAM_RING;
- }
- if (isInCall()) {
- return AUDIO_STREAM_VOICE_CALL;
- }
- return AUDIO_STREAM_ACCESSIBILITY;
- case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
- return AUDIO_STREAM_SYSTEM;
- case AUDIO_USAGE_VOICE_COMMUNICATION:
- return AUDIO_STREAM_VOICE_CALL;
-
- case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
- return AUDIO_STREAM_DTMF;
-
- case AUDIO_USAGE_ALARM:
- return AUDIO_STREAM_ALARM;
- case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
- return AUDIO_STREAM_RING;
-
- case AUDIO_USAGE_NOTIFICATION:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
- case AUDIO_USAGE_NOTIFICATION_EVENT:
- return AUDIO_STREAM_NOTIFICATION;
-
- case AUDIO_USAGE_UNKNOWN:
- default:
- return AUDIO_STREAM_MUSIC;
- }
-}
-
-bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) {
- // has flags that map to a strategy?
- if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) {
- return true;
- }
-
- // has known usage?
- switch (paa->usage) {
- case AUDIO_USAGE_UNKNOWN:
- case AUDIO_USAGE_MEDIA:
- case AUDIO_USAGE_VOICE_COMMUNICATION:
- case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
- case AUDIO_USAGE_ALARM:
- case AUDIO_USAGE_NOTIFICATION:
- case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
- case AUDIO_USAGE_NOTIFICATION_EVENT:
- case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
- case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
- case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
- case AUDIO_USAGE_GAME:
- case AUDIO_USAGE_VIRTUAL_SOURCE:
- break;
- default:
- return false;
- }
- return true;
-}
-
-}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
deleted file mode 100644
index cbdafa6..0000000
--- a/services/audiopolicy/AudioPolicyManager.h
+++ /dev/null
@@ -1,937 +0,0 @@
-/*
- * 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.
- */
-
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <cutils/config_utils.h>
-#include <cutils/misc.h>
-#include <utils/Timers.h>
-#include <utils/Errors.h>
-#include <utils/KeyedVector.h>
-#include <utils/SortedVector.h>
-#include <media/AudioPolicy.h>
-#include "AudioPolicyInterface.h"
-
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
-#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
-// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
-#define SONIFICATION_HEADSET_VOLUME_MIN 0.016
-// Time in milliseconds during which we consider that music is still active after a music
-// track was stopped - see computeVolume()
-#define SONIFICATION_HEADSET_MUSIC_DELAY 5000
-// Time in milliseconds after media stopped playing during which we consider that the
-// sonification should be as unobtrusive as during the time media was playing.
-#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
-// Time in milliseconds during witch some streams are muted while the audio path
-// is switched
-#define MUTE_TIME_MS 2000
-
-#define NUM_TEST_OUTPUTS 5
-
-#define NUM_VOL_CURVE_KNEES 2
-
-// Default minimum length allowed for offloading a compressed track
-// Can be overridden by the audio.offload.min.duration.secs property
-#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
-
-#define MAX_MIXER_SAMPLING_RATE 48000
-#define MAX_MIXER_CHANNEL_COUNT 8
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManager implements audio policy manager behavior common to all platforms.
-// ----------------------------------------------------------------------------
-
-class AudioPolicyManager: public AudioPolicyInterface
-#ifdef AUDIO_POLICY_TEST
- , public Thread
-#endif //AUDIO_POLICY_TEST
-{
-
-public:
- AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
- virtual ~AudioPolicyManager();
-
- // AudioPolicyInterface
- virtual status_t setDeviceConnectionState(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address);
- virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
- const char *device_address);
- virtual void setPhoneState(audio_mode_t state);
- virtual void setForceUse(audio_policy_force_use_t usage,
- audio_policy_forced_cfg_t config);
- virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
- virtual void setSystemProperty(const char* property, const char* value);
- virtual status_t initCheck();
- 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 status_t getOutputForAttr(const audio_attributes_t *attr,
- audio_io_handle_t *output,
- audio_session_t session,
- 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 status_t startOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- virtual status_t stopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- virtual void releaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- virtual status_t getInputForAttr(const audio_attributes_t *attr,
- audio_io_handle_t *input,
- audio_session_t session,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_input_flags_t flags,
- input_type_t *inputType);
-
- // indicates to the audio policy manager that the input starts being used.
- virtual status_t startInput(audio_io_handle_t input,
- audio_session_t session);
-
- // indicates to the audio policy manager that the input stops being used.
- virtual status_t stopInput(audio_io_handle_t input,
- audio_session_t session);
- virtual void releaseInput(audio_io_handle_t input,
- audio_session_t session);
- virtual void closeAllInputs();
- virtual void initStreamVolume(audio_stream_type_t stream,
- int indexMin,
- int indexMax);
- virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
- int index,
- audio_devices_t device);
- virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
- int *index,
- audio_devices_t device);
-
- // return the strategy corresponding to a given stream type
- virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
- // return the strategy corresponding to the given audio attributes
- virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr);
-
- // return the enabled output devices for the given stream type
- virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
-
- virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
- virtual status_t registerEffect(const effect_descriptor_t *desc,
- audio_io_handle_t io,
- uint32_t strategy,
- int session,
- int id);
- virtual status_t unregisterEffect(int id);
- virtual status_t setEffectEnabled(int id, bool enabled);
-
- virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
- // return whether a stream is playing remotely, override to change the definition of
- // local/remote playback, used for instance by notification manager to not make
- // media players lose audio focus when not playing locally
- // For the base implementation, "remotely" means playing during screen mirroring which
- // uses an output for playback with a non-empty, non "0" address.
- virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
- virtual bool isSourceActive(audio_source_t source) const;
-
- virtual status_t dump(int fd);
-
- virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
-
- virtual status_t listAudioPorts(audio_port_role_t role,
- audio_port_type_t type,
- unsigned int *num_ports,
- struct audio_port *ports,
- unsigned int *generation);
- virtual status_t getAudioPort(struct audio_port *port);
- virtual status_t createAudioPatch(const struct audio_patch *patch,
- audio_patch_handle_t *handle,
- uid_t uid);
- virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
- uid_t uid);
- virtual status_t listAudioPatches(unsigned int *num_patches,
- struct audio_patch *patches,
- unsigned int *generation);
- virtual status_t setAudioPortConfig(const struct audio_port_config *config);
- virtual void clearAudioPatches(uid_t uid);
-
- virtual status_t acquireSoundTriggerSession(audio_session_t *session,
- audio_io_handle_t *ioHandle,
- audio_devices_t *device);
-
- virtual status_t releaseSoundTriggerSession(audio_session_t session);
-
- virtual status_t registerPolicyMixes(Vector<AudioMix> mixes);
- virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
-
-protected:
-
- enum routing_strategy {
- STRATEGY_MEDIA,
- STRATEGY_PHONE,
- STRATEGY_SONIFICATION,
- STRATEGY_SONIFICATION_RESPECTFUL,
- STRATEGY_DTMF,
- STRATEGY_ENFORCED_AUDIBLE,
- STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
- STRATEGY_ACCESSIBILITY,
- STRATEGY_REROUTING,
- NUM_STRATEGIES
- };
-
- // 4 points to define the volume attenuation curve, each characterized by the volume
- // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
- // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
-
- enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
-
- class VolumeCurvePoint
- {
- public:
- int mIndex;
- float mDBAttenuation;
- };
-
- // device categories used for volume curve management.
- enum device_category {
- DEVICE_CATEGORY_HEADSET,
- DEVICE_CATEGORY_SPEAKER,
- DEVICE_CATEGORY_EARPIECE,
- DEVICE_CATEGORY_EXT_MEDIA,
- DEVICE_CATEGORY_CNT
- };
-
- class HwModule;
-
- class AudioGain: public RefBase
- {
- public:
- AudioGain(int index, bool useInChannelMask);
- virtual ~AudioGain() {}
-
- void dump(int fd, int spaces, int index) const;
-
- void getDefaultConfig(struct audio_gain_config *config);
- status_t checkConfig(const struct audio_gain_config *config);
- int mIndex;
- struct audio_gain mGain;
- bool mUseInChannelMask;
- };
-
- class AudioPort: public virtual RefBase
- {
- public:
- AudioPort(const String8& name, audio_port_type_t type,
- audio_port_role_t role, const sp<HwModule>& module);
- virtual ~AudioPort() {}
-
- virtual void toAudioPort(struct audio_port *port) const;
-
- void importAudioPort(const sp<AudioPort> port);
- void clearCapabilities();
-
- void loadSamplingRates(char *name);
- void loadFormats(char *name);
- void loadOutChannels(char *name);
- void loadInChannels(char *name);
-
- audio_gain_mode_t loadGainMode(char *name);
- void loadGain(cnode *root, int index);
- virtual void loadGains(cnode *root);
-
- // searches for an exact match
- status_t checkExactSamplingRate(uint32_t samplingRate) const;
- // searches for a compatible match, and returns the best match via updatedSamplingRate
- status_t checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t *updatedSamplingRate) const;
- // searches for an exact match
- status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
- // searches for a compatible match, currently implemented for input channel masks only
- status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
- status_t checkFormat(audio_format_t format) const;
- status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
-
- uint32_t pickSamplingRate() const;
- audio_channel_mask_t pickChannelMask() const;
- audio_format_t pickFormat() const;
-
- static const audio_format_t sPcmFormatCompareTable[];
- static int compareFormats(audio_format_t format1, audio_format_t format2);
-
- void dump(int fd, int spaces) const;
-
- String8 mName;
- audio_port_type_t mType;
- audio_port_role_t mRole;
- bool mUseInChannelMask;
- // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
- // indicates the supported parameters should be read from the output stream
- // after it is opened for the first time
- Vector <uint32_t> mSamplingRates; // supported sampling rates
- Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
- Vector <audio_format_t> mFormats; // supported audio formats
- Vector < sp<AudioGain> > mGains; // gain controllers
- sp<HwModule> mModule; // audio HW module exposing this I/O stream
- uint32_t mFlags; // attribute flags (e.g primary output,
- // direct output...).
- };
-
- class AudioPortConfig: public virtual RefBase
- {
- public:
- AudioPortConfig();
- virtual ~AudioPortConfig() {}
-
- status_t applyAudioPortConfig(const struct audio_port_config *config,
- struct audio_port_config *backupConfig = NULL);
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const = 0;
- virtual sp<AudioPort> getAudioPort() const = 0;
- uint32_t mSamplingRate;
- audio_format_t mFormat;
- audio_channel_mask_t mChannelMask;
- struct audio_gain_config mGain;
- };
-
-
- class AudioPatch: public RefBase
- {
- public:
- AudioPatch(audio_patch_handle_t handle,
- const struct audio_patch *patch, uid_t uid) :
- mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0) {}
-
- status_t dump(int fd, int spaces, int index) const;
-
- audio_patch_handle_t mHandle;
- struct audio_patch mPatch;
- uid_t mUid;
- audio_patch_handle_t mAfPatchHandle;
- };
-
- class DeviceDescriptor: public AudioPort, public AudioPortConfig
- {
- public:
- DeviceDescriptor(const String8& name, audio_devices_t type);
-
- virtual ~DeviceDescriptor() {}
-
- bool equals(const sp<DeviceDescriptor>& other) const;
-
- // AudioPortConfig
- virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
-
- // AudioPort
- virtual void loadGains(cnode *root);
- virtual void toAudioPort(struct audio_port *port) const;
-
- status_t dump(int fd, int spaces, int index) const;
-
- audio_devices_t mDeviceType;
- String8 mAddress;
- audio_port_handle_t mId;
- };
-
- class DeviceVector : public SortedVector< sp<DeviceDescriptor> >
- {
- public:
- DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
-
- ssize_t add(const sp<DeviceDescriptor>& item);
- ssize_t remove(const sp<DeviceDescriptor>& item);
- ssize_t indexOf(const sp<DeviceDescriptor>& item) const;
-
- audio_devices_t types() const { return mDeviceTypes; }
-
- void loadDevicesFromType(audio_devices_t types);
- void loadDevicesFromName(char *name, const DeviceVector& declaredDevices);
-
- sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const;
- DeviceVector getDevicesFromType(audio_devices_t types) const;
- sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
- sp<DeviceDescriptor> getDeviceFromName(const String8& name) const;
- DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address)
- const;
-
- private:
- void refreshTypes();
- audio_devices_t mDeviceTypes;
- };
-
- // the IOProfile class describes the capabilities of an output or input stream.
- // It is currently assumed that all combination of listed parameters are supported.
- // It is used by the policy manager to determine if an output or input is suitable for
- // a given use case, open/close it accordingly and connect/disconnect audio tracks
- // to/from it.
- class IOProfile : public AudioPort
- {
- public:
- IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module);
- virtual ~IOProfile();
-
- // This method is used for both output and input.
- // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
- // For input, flags is interpreted as audio_input_flags_t.
- // TODO: merge audio_output_flags_t and audio_input_flags_t.
- bool isCompatibleProfile(audio_devices_t device,
- String8 address,
- uint32_t samplingRate,
- uint32_t *updatedSamplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- uint32_t flags) const;
-
- void dump(int fd);
- void log();
-
- DeviceVector mSupportedDevices; // supported devices
- // (devices this output can be routed to)
- };
-
- class HwModule : public RefBase
- {
- public:
- HwModule(const char *name);
- ~HwModule();
-
- status_t loadOutput(cnode *root);
- status_t loadInput(cnode *root);
- status_t loadDevice(cnode *root);
-
- status_t addOutputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address);
- status_t removeOutputProfile(String8 name);
- status_t addInputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address);
- status_t removeInputProfile(String8 name);
-
- void dump(int fd);
-
- const char *const mName; // base name of the audio HW module (primary, a2dp ...)
- uint32_t mHalVersion; // audio HAL API version
- audio_module_handle_t mHandle;
- Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
- Vector < sp<IOProfile> > mInputProfiles; // input profiles exposed by this module
- DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf
-
- };
-
- // default volume curve
- static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManager::VOLCNT];
- // default volume curve for media strategy
- static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT];
- // volume curve for non-media audio on ext media outputs (HDMI, Line, etc)
- static const VolumeCurvePoint sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT];
- // volume curve for media strategy on speakers
- static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT];
- // volume curve for sonification strategy on speakers
- static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sLinearVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSilentVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sFullScaleVolumeCurve[AudioPolicyManager::VOLCNT];
- // default volume curves per stream and device category. See initializeVolumeCurves()
- static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT];
-
- // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
- // and keep track of the usage of this output by each audio stream type.
- class AudioOutputDescriptor: public AudioPortConfig
- {
- public:
- AudioOutputDescriptor(const sp<IOProfile>& profile);
-
- status_t dump(int fd);
-
- audio_devices_t device() const;
- void changeRefCount(audio_stream_type_t stream, int delta);
-
- bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
- audio_devices_t supportedDevices();
- uint32_t latency();
- bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
- bool isActive(uint32_t inPastMs = 0) const;
- bool isStreamActive(audio_stream_type_t stream,
- uint32_t inPastMs = 0,
- nsecs_t sysTime = 0) const;
- bool isStrategyActive(routing_strategy strategy,
- uint32_t inPastMs = 0,
- nsecs_t sysTime = 0) const;
-
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
- virtual sp<AudioPort> getAudioPort() const { return mProfile; }
- void toAudioPort(struct audio_port *port) const;
-
- audio_port_handle_t mId;
- audio_io_handle_t mIoHandle; // output handle
- uint32_t mLatency; //
- audio_output_flags_t mFlags; //
- audio_devices_t mDevice; // current device this output is routed to
- AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
- audio_patch_handle_t mPatchHandle;
- uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
- nsecs_t mStopTime[AUDIO_STREAM_CNT];
- sp<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output
- sp<AudioOutputDescriptor> mOutput2; // used by duplicated outputs: second output
- float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume
- int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter
- const sp<IOProfile> mProfile; // I/O profile this output derives from
- bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
- // device selection. See checkDeviceMuteStrategies()
- uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
- };
-
- // descriptor for audio inputs. Used to maintain current configuration of each opened audio input
- // and keep track of the usage of this input.
- class AudioInputDescriptor: public AudioPortConfig
- {
- public:
- AudioInputDescriptor(const sp<IOProfile>& profile);
-
- status_t dump(int fd);
-
- audio_port_handle_t mId;
- audio_io_handle_t mIoHandle; // input handle
- audio_devices_t mDevice; // current device this input is routed to
- AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
- audio_patch_handle_t mPatchHandle;
- uint32_t mRefCount; // number of AudioRecord clients using
- // this input
- uint32_t mOpenRefCount;
- audio_source_t mInputSource; // input source selected by application
- //(mediarecorder.h)
- const sp<IOProfile> mProfile; // I/O profile this output derives from
- SortedVector<audio_session_t> mSessions; // audio sessions attached to this input
- bool mIsSoundTrigger; // used by a soundtrigger capture
-
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
- virtual sp<AudioPort> getAudioPort() const { return mProfile; }
- void toAudioPort(struct audio_port *port) const;
- };
-
- // stream descriptor used for volume control
- class StreamDescriptor
- {
- public:
- StreamDescriptor();
-
- int getVolumeIndex(audio_devices_t device);
- void dump(int fd);
-
- int mIndexMin; // min volume index
- int mIndexMax; // max volume index
- KeyedVector<audio_devices_t, int> mIndexCur; // current volume index per device
- bool mCanBeMuted; // true is the stream can be muted
-
- const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT];
- };
-
- // stream descriptor used for volume control
- class EffectDescriptor : public RefBase
- {
- public:
-
- status_t dump(int fd);
-
- int mIo; // io the effect is attached to
- routing_strategy mStrategy; // routing strategy the effect is associated to
- int mSession; // audio session the effect is on
- effect_descriptor_t mDesc; // effect descriptor
- bool mEnabled; // enabled state: CPU load being used or not
- };
-
- void addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc);
- void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc);
-
- // return the strategy corresponding to a given stream type
- static routing_strategy getStrategy(audio_stream_type_t stream);
-
- // return appropriate device for streams handled by the specified strategy according to current
- // phone state, connected devices...
- // if fromCache is true, the device is returned from mDeviceForStrategy[],
- // otherwise it is determine by current state
- // (device connected,phone state, force use, a2dp output...)
- // This allows to:
- // 1 speed up process when the state is stable (when starting or stopping an output)
- // 2 access to either current device selection (fromCache == true) or
- // "future" device selection (fromCache == false) when called from a context
- // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
- // before updateDevicesAndOutputs() is called.
- virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
- bool fromCache);
-
- // change the route of the specified output. Returns the number of ms we have slept to
- // allow new routing to take effect in certain cases.
- virtual uint32_t setOutputDevice(audio_io_handle_t output,
- audio_devices_t device,
- bool force = false,
- int delayMs = 0,
- audio_patch_handle_t *patchHandle = NULL,
- const char* address = NULL);
- status_t resetOutputDevice(audio_io_handle_t output,
- int delayMs = 0,
- audio_patch_handle_t *patchHandle = NULL);
- status_t setInputDevice(audio_io_handle_t input,
- audio_devices_t device,
- bool force = false,
- audio_patch_handle_t *patchHandle = NULL);
- status_t resetInputDevice(audio_io_handle_t input,
- audio_patch_handle_t *patchHandle = NULL);
-
- // select input device corresponding to requested audio source
- virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
-
- // return io handle of active input or 0 if no input is active
- // Only considers inputs from physical devices (e.g. main mic, headset mic) when
- // ignoreVirtualInputs is true.
- audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
-
- uint32_t activeInputsCount() const;
-
- // initialize volume curves for each strategy and device category
- void initializeVolumeCurves();
-
- // compute the actual volume for a given stream according to the requested index and a particular
- // device
- virtual float computeVolume(audio_stream_type_t stream, int index,
- audio_io_handle_t output, audio_devices_t device);
-
- // check that volume change is permitted, compute and send new volume to audio hardware
- virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index,
- audio_io_handle_t output,
- audio_devices_t device,
- int delayMs = 0, bool force = false);
-
- // apply all stream volumes to the specified output and device
- void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
-
- // Mute or unmute all streams handled by the specified strategy on the specified output
- void setStrategyMute(routing_strategy strategy,
- bool on,
- audio_io_handle_t output,
- int delayMs = 0,
- audio_devices_t device = (audio_devices_t)0);
-
- // Mute or unmute the stream on the specified output
- void setStreamMute(audio_stream_type_t stream,
- bool on,
- audio_io_handle_t output,
- int delayMs = 0,
- audio_devices_t device = (audio_devices_t)0);
-
- // handle special cases for sonification strategy while in call: mute streams or replace by
- // a special tone in the device used for communication
- void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange);
-
- // true if device is in a telephony or VoIP call
- virtual bool isInCall();
-
- // true if given state represents a device in a telephony or VoIP call
- virtual bool isStateInCall(int state);
-
- // when a device is connected, checks if an open output can be routed
- // to this device. If none is open, tries to open one of the available outputs.
- // Returns an output suitable to this device or 0.
- // when a device is disconnected, checks if an output is not used any more and
- // returns its handle if any.
- // transfers the audio tracks and effects from one output thread to another accordingly.
- status_t checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
- audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& outputs,
- const String8 address);
-
- status_t checkInputsForDevice(audio_devices_t device,
- audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& inputs,
- const String8 address);
-
- // close an output and its companion duplicating output.
- void closeOutput(audio_io_handle_t output);
-
- // close an input.
- void closeInput(audio_io_handle_t input);
-
- // checks and if necessary changes outputs used for all strategies.
- // must be called every time a condition that affects the output choice for a given strategy
- // changes: connected device, phone state, force use...
- // Must be called before updateDevicesAndOutputs()
- void checkOutputForStrategy(routing_strategy strategy);
-
- // Same as checkOutputForStrategy() but for a all strategies in order of priority
- void checkOutputForAllStrategies();
-
- // manages A2DP output suspend/restore according to phone state and BT SCO usage
- void checkA2dpSuspend();
-
- // returns the A2DP output handle if it is open or 0 otherwise
- audio_io_handle_t getA2dpOutput();
-
- // selects the most appropriate device on output for current state
- // must be called every time a condition that affects the device choice for a given output is
- // changed: connected device, phone state, force use, output start, output stop..
- // see getDeviceForStrategy() for the use of fromCache parameter
- audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache);
-
- // updates cache of device used by all strategies (mDeviceForStrategy[])
- // must be called every time a condition that affects the device choice for a given strategy is
- // changed: connected device, phone state, force use...
- // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
- // Must be called after checkOutputForAllStrategies()
- void updateDevicesAndOutputs();
-
- // selects the most appropriate device on input for current state
- audio_devices_t getNewInputDevice(audio_io_handle_t input);
-
- virtual uint32_t getMaxEffectsCpuLoad();
- virtual uint32_t getMaxEffectsMemory();
-#ifdef AUDIO_POLICY_TEST
- virtual bool threadLoop();
- void exit();
- int testOutputIndex(audio_io_handle_t output);
-#endif //AUDIO_POLICY_TEST
-
- status_t setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled);
-
- // returns the category the device belongs to with regard to volume curve management
- static device_category getDeviceCategory(audio_devices_t device);
-
- // extract one device relevant for volume control from multiple device selection
- static audio_devices_t getDeviceForVolume(audio_devices_t device);
-
- SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
- DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs);
- bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
- SortedVector<audio_io_handle_t>& outputs2);
-
- // mute/unmute strategies using an incompatible device combination
- // if muting, wait for the audio in pcm buffer to be drained before proceeding
- // if unmuting, unmute only after the specified delay
- // Returns the number of ms waited
- virtual uint32_t checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
- audio_devices_t prevDevice,
- uint32_t delayMs);
-
- audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
- audio_output_flags_t flags,
- audio_format_t format);
- // samplingRate parameter is an in/out and so may be modified
- sp<IOProfile> getInputProfile(audio_devices_t device,
- String8 address,
- uint32_t& samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_input_flags_t flags);
- sp<IOProfile> getProfileForDirectOutput(audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags);
-
- audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
-
- bool isNonOffloadableEffectEnabled();
-
- virtual status_t addAudioPatch(audio_patch_handle_t handle,
- const sp<AudioPatch>& patch);
- virtual status_t removeAudioPatch(audio_patch_handle_t handle);
-
- sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
- sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
- sp<HwModule> getModuleForDevice(audio_devices_t device) const;
- sp<HwModule> getModuleFromName(const char *name) const;
- audio_devices_t availablePrimaryOutputDevices();
- audio_devices_t availablePrimaryInputDevices();
-
- void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0);
-
- //
- // Audio policy configuration file parsing (audio_policy.conf)
- //
- static uint32_t stringToEnum(const struct StringToEnum *table,
- size_t size,
- const char *name);
- static const char *enumToString(const struct StringToEnum *table,
- size_t size,
- uint32_t value);
- static bool stringToBool(const char *value);
- static uint32_t parseOutputFlagNames(char *name);
- static uint32_t parseInputFlagNames(char *name);
- static audio_devices_t parseDeviceNames(char *name);
- void loadHwModule(cnode *root);
- void loadHwModules(cnode *root);
- void loadGlobalConfig(cnode *root, const sp<HwModule>& module);
- status_t loadAudioPolicyConfig(const char *path);
- void defaultAudioPolicyConfig(void);
-
-
- uid_t mUidCached;
- AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
- audio_io_handle_t mPrimaryOutput; // primary output handle
- // list of descriptors for outputs currently opened
- DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mOutputs;
- // copy of mOutputs before setDeviceConnectionState() opens new outputs
- // reset to mOutputs when updateDevicesAndOutputs() is called.
- DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mPreviousOutputs;
- DefaultKeyedVector<audio_io_handle_t, sp<AudioInputDescriptor> > mInputs; // list of input descriptors
- DeviceVector mAvailableOutputDevices; // all available output devices
- DeviceVector mAvailableInputDevices; // all available input devices
- int mPhoneState; // current phone state
- audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; // current forced use configuration
-
- StreamDescriptor mStreams[AUDIO_STREAM_CNT]; // stream descriptors for volume control
- bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected
- audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
- float mLastVoiceVolume; // last voice volume value sent to audio HAL
-
- // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
- static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
- // Maximum memory allocated to audio effects in KB
- static const uint32_t MAX_EFFECTS_MEMORY = 512;
- uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
- uint32_t mTotalEffectsMemory; // current memory used by effects
- KeyedVector<int, sp<EffectDescriptor> > mEffects; // list of registered audio effects
- bool mA2dpSuspended; // true if A2DP output is suspended
- sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
- bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
- // to boost soft sounds, used to adjust volume curves accordingly
-
- Vector < sp<HwModule> > mHwModules;
- volatile int32_t mNextUniqueId;
- volatile int32_t mAudioPortGeneration;
-
- DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches;
-
- DefaultKeyedVector<audio_session_t, audio_io_handle_t> mSoundTriggerSessions;
-
- sp<AudioPatch> mCallTxPatch;
- sp<AudioPatch> mCallRxPatch;
-
- // for supporting "beacon" streams, i.e. streams that only play on speaker, and never
- // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing
- enum {
- STARTING_OUTPUT,
- STARTING_BEACON,
- STOPPING_OUTPUT,
- STOPPING_BEACON
- };
- uint32_t mBeaconMuteRefCount; // ref count for stream that would mute beacon
- uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams
- bool mBeaconMuted; // has STREAM_TTS been muted
-
- // custom mix entry in mPolicyMixes
- class AudioPolicyMix : public RefBase {
- public:
- AudioPolicyMix() {}
-
- AudioMix mMix; // Audio policy mix descriptor
- sp<AudioOutputDescriptor> mOutput; // Corresponding output stream
- };
- DefaultKeyedVector<String8, sp<AudioPolicyMix> > mPolicyMixes; // list of registered mixes
-
-
-#ifdef AUDIO_POLICY_TEST
- Mutex mLock;
- Condition mWaitWorkCV;
-
- int mCurOutput;
- bool mDirectOutput;
- audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
- int mTestInput;
- uint32_t mTestDevice;
- uint32_t mTestSamplingRate;
- uint32_t mTestFormat;
- uint32_t mTestChannels;
- uint32_t mTestLatencyMs;
-#endif //AUDIO_POLICY_TEST
- static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
- int indexInUi);
- static bool isVirtualInputDevice(audio_devices_t device);
- uint32_t nextUniqueId();
- uint32_t nextAudioPortGeneration();
-private:
- // updates device caching and output for streams that can influence the
- // routing of notifications
- void handleNotificationRoutingForStream(audio_stream_type_t stream);
- static bool deviceDistinguishesOnAddress(audio_devices_t device);
- // find the outputs on a given output descriptor that have the given address.
- // to be called on an AudioOutputDescriptor whose supported devices (as defined
- // in mProfile->mSupportedDevices) matches the device whose address is to be matched.
- // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
- // where addresses are used to distinguish between one connected device and another.
- void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
- const audio_devices_t device /*in*/,
- const String8 address /*in*/,
- SortedVector<audio_io_handle_t>& outputs /*out*/);
- uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
- // internal method to return the output handle for the given device and format
- audio_io_handle_t getOutputForDevice(
- audio_devices_t device,
- audio_session_t session,
- 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);
- // internal function to derive a stream type value from audio attributes
- audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
- // return true if any output is playing anything besides the stream to ignore
- bool isAnyOutputActive(audio_stream_type_t streamToIgnore);
- // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
- // returns 0 if no mute/unmute event happened, the largest latency of the device where
- // the mute/unmute happened
- uint32_t handleEventForBeacon(int event);
- uint32_t setBeaconMute(bool mute);
- bool isValidAttributes(const audio_attributes_t *paa);
-
- // select input device corresponding to requested audio source and return associated policy
- // mix if any. Calls getDeviceForInputSource().
- audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
- AudioMix **policyMix = NULL);
-
- // Called by setDeviceConnectionState().
- status_t setDeviceConnectionStateInt(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address);
- sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device,
- const char *device_address);
-
-};
-
-};
diff --git a/services/audiopolicy/common/Android.mk b/services/audiopolicy/common/Android.mk
new file mode 100644
index 0000000..dcce8e3
--- /dev/null
+++ b/services/audiopolicy/common/Android.mk
@@ -0,0 +1,9 @@
+
+LOCAL_PATH := $(call my-dir)
+include $(CLEAR_VARS)
+
+#######################################################################
+# Recursive call sub-folder Android.mk
+#
+include $(call all-makefiles-under,$(LOCAL_PATH))
+
diff --git a/services/audiopolicy/common/include/RoutingStrategy.h b/services/audiopolicy/common/include/RoutingStrategy.h
new file mode 100644
index 0000000..d38967e
--- /dev/null
+++ b/services/audiopolicy/common/include/RoutingStrategy.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+namespace android {
+
+// Time in milliseconds after media stopped playing during which we consider that the
+// sonification should be as unobtrusive as during the time media was playing.
+#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
+
+enum routing_strategy {
+ STRATEGY_MEDIA,
+ STRATEGY_PHONE,
+ STRATEGY_SONIFICATION,
+ STRATEGY_SONIFICATION_RESPECTFUL,
+ STRATEGY_DTMF,
+ STRATEGY_ENFORCED_AUDIBLE,
+ STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
+ STRATEGY_ACCESSIBILITY,
+ STRATEGY_REROUTING,
+ NUM_STRATEGIES
+};
+
+}; //namespace android
diff --git a/services/audiopolicy/common/include/Volume.h b/services/audiopolicy/common/include/Volume.h
new file mode 100755
index 0000000..a4cc759
--- /dev/null
+++ b/services/audiopolicy/common/include/Volume.h
@@ -0,0 +1,137 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <system/audio.h>
+#include <utils/Log.h>
+
+class VolumeCurvePoint
+{
+public:
+ int mIndex;
+ float mDBAttenuation;
+};
+
+class Volume
+{
+public:
+ /**
+ * 4 points to define the volume attenuation curve, each characterized by the volume
+ * index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
+ * we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
+ *
+ * @todo shall become configurable
+ */
+ enum {
+ VOLMIN = 0,
+ VOLKNEE1 = 1,
+ VOLKNEE2 = 2,
+ VOLMAX = 3,
+
+ VOLCNT = 4
+ };
+
+ /**
+ * device categories used for volume curve management.
+ */
+ enum device_category {
+ DEVICE_CATEGORY_HEADSET,
+ DEVICE_CATEGORY_SPEAKER,
+ DEVICE_CATEGORY_EARPIECE,
+ DEVICE_CATEGORY_EXT_MEDIA,
+ DEVICE_CATEGORY_CNT
+ };
+
+ /**
+ * extract one device relevant for volume control from multiple device selection
+ *
+ * @param[in] device for which the volume category is associated
+ *
+ * @return subset of device required to limit the number of volume category per device
+ */
+ static audio_devices_t getDeviceForVolume(audio_devices_t device)
+ {
+ if (device == AUDIO_DEVICE_NONE) {
+ // this happens when forcing a route update and no track is active on an output.
+ // In this case the returned category is not important.
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (popcount(device) > 1) {
+ // Multiple device selection is either:
+ // - speaker + one other device: give priority to speaker in this case.
+ // - one A2DP device + another device: happens with duplicated output. In this case
+ // retain the device on the A2DP output as the other must not correspond to an active
+ // selection if not the speaker.
+ // - HDMI-CEC system audio mode only output: give priority to available item in order.
+ if (device & AUDIO_DEVICE_OUT_SPEAKER) {
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
+ device = AUDIO_DEVICE_OUT_HDMI_ARC;
+ } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
+ device = AUDIO_DEVICE_OUT_AUX_LINE;
+ } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
+ device = AUDIO_DEVICE_OUT_SPDIF;
+ } else {
+ device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
+ }
+ }
+
+ /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
+ if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+
+ ALOGW_IF(popcount(device) != 1,
+ "getDeviceForVolume() invalid device combination: %08x",
+ device);
+
+ return device;
+ }
+
+ /**
+ * returns the category the device belongs to with regard to volume curve management
+ *
+ * @param[in] device to check upon the category to whom it belongs to.
+ *
+ * @return device category.
+ */
+ static device_category getDeviceCategory(audio_devices_t device)
+ {
+ switch(getDeviceForVolume(device)) {
+ case AUDIO_DEVICE_OUT_EARPIECE:
+ return DEVICE_CATEGORY_EARPIECE;
+ case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+ case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+ return DEVICE_CATEGORY_HEADSET;
+ case AUDIO_DEVICE_OUT_LINE:
+ case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+ /*USB? Remote submix?*/
+ return DEVICE_CATEGORY_EXT_MEDIA;
+ case AUDIO_DEVICE_OUT_SPEAKER:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+ case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+ case AUDIO_DEVICE_OUT_USB_DEVICE:
+ case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+ default:
+ return DEVICE_CATEGORY_SPEAKER;
+ }
+ }
+
+};
diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h
new file mode 100755
index 0000000..a2327ee
--- /dev/null
+++ b/services/audiopolicy/common/include/policy.h
@@ -0,0 +1,84 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <system/audio.h>
+
+// For mixed output and inputs, the policy will use max mixer sampling rates.
+// Do not limit sampling rate otherwise
+#define MAX_MIXER_SAMPLING_RATE 48000
+
+// For mixed output and inputs, the policy will use max mixer channel count.
+// Do not limit channel count otherwise
+#define MAX_MIXER_CHANNEL_COUNT 8
+
+/**
+ * A device mask for all audio input devices that are considered "virtual" when evaluating
+ * active inputs in getActiveInput()
+ */
+#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_FM_TUNER)
+
+
+/**
+ * A device mask for all audio input and output devices where matching inputs/outputs on device
+ * type alone is not enough: the address must match too
+ */
+#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
+
+/**
+ * Check if the state given correspond to an in call state.
+ * @TODO find a better name for widely call state
+ *
+ * @param[in] state to consider
+ *
+ * @return true if given state represents a device in a telephony or VoIP call
+ */
+static inline bool is_state_in_call(int state)
+{
+ return (state == AUDIO_MODE_IN_CALL) || (state == AUDIO_MODE_IN_COMMUNICATION);
+}
+
+/**
+ * Check if the input device given is considered as a virtual device.
+ *
+ * @param[in] device to consider
+ *
+ * @return true if the device is a virtual one, false otherwise.
+ */
+static bool is_virtual_input_device(audio_devices_t device)
+{
+ if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
+ device &= ~AUDIO_DEVICE_BIT_IN;
+ if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0))
+ return true;
+ }
+ return false;
+}
+
+/**
+ * Check whether the device type is one
+ * where addresses are used to distinguish between one connected device and another
+ *
+ * @param[in] device to consider
+ *
+ * @return true if the device needs distinguish on address, false otherwise..
+ */
+static bool device_distinguishes_on_address(audio_devices_t device)
+{
+ return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL & ~AUDIO_DEVICE_BIT_IN) != 0);
+}
diff --git a/services/audiopolicy/common/managerdefinitions/Android.mk b/services/audiopolicy/common/managerdefinitions/Android.mk
new file mode 100644
index 0000000..71ba1cb
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/Android.mk
@@ -0,0 +1,34 @@
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ src/DeviceDescriptor.cpp \
+ src/AudioGain.cpp \
+ src/StreamDescriptor.cpp \
+ src/HwModule.cpp \
+ src/IOProfile.cpp \
+ src/AudioPort.cpp \
+ src/AudioPolicyMix.cpp \
+ src/AudioPatch.cpp \
+ src/AudioInputDescriptor.cpp \
+ src/AudioOutputDescriptor.cpp \
+ src/EffectDescriptor.cpp \
+ src/ConfigParsingUtils.cpp \
+ src/SoundTriggerSession.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libutils \
+ liblog \
+
+LOCAL_C_INCLUDES += \
+ $(LOCAL_PATH)/include \
+ $(TOPDIR)frameworks/av/services/audiopolicy/common/include \
+
+LOCAL_EXPORT_C_INCLUDE_DIRS := \
+ $(LOCAL_PATH)/include
+
+LOCAL_MODULE := libaudiopolicycomponents
+
+include $(BUILD_STATIC_LIBRARY)
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioGain.h b/services/audiopolicy/common/managerdefinitions/include/AudioGain.h
new file mode 100644
index 0000000..21fbf9b
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioGain.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+#include <system/audio.h>
+
+namespace android {
+
+class AudioGain: public RefBase
+{
+public:
+ AudioGain(int index, bool useInChannelMask);
+ virtual ~AudioGain() {}
+
+ void dump(int fd, int spaces, int index) const;
+
+ void getDefaultConfig(struct audio_gain_config *config);
+ status_t checkConfig(const struct audio_gain_config *config);
+ int mIndex;
+ struct audio_gain mGain;
+ bool mUseInChannelMask;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
new file mode 100644
index 0000000..7536a37
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
@@ -0,0 +1,85 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include "AudioPort.h"
+#include <utils/Errors.h>
+#include <system/audio.h>
+#include <utils/SortedVector.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+class IOProfile;
+class AudioMix;
+
+// descriptor for audio inputs. Used to maintain current configuration of each opened audio input
+// and keep track of the usage of this input.
+class AudioInputDescriptor: public AudioPortConfig
+{
+public:
+ AudioInputDescriptor(const sp<IOProfile>& profile);
+ void setIoHandle(audio_io_handle_t ioHandle);
+
+ audio_module_handle_t getModuleHandle() const;
+
+ status_t dump(int fd);
+
+ audio_port_handle_t mId;
+ audio_io_handle_t mIoHandle; // input handle
+ audio_devices_t mDevice; // current device this input is routed to
+ AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
+ audio_patch_handle_t mPatchHandle;
+ uint32_t mRefCount; // number of AudioRecord clients using
+ // this input
+ uint32_t mOpenRefCount;
+ audio_source_t mInputSource; // input source selected by application
+ //(mediarecorder.h)
+ const sp<IOProfile> mProfile; // I/O profile this output derives from
+ SortedVector<audio_session_t> mSessions; // audio sessions attached to this input
+ bool mIsSoundTrigger; // used by a soundtrigger capture
+
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+ virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+ void toAudioPort(struct audio_port *port) const;
+};
+
+class AudioInputCollection :
+ public DefaultKeyedVector< audio_io_handle_t, sp<AudioInputDescriptor> >
+{
+public:
+ bool isSourceActive(audio_source_t source) const;
+
+ sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
+
+ uint32_t activeInputsCount() const;
+
+ /**
+ * return io handle of active input or 0 if no input is active
+ * Only considers inputs from physical devices (e.g. main mic, headset mic) when
+ * ignoreVirtualInputs is true.
+ */
+ audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
+
+ audio_devices_t getSupportedDevices(audio_io_handle_t handle) const;
+
+ status_t dump(int fd) const;
+};
+
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
new file mode 100644
index 0000000..43ee691
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -0,0 +1,113 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include "AudioPort.h"
+#include <RoutingStrategy.h>
+#include <utils/Errors.h>
+#include <utils/Timers.h>
+#include <utils/KeyedVector.h>
+#include <system/audio.h>
+
+namespace android {
+
+class IOProfile;
+class AudioMix;
+
+// descriptor for audio outputs. Used to maintain current configuration of each opened audio output
+// and keep track of the usage of this output by each audio stream type.
+class AudioOutputDescriptor: public AudioPortConfig
+{
+public:
+ AudioOutputDescriptor(const sp<IOProfile>& profile);
+
+ status_t dump(int fd);
+
+ audio_devices_t device() const;
+ void changeRefCount(audio_stream_type_t stream, int delta);
+
+ void setIoHandle(audio_io_handle_t ioHandle);
+ bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
+ audio_devices_t supportedDevices();
+ uint32_t latency();
+ bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
+ bool isActive(uint32_t inPastMs = 0) const;
+ bool isStreamActive(audio_stream_type_t stream,
+ uint32_t inPastMs = 0,
+ nsecs_t sysTime = 0) const;
+
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+ virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+ void toAudioPort(struct audio_port *port) const;
+
+ audio_module_handle_t getModuleHandle() const;
+
+ audio_port_handle_t mId;
+ audio_io_handle_t mIoHandle; // output handle
+ uint32_t mLatency; //
+ audio_output_flags_t mFlags; //
+ audio_devices_t mDevice; // current device this output is routed to
+ AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
+ audio_patch_handle_t mPatchHandle;
+ uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
+ nsecs_t mStopTime[AUDIO_STREAM_CNT];
+ sp<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output
+ sp<AudioOutputDescriptor> mOutput2; // used by duplicated outputs: second output
+ float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume
+ int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter
+ const sp<IOProfile> mProfile; // I/O profile this output derives from
+ bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
+ // device selection. See checkDeviceMuteStrategies()
+ uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
+};
+
+class AudioOutputCollection :
+ public DefaultKeyedVector< audio_io_handle_t, sp<AudioOutputDescriptor> >
+{
+public:
+ bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+
+ /**
+ * return whether a stream is playing remotely, override to change the definition of
+ * local/remote playback, used for instance by notification manager to not make
+ * media players lose audio focus when not playing locally
+ * For the base implementation, "remotely" means playing during screen mirroring which
+ * uses an output for playback with a non-empty, non "0" address.
+ */
+ bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+
+ /**
+ * returns the A2DP output handle if it is open or 0 otherwise
+ */
+ audio_io_handle_t getA2dpOutput() const;
+
+ sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
+
+ sp<AudioOutputDescriptor> getPrimaryOutput() const;
+
+ /**
+ * return true if any output is playing anything besides the stream to ignore
+ */
+ bool isAnyOutputActive(audio_stream_type_t streamToIgnore) const;
+
+ audio_devices_t getSupportedDevices(audio_io_handle_t handle) const;
+
+ status_t dump(int fd) const;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h
new file mode 100644
index 0000000..385f257
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h
@@ -0,0 +1,55 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <system/audio.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+class AudioPatch : public RefBase
+{
+public:
+ AudioPatch(const struct audio_patch *patch, uid_t uid);
+
+ status_t dump(int fd, int spaces, int index) const;
+
+ audio_patch_handle_t mHandle;
+ struct audio_patch mPatch;
+ uid_t mUid;
+ audio_patch_handle_t mAfPatchHandle;
+
+private:
+ static volatile int32_t mNextUniqueId;
+};
+
+class AudioPatchCollection : public DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> >
+{
+public:
+ status_t addAudioPatch(audio_patch_handle_t handle, const sp<AudioPatch>& patch);
+
+ status_t removeAudioPatch(audio_patch_handle_t handle);
+
+ status_t listAudioPatches(unsigned int *num_patches, struct audio_patch *patches) const;
+
+ status_t dump(int fd) const;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
new file mode 100644
index 0000000..988aed6
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
@@ -0,0 +1,81 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <utils/RefBase.h>
+#include <media/AudioPolicy.h>
+#include <utils/KeyedVector.h>
+#include <hardware/audio.h>
+#include <utils/String8.h>
+
+namespace android {
+
+class AudioOutputDescriptor;
+
+/**
+ * custom mix entry in mPolicyMixes
+ */
+class AudioPolicyMix : public RefBase {
+public:
+ AudioPolicyMix() {}
+
+ const sp<AudioOutputDescriptor> &getOutput() const;
+
+ void setOutput(sp<AudioOutputDescriptor> &output);
+
+ void clearOutput();
+
+ android::AudioMix &getMix();
+
+ void setMix(AudioMix &mix);
+
+private:
+ AudioMix mMix; // Audio policy mix descriptor
+ sp<AudioOutputDescriptor> mOutput; // Corresponding output stream
+};
+
+
+class AudioPolicyMixCollection : public DefaultKeyedVector<String8, sp<AudioPolicyMix> >
+{
+public:
+ status_t getAudioPolicyMix(String8 address, sp<AudioPolicyMix> &policyMix) const;
+
+ status_t registerMix(String8 address, AudioMix mix);
+
+ status_t unregisterMix(String8 address);
+
+ void closeOutput(sp<AudioOutputDescriptor> &desc);
+
+ /**
+ * Try to find an output descriptor for the given attributes.
+ *
+ * @param[in] attributes to consider for the research of output descriptor.
+ * @param[out] desc to return if an output could be found.
+ *
+ * @return NO_ERROR if an output was found for the given attribute (in this case, the
+ * descriptor output param is initialized), error code otherwise.
+ */
+ status_t getOutputForAttr(audio_attributes_t attributes, sp<AudioOutputDescriptor> &desc);
+
+ audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
+ audio_devices_t availableDeviceTypes,
+ AudioMix **policyMix);
+
+ status_t getInputMixForAttr(audio_attributes_t attr, AudioMix *&policyMix);
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
new file mode 100644
index 0000000..4f7f2bc
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
@@ -0,0 +1,125 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <utils/String8.h>
+#include <utils/Vector.h>
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <system/audio.h>
+#include <cutils/config_utils.h>
+
+namespace android {
+
+class HwModule;
+class AudioGain;
+
+class AudioPort : public virtual RefBase
+{
+public:
+ AudioPort(const String8& name, audio_port_type_t type,
+ audio_port_role_t role, const sp<HwModule>& module);
+ virtual ~AudioPort() {}
+
+ audio_port_handle_t getHandle() { return mId; }
+
+ void attach(const sp<HwModule>& module);
+ bool isAttached() { return mId != 0; }
+
+ static audio_port_handle_t getNextUniqueId();
+
+ virtual void toAudioPort(struct audio_port *port) const;
+
+ void importAudioPort(const sp<AudioPort> port);
+ void clearCapabilities();
+
+ void loadSamplingRates(char *name);
+ void loadFormats(char *name);
+ void loadOutChannels(char *name);
+ void loadInChannels(char *name);
+
+ audio_gain_mode_t loadGainMode(char *name);
+ void loadGain(cnode *root, int index);
+ virtual void loadGains(cnode *root);
+
+ // searches for an exact match
+ status_t checkExactSamplingRate(uint32_t samplingRate) const;
+ // searches for a compatible match, and returns the best match via updatedSamplingRate
+ status_t checkCompatibleSamplingRate(uint32_t samplingRate,
+ uint32_t *updatedSamplingRate) const;
+ // searches for an exact match
+ status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
+ // searches for a compatible match, currently implemented for input channel masks only
+ status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
+ status_t checkFormat(audio_format_t format) const;
+ status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
+
+ uint32_t pickSamplingRate() const;
+ audio_channel_mask_t pickChannelMask() const;
+ audio_format_t pickFormat() const;
+
+ static const audio_format_t sPcmFormatCompareTable[];
+ static int compareFormats(audio_format_t format1, audio_format_t format2);
+
+ audio_module_handle_t getModuleHandle() const;
+
+ void dump(int fd, int spaces) const;
+
+ String8 mName;
+ audio_port_type_t mType;
+ audio_port_role_t mRole;
+ bool mUseInChannelMask;
+ // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
+ // indicates the supported parameters should be read from the output stream
+ // after it is opened for the first time
+ Vector <uint32_t> mSamplingRates; // supported sampling rates
+ Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
+ Vector <audio_format_t> mFormats; // supported audio formats
+ Vector < sp<AudioGain> > mGains; // gain controllers
+ sp<HwModule> mModule; // audio HW module exposing this I/O stream
+ uint32_t mFlags; // attribute flags (e.g primary output,
+ // direct output...).
+
+
+protected:
+ //TODO - clarify the role of mId in this case, both an "attached" indicator
+ // and a unique ID for identifying a port to the (upcoming) selection API,
+ // and its relationship to the mId in AudioOutputDescriptor and AudioInputDescriptor.
+ audio_port_handle_t mId;
+
+private:
+ static volatile int32_t mNextUniqueId;
+};
+
+class AudioPortConfig : public virtual RefBase
+{
+public:
+ AudioPortConfig();
+ virtual ~AudioPortConfig() {}
+
+ status_t applyAudioPortConfig(const struct audio_port_config *config,
+ struct audio_port_config *backupConfig = NULL);
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const = 0;
+ virtual sp<AudioPort> getAudioPort() const = 0;
+ uint32_t mSamplingRate;
+ audio_format_t mFormat;
+ audio_channel_mask_t mChannelMask;
+ struct audio_gain_config mGain;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h
new file mode 100644
index 0000000..14a7d36
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h
@@ -0,0 +1,202 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include "DeviceDescriptor.h"
+#include "HwModule.h"
+#include "audio_policy_conf.h"
+#include <system/audio.h>
+#include <utils/Log.h>
+#include <utils/Vector.h>
+#include <utils/SortedVector.h>
+#include <cutils/config_utils.h>
+#include <utils/RefBase.h>
+#include <system/audio_policy.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// Definitions for audio_policy.conf file parsing
+// ----------------------------------------------------------------------------
+
+struct StringToEnum {
+ const char *name;
+ uint32_t value;
+};
+
+#define STRING_TO_ENUM(string) { #string, string }
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+#endif
+
+const StringToEnum sDeviceNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
+};
+
+const StringToEnum sOutputFlagNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
+};
+
+const StringToEnum sInputFlagNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
+ STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
+};
+
+const StringToEnum sFormatNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+ STRING_TO_ENUM(AUDIO_FORMAT_MP3),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
+ STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
+ STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
+ STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
+ STRING_TO_ENUM(AUDIO_FORMAT_AC3),
+ STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
+ STRING_TO_ENUM(AUDIO_FORMAT_DTS),
+ STRING_TO_ENUM(AUDIO_FORMAT_DTS_HD),
+};
+
+const StringToEnum sOutChannelsNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+const StringToEnum sInChannelsNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
+};
+
+const StringToEnum sGainModeNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
+};
+
+class ConfigParsingUtils
+{
+public:
+ static uint32_t stringToEnum(const struct StringToEnum *table,
+ size_t size,
+ const char *name);
+ static const char *enumToString(const struct StringToEnum *table,
+ size_t size,
+ uint32_t value);
+ static bool stringToBool(const char *value);
+ static uint32_t parseOutputFlagNames(char *name);
+ static uint32_t parseInputFlagNames(char *name);
+ static audio_devices_t parseDeviceNames(char *name);
+
+ static void loadHwModules(cnode *root, HwModuleCollection &hwModules,
+ DeviceVector &availableInputDevices,
+ DeviceVector &availableOutputDevices,
+ sp<DeviceDescriptor> &defaultOutputDevices,
+ bool &isSpeakerDrcEnabled);
+
+ static void loadGlobalConfig(cnode *root, const sp<HwModule>& module,
+ DeviceVector &availableInputDevices,
+ DeviceVector &availableOutputDevices,
+ sp<DeviceDescriptor> &defaultOutputDevices,
+ bool &isSpeakerDrcEnabled);
+
+ static status_t loadAudioPolicyConfig(const char *path,
+ HwModuleCollection &hwModules,
+ DeviceVector &availableInputDevices,
+ DeviceVector &availableOutputDevices,
+ sp<DeviceDescriptor> &defaultOutputDevices,
+ bool &isSpeakerDrcEnabled);
+
+private:
+ static void loadHwModule(cnode *root, HwModuleCollection &hwModules,
+ DeviceVector &availableInputDevices,
+ DeviceVector &availableOutputDevices,
+ sp<DeviceDescriptor> &defaultOutputDevices,
+ bool &isSpeakerDrcEnabled);
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
new file mode 100644
index 0000000..d15f6b4
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -0,0 +1,92 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include "AudioPort.h"
+#include <utils/Errors.h>
+#include <utils/String8.h>
+#include <utils/SortedVector.h>
+#include <cutils/config_utils.h>
+#include <system/audio.h>
+#include <system/audio_policy.h>
+
+namespace android {
+
+class DeviceDescriptor : public AudioPort, public AudioPortConfig
+{
+public:
+ DeviceDescriptor(const String8& name, audio_devices_t type);
+
+ virtual ~DeviceDescriptor() {}
+
+ bool equals(const sp<DeviceDescriptor>& other) const;
+
+ // AudioPortConfig
+ virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+
+ // AudioPort
+ virtual void loadGains(cnode *root);
+ virtual void toAudioPort(struct audio_port *port) const;
+
+ audio_devices_t type() const { return mDeviceType; }
+ status_t dump(int fd, int spaces, int index) const;
+
+ String8 mAddress;
+ audio_port_handle_t mId;
+
+ static String8 emptyNameStr;
+
+private:
+ audio_devices_t mDeviceType;
+
+friend class DeviceVector;
+};
+
+class DeviceVector : public SortedVector< sp<DeviceDescriptor> >
+{
+public:
+ DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
+
+ ssize_t add(const sp<DeviceDescriptor>& item);
+ ssize_t remove(const sp<DeviceDescriptor>& item);
+ ssize_t indexOf(const sp<DeviceDescriptor>& item) const;
+
+ audio_devices_t types() const { return mDeviceTypes; }
+
+ void loadDevicesFromType(audio_devices_t types);
+ void loadDevicesFromName(char *name, const DeviceVector& declaredDevices);
+
+ sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const;
+ DeviceVector getDevicesFromType(audio_devices_t types) const;
+ sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
+ sp<DeviceDescriptor> getDeviceFromName(const String8& name) const;
+ DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address) const;
+
+ audio_devices_t getDevicesFromHwModule(audio_module_handle_t moduleHandle) const;
+
+ audio_policy_dev_state_t getDeviceConnectionState(const sp<DeviceDescriptor> &devDesc) const;
+
+ status_t dump(int fd, const String8 &direction) const;
+
+private:
+ void refreshTypes();
+ audio_devices_t mDeviceTypes;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/EffectDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/EffectDescriptor.h
new file mode 100644
index 0000000..c9783a1
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/EffectDescriptor.h
@@ -0,0 +1,71 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <RoutingStrategy.h>
+#include <hardware/audio_effect.h>
+#include <utils/KeyedVector.h>
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+
+namespace android {
+
+
+class EffectDescriptor : public RefBase
+{
+public:
+ status_t dump(int fd);
+
+ int mIo; // io the effect is attached to
+ routing_strategy mStrategy; // routing strategy the effect is associated to
+ int mSession; // audio session the effect is on
+ effect_descriptor_t mDesc; // effect descriptor
+ bool mEnabled; // enabled state: CPU load being used or not
+};
+
+class EffectDescriptorCollection : public KeyedVector<int, sp<EffectDescriptor> >
+{
+public:
+ EffectDescriptorCollection();
+
+ status_t registerEffect(const effect_descriptor_t *desc, audio_io_handle_t io,
+ uint32_t strategy, int session, int id);
+ status_t unregisterEffect(int id);
+ status_t setEffectEnabled(int id, bool enabled);
+ uint32_t getMaxEffectsCpuLoad() const;
+ uint32_t getMaxEffectsMemory() const;
+ bool isNonOffloadableEffectEnabled();
+
+ status_t dump(int fd);
+
+private:
+ status_t setEffectEnabled(const sp<EffectDescriptor> &effectDesc, bool enabled);
+
+ uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
+ uint32_t mTotalEffectsMemory; // current memory used by effects
+
+ /**
+ * Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
+ */
+ static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
+ /**
+ * Maximum memory allocated to audio effects in KB
+ */
+ static const uint32_t MAX_EFFECTS_MEMORY = 512;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
new file mode 100644
index 0000000..92c3ea2
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
@@ -0,0 +1,74 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include "DeviceDescriptor.h"
+#include <utils/RefBase.h>
+#include <utils/String8.h>
+#include <utils/Errors.h>
+#include <utils/Vector.h>
+#include <system/audio.h>
+#include <cutils/config_utils.h>
+
+namespace android {
+
+class IOProfile;
+
+class HwModule : public RefBase
+{
+public:
+ HwModule(const char *name);
+ ~HwModule();
+
+ status_t loadOutput(cnode *root);
+ status_t loadInput(cnode *root);
+ status_t loadDevice(cnode *root);
+
+ status_t addOutputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address);
+ status_t removeOutputProfile(String8 name);
+ status_t addInputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address);
+ status_t removeInputProfile(String8 name);
+
+ audio_module_handle_t getHandle() const { return mHandle; }
+
+ void dump(int fd);
+
+ const char *const mName; // base name of the audio HW module (primary, a2dp ...)
+ uint32_t mHalVersion; // audio HAL API version
+ audio_module_handle_t mHandle;
+ Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
+ Vector < sp<IOProfile> > mInputProfiles; // input profiles exposed by this module
+ DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf
+};
+
+class HwModuleCollection : public Vector< sp<HwModule> >
+{
+public:
+ sp<HwModule> getModuleFromName(const char *name) const;
+
+ sp <HwModule> getModuleForDevice(audio_devices_t device) const;
+
+ sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device,
+ const char *device_address,
+ const char *device_name) const;
+
+ status_t dump(int fd) const;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
new file mode 100644
index 0000000..095e759
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include "AudioPort.h"
+#include "DeviceDescriptor.h"
+#include <utils/String8.h>
+#include <system/audio.h>
+
+namespace android {
+
+class HwModule;
+
+// the IOProfile class describes the capabilities of an output or input stream.
+// It is currently assumed that all combination of listed parameters are supported.
+// It is used by the policy manager to determine if an output or input is suitable for
+// a given use case, open/close it accordingly and connect/disconnect audio tracks
+// to/from it.
+class IOProfile : public AudioPort
+{
+public:
+ IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module);
+ virtual ~IOProfile();
+
+ // This method is used for both output and input.
+ // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
+ // For input, flags is interpreted as audio_input_flags_t.
+ // TODO: merge audio_output_flags_t and audio_input_flags_t.
+ bool isCompatibleProfile(audio_devices_t device,
+ String8 address,
+ uint32_t samplingRate,
+ uint32_t *updatedSamplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ uint32_t flags) const;
+
+ void dump(int fd);
+ void log();
+
+ DeviceVector mSupportedDevices; // supported devices
+ // (devices this output can be routed to)
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audiopolicy/common/managerdefinitions/include/SoundTriggerSession.h
similarity index 60%
copy from services/audiopolicy/AudioPolicyFactory.cpp
copy to services/audiopolicy/common/managerdefinitions/include/SoundTriggerSession.h
index 2ae7bc1..420e6d7 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/services/audiopolicy/common/managerdefinitions/include/SoundTriggerSession.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2014 The Android Open Source Project
+ * Copyright (C) 2015 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.
@@ -14,19 +14,20 @@
* limitations under the License.
*/
-#include "AudioPolicyManager.h"
+#pragma once
+
+#include <system/audio.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
namespace android {
-extern "C" AudioPolicyInterface* createAudioPolicyManager(
- AudioPolicyClientInterface *clientInterface)
+class SoundTriggerSessionCollection : public DefaultKeyedVector<audio_session_t, audio_io_handle_t>
{
- return new AudioPolicyManager(clientInterface);
-}
+public:
+ status_t releaseSession(audio_session_t session);
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
-{
- delete interface;
-}
+ status_t acquireSession(audio_session_t session, audio_io_handle_t ioHandle);
+};
}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/StreamDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/StreamDescriptor.h
new file mode 100644
index 0000000..84db5ab
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/StreamDescriptor.h
@@ -0,0 +1,85 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <Volume.h>
+#include <utils/KeyedVector.h>
+#include <utils/StrongPointer.h>
+#include <utils/SortedVector.h>
+#include <hardware/audio.h>
+
+namespace android {
+
+// stream descriptor used for volume control
+class StreamDescriptor
+{
+public:
+ StreamDescriptor();
+
+ int getVolumeIndex(audio_devices_t device) const;
+ bool canBeMuted() const { return mCanBeMuted; }
+ void clearCurrentVolumeIndex();
+ void addCurrentVolumeIndex(audio_devices_t device, int index);
+ int getVolumeIndexMin() const { return mIndexMin; }
+ int getVolumeIndexMax() const { return mIndexMax; }
+ void setVolumeIndexMin(int volIndexMin);
+ void setVolumeIndexMax(int volIndexMax);
+
+ void dump(int fd) const;
+
+ void setVolumeCurvePoint(Volume::device_category deviceCategory, const VolumeCurvePoint *point);
+ const VolumeCurvePoint *getVolumeCurvePoint(Volume::device_category deviceCategory) const
+ {
+ return mVolumeCurve[deviceCategory];
+ }
+
+private:
+ const VolumeCurvePoint *mVolumeCurve[Volume::DEVICE_CATEGORY_CNT];
+ KeyedVector<audio_devices_t, int> mIndexCur; /**< current volume index per device. */
+ int mIndexMin; /**< min volume index. */
+ int mIndexMax; /**< max volume index. */
+ bool mCanBeMuted; /**< true is the stream can be muted. */
+};
+
+/**
+ * stream descriptors collection for volume control
+ */
+class StreamDescriptorCollection : public DefaultKeyedVector<audio_stream_type_t, StreamDescriptor>
+{
+public:
+ StreamDescriptorCollection();
+
+ void clearCurrentVolumeIndex(audio_stream_type_t stream);
+ void addCurrentVolumeIndex(audio_stream_type_t stream, audio_devices_t device, int index);
+
+ bool canBeMuted(audio_stream_type_t stream);
+
+ status_t dump(int fd) const;
+
+ void setVolumeCurvePoint(audio_stream_type_t stream,
+ Volume::device_category deviceCategory,
+ const VolumeCurvePoint *point);
+
+ const VolumeCurvePoint *getVolumeCurvePoint(audio_stream_type_t stream,
+ Volume::device_category deviceCategory) const;
+
+ void setVolumeIndexMin(audio_stream_type_t stream,int volIndexMin);
+ void setVolumeIndexMax(audio_stream_type_t stream,int volIndexMax);
+
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/audio_policy_conf.h b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
similarity index 91%
rename from services/audiopolicy/audio_policy_conf.h
rename to services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
index 2535a67..a393e3b 100644
--- a/services/audiopolicy/audio_policy_conf.h
+++ b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
@@ -14,9 +14,7 @@
* limitations under the License.
*/
-
-#ifndef ANDROID_AUDIO_POLICY_CONF_H
-#define ANDROID_AUDIO_POLICY_CONF_H
+#pragma once
/////////////////////////////////////////////////
@@ -53,9 +51,9 @@
// "formats" in outputs descriptors indicating that supported
// values should be queried after opening the output.
-#define DEVICES_TAG "devices"
-#define DEVICE_TYPE "type"
-#define DEVICE_ADDRESS "address"
+#define APM_DEVICES_TAG "devices"
+#define APM_DEVICE_TYPE "type"
+#define APM_DEVICE_ADDRESS "address"
#define MIXERS_TAG "mixers"
#define MIXER_TYPE "type"
@@ -71,7 +69,3 @@
#define GAIN_STEP_VALUE "step_value_mB"
#define GAIN_MIN_RAMP_MS "min_ramp_ms"
#define GAIN_MAX_RAMP_MS "max_ramp_ms"
-
-
-
-#endif // ANDROID_AUDIO_POLICY_CONF_H
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp
new file mode 100644
index 0000000..fc7b0cc
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp
@@ -0,0 +1,130 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioGain"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include "AudioGain.h"
+#include "StreamDescriptor.h"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include <math.h>
+
+namespace android {
+
+AudioGain::AudioGain(int index, bool useInChannelMask)
+{
+ mIndex = index;
+ mUseInChannelMask = useInChannelMask;
+ memset(&mGain, 0, sizeof(struct audio_gain));
+}
+
+void AudioGain::getDefaultConfig(struct audio_gain_config *config)
+{
+ config->index = mIndex;
+ config->mode = mGain.mode;
+ config->channel_mask = mGain.channel_mask;
+ if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ config->values[0] = mGain.default_value;
+ } else {
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ config->values[i] = mGain.default_value;
+ }
+ }
+ if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ config->ramp_duration_ms = mGain.min_ramp_ms;
+ }
+}
+
+status_t AudioGain::checkConfig(const struct audio_gain_config *config)
+{
+ if ((config->mode & ~mGain.mode) != 0) {
+ return BAD_VALUE;
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ if ((config->values[0] < mGain.min_value) ||
+ (config->values[0] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ } else {
+ if ((config->channel_mask & ~mGain.channel_mask) != 0) {
+ return BAD_VALUE;
+ }
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(config->channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(config->channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ if ((config->values[i] < mGain.min_value) ||
+ (config->values[i] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ }
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
+ (config->ramp_duration_ms > mGain.max_ramp_ms)) {
+ return BAD_VALUE;
+ }
+ }
+ return NO_ERROR;
+}
+
+void AudioGain::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
new file mode 100644
index 0000000..fa66728
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
@@ -0,0 +1,187 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioInputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioInputDescriptor.h"
+#include "IOProfile.h"
+#include "AudioGain.h"
+#include "HwModule.h"
+#include <media/AudioPolicy.h>
+#include <policy.h>
+
+namespace android {
+
+AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
+ : mId(0), mIoHandle(0),
+ mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
+ mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
+{
+ if (profile != NULL) {
+ mSamplingRate = profile->pickSamplingRate();
+ mFormat = profile->pickFormat();
+ mChannelMask = profile->pickChannelMask();
+ if (profile->mGains.size() > 0) {
+ profile->mGains[0]->getDefaultConfig(&mGain);
+ }
+ }
+}
+
+void AudioInputDescriptor::setIoHandle(audio_io_handle_t ioHandle)
+{
+ mId = AudioPort::getNextUniqueId();
+ mIoHandle = ioHandle;
+}
+
+audio_module_handle_t AudioInputDescriptor::getModuleHandle() const
+{
+ return mProfile->getModuleHandle();
+}
+
+void AudioInputDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ ALOG_ASSERT(mProfile != 0,
+ "toAudioPortConfig() called on input with null profile %d", mIoHandle);
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+ AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = AUDIO_PORT_ROLE_SINK;
+ dstConfig->type = AUDIO_PORT_TYPE_MIX;
+ dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+ dstConfig->ext.mix.handle = mIoHandle;
+ dstConfig->ext.mix.usecase.source = mInputSource;
+}
+
+void AudioInputDescriptor::toAudioPort(struct audio_port *port) const
+{
+ ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
+
+ mProfile->toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.mix.hw_module = mProfile->mModule->mHandle;
+ port->ext.mix.handle = mIoHandle;
+ port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioInputDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " ID: %d\n", mId);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Format: %d\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+bool AudioInputCollection::isSourceActive(audio_source_t source) const
+{
+ for (size_t i = 0; i < size(); i++) {
+ const sp<AudioInputDescriptor> inputDescriptor = valueAt(i);
+ if (inputDescriptor->mRefCount == 0) {
+ continue;
+ }
+ if (inputDescriptor->mInputSource == (int)source) {
+ return true;
+ }
+ }
+ return false;
+}
+
+sp<AudioInputDescriptor> AudioInputCollection::getInputFromId(audio_port_handle_t id) const
+{
+ sp<AudioInputDescriptor> inputDesc = NULL;
+ for (size_t i = 0; i < size(); i++) {
+ inputDesc = valueAt(i);
+ if (inputDesc->mId == id) {
+ break;
+ }
+ }
+ return inputDesc;
+}
+
+uint32_t AudioInputCollection::activeInputsCount() const
+{
+ uint32_t count = 0;
+ for (size_t i = 0; i < size(); i++) {
+ const sp<AudioInputDescriptor> desc = valueAt(i);
+ if (desc->mRefCount > 0) {
+ count++;
+ }
+ }
+ return count;
+}
+
+audio_io_handle_t AudioInputCollection::getActiveInput(bool ignoreVirtualInputs)
+{
+ for (size_t i = 0; i < size(); i++) {
+ const sp<AudioInputDescriptor> input_descriptor = valueAt(i);
+ if ((input_descriptor->mRefCount > 0)
+ && (!ignoreVirtualInputs || !is_virtual_input_device(input_descriptor->mDevice))) {
+ return keyAt(i);
+ }
+ }
+ return 0;
+}
+
+audio_devices_t AudioInputCollection::getSupportedDevices(audio_io_handle_t handle) const
+{
+ sp<AudioInputDescriptor> inputDesc = valueFor(handle);
+ audio_devices_t devices = inputDesc->mProfile->mSupportedDevices.types();
+ return devices;
+}
+
+status_t AudioInputCollection::dump(int fd) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "\nInputs dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < size(); i++) {
+ snprintf(buffer, SIZE, "- Input %d dump:\n", keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ valueAt(i)->dump(fd);
+ }
+
+ return NO_ERROR;
+}
+
+}; //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
new file mode 100644
index 0000000..cdb5b51
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -0,0 +1,333 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioOutputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioOutputDescriptor.h"
+#include "IOProfile.h"
+#include "AudioGain.h"
+#include "HwModule.h"
+#include <media/AudioPolicy.h>
+
+// A device mask for all audio output devices that are considered "remote" when evaluating
+// active output devices in isStreamActiveRemotely()
+#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+
+namespace android {
+
+AudioOutputDescriptor::AudioOutputDescriptor(const sp<IOProfile>& profile)
+ : mId(0), mIoHandle(0), mLatency(0),
+ mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
+ mPatchHandle(0),
+ mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
+{
+ // clear usage count for all stream types
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ mRefCount[i] = 0;
+ mCurVolume[i] = -1.0;
+ mMuteCount[i] = 0;
+ mStopTime[i] = 0;
+ }
+ for (int i = 0; i < NUM_STRATEGIES; i++) {
+ mStrategyMutedByDevice[i] = false;
+ }
+ if (profile != NULL) {
+ mFlags = (audio_output_flags_t)profile->mFlags;
+ mSamplingRate = profile->pickSamplingRate();
+ mFormat = profile->pickFormat();
+ mChannelMask = profile->pickChannelMask();
+ if (profile->mGains.size() > 0) {
+ profile->mGains[0]->getDefaultConfig(&mGain);
+ }
+ }
+}
+
+audio_module_handle_t AudioOutputDescriptor::getModuleHandle() const
+{
+ return mProfile->getModuleHandle();
+}
+
+audio_devices_t AudioOutputDescriptor::device() const
+{
+ if (isDuplicated()) {
+ return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
+ } else {
+ return mDevice;
+ }
+}
+
+void AudioOutputDescriptor::setIoHandle(audio_io_handle_t ioHandle)
+{
+ mId = AudioPort::getNextUniqueId();
+ mIoHandle = ioHandle;
+}
+
+uint32_t AudioOutputDescriptor::latency()
+{
+ if (isDuplicated()) {
+ return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
+ } else {
+ return mLatency;
+ }
+}
+
+bool AudioOutputDescriptor::sharesHwModuleWith(
+ const sp<AudioOutputDescriptor> outputDesc)
+{
+ if (isDuplicated()) {
+ return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
+ } else if (outputDesc->isDuplicated()){
+ return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
+ } else {
+ return (mProfile->mModule == outputDesc->mProfile->mModule);
+ }
+}
+
+void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
+ int delta)
+{
+ // forward usage count change to attached outputs
+ if (isDuplicated()) {
+ mOutput1->changeRefCount(stream, delta);
+ mOutput2->changeRefCount(stream, delta);
+ }
+ if ((delta + (int)mRefCount[stream]) < 0) {
+ ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
+ delta, stream, mRefCount[stream]);
+ mRefCount[stream] = 0;
+ return;
+ }
+ mRefCount[stream] += delta;
+ ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+audio_devices_t AudioOutputDescriptor::supportedDevices()
+{
+ if (isDuplicated()) {
+ return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
+ } else {
+ return mProfile->mSupportedDevices.types() ;
+ }
+}
+
+bool AudioOutputDescriptor::isActive(uint32_t inPastMs) const
+{
+ nsecs_t sysTime = 0;
+ if (inPastMs != 0) {
+ sysTime = systemTime();
+ }
+ for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+ if (i == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
+ uint32_t inPastMs,
+ nsecs_t sysTime) const
+{
+ if (mRefCount[stream] != 0) {
+ return true;
+ }
+ if (inPastMs == 0) {
+ return false;
+ }
+ if (sysTime == 0) {
+ sysTime = systemTime();
+ }
+ if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
+ return true;
+ }
+ return false;
+}
+
+void AudioOutputDescriptor::toAudioPortConfig(
+ struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
+
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+ AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
+ dstConfig->type = AUDIO_PORT_TYPE_MIX;
+ dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+ dstConfig->ext.mix.handle = mIoHandle;
+ dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
+}
+
+void AudioOutputDescriptor::toAudioPort(
+ struct audio_port *port) const
+{
+ ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
+ mProfile->toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.mix.hw_module = mProfile->mModule->mHandle;
+ port->ext.mix.handle = mIoHandle;
+ port->ext.mix.latency_class =
+ mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioOutputDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " ID: %d\n", mId);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Devices %08x\n", device());
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
+ result.append(buffer);
+ for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+ snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n",
+ i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
+ result.append(buffer);
+ }
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+bool AudioOutputCollection::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
+{
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < this->size(); i++) {
+ const sp<AudioOutputDescriptor> outputDesc = this->valueAt(i);
+ if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioOutputCollection::isStreamActiveRemotely(audio_stream_type_t stream,
+ uint32_t inPastMs) const
+{
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < size(); i++) {
+ const sp<AudioOutputDescriptor> outputDesc = valueAt(i);
+ if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
+ outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
+ // do not consider re routing (when the output is going to a dynamic policy)
+ // as "remote playback"
+ if (outputDesc->mPolicyMix == NULL) {
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+audio_io_handle_t AudioOutputCollection::getA2dpOutput() const
+{
+ for (size_t i = 0; i < size(); i++) {
+ sp<AudioOutputDescriptor> outputDesc = valueAt(i);
+ if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ return this->keyAt(i);
+ }
+ }
+ return 0;
+}
+
+sp<AudioOutputDescriptor> AudioOutputCollection::getPrimaryOutput() const
+{
+ for (size_t i = 0; i < size(); i++) {
+ const sp<AudioOutputDescriptor> outputDesc = valueAt(i);
+ if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ return outputDesc;
+ }
+ }
+ return NULL;
+}
+
+sp<AudioOutputDescriptor> AudioOutputCollection::getOutputFromId(audio_port_handle_t id) const
+{
+ sp<AudioOutputDescriptor> outputDesc = NULL;
+ for (size_t i = 0; i < size(); i++) {
+ outputDesc = valueAt(i);
+ if (outputDesc->mId == id) {
+ break;
+ }
+ }
+ return outputDesc;
+}
+
+bool AudioOutputCollection::isAnyOutputActive(audio_stream_type_t streamToIgnore) const
+{
+ for (size_t s = 0 ; s < AUDIO_STREAM_CNT ; s++) {
+ if (s == (size_t) streamToIgnore) {
+ continue;
+ }
+ for (size_t i = 0; i < size(); i++) {
+ const sp<AudioOutputDescriptor> outputDesc = valueAt(i);
+ if (outputDesc->mRefCount[s] != 0) {
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+audio_devices_t AudioOutputCollection::getSupportedDevices(audio_io_handle_t handle) const
+{
+ sp<AudioOutputDescriptor> outputDesc = valueFor(handle);
+ audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types();
+ return devices;
+}
+
+
+status_t AudioOutputCollection::dump(int fd) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "\nOutputs dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < size(); i++) {
+ snprintf(buffer, SIZE, "- Output %d dump:\n", keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ valueAt(i)->dump(fd);
+ }
+
+ return NO_ERROR;
+}
+
+}; //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
new file mode 100644
index 0000000..3a317fa
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
@@ -0,0 +1,154 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioPatch"
+//#define LOG_NDEBUG 0
+
+#include "AudioPatch.h"
+#include "AudioGain.h"
+#include "ConfigParsingUtils.h"
+#include <cutils/log.h>
+#include <utils/String8.h>
+
+namespace android {
+
+int32_t volatile AudioPatch::mNextUniqueId = 1;
+
+AudioPatch::AudioPatch(const struct audio_patch *patch, uid_t uid) :
+ mHandle(static_cast<audio_patch_handle_t>(android_atomic_inc(&mNextUniqueId))),
+ mPatch(*patch),
+ mUid(uid),
+ mAfPatchHandle(0)
+{
+}
+
+status_t AudioPatch::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
+ result.append(buffer);
+ for (size_t i = 0; i < mPatch.num_sources; i++) {
+ if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
+ snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+ mPatch.sources[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mPatch.sources[i].ext.device.type));
+ } else {
+ snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+ mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
+ }
+ result.append(buffer);
+ }
+ snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
+ result.append(buffer);
+ for (size_t i = 0; i < mPatch.num_sinks; i++) {
+ if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
+ snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+ mPatch.sinks[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mPatch.sinks[i].ext.device.type));
+ } else {
+ snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+ mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
+ }
+ result.append(buffer);
+ }
+
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioPatchCollection::addAudioPatch(audio_patch_handle_t handle,
+ const sp<AudioPatch>& patch)
+{
+ ssize_t index = indexOfKey(handle);
+
+ if (index >= 0) {
+ ALOGW("addAudioPatch() patch %d already in", handle);
+ return ALREADY_EXISTS;
+ }
+ add(handle, patch);
+ ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d"
+ "sink handle %d",
+ handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks,
+ patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id);
+ return NO_ERROR;
+}
+
+status_t AudioPatchCollection::removeAudioPatch(audio_patch_handle_t handle)
+{
+ ssize_t index = indexOfKey(handle);
+
+ if (index < 0) {
+ ALOGW("removeAudioPatch() patch %d not in", handle);
+ return ALREADY_EXISTS;
+ }
+ ALOGV("removeAudioPatch() handle %d af handle %d", handle, valueAt(index)->mAfPatchHandle);
+ removeItemsAt(index);
+ return NO_ERROR;
+}
+
+status_t AudioPatchCollection::listAudioPatches(unsigned int *num_patches,
+ struct audio_patch *patches) const
+{
+ if (num_patches == NULL || (*num_patches != 0 && patches == NULL)) {
+ return BAD_VALUE;
+ }
+ ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu",
+ *num_patches, patches, size());
+ if (patches == NULL) {
+ *num_patches = 0;
+ }
+
+ size_t patchesWritten = 0;
+ size_t patchesMax = *num_patches;
+ for (size_t i = 0; i < size() && patchesWritten < patchesMax; i++) {
+ const sp<AudioPatch> patch = valueAt(i);
+ patches[patchesWritten] = patch->mPatch;
+ patches[patchesWritten++].id = patch->mHandle;
+ ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d",
+ i, patch->mPatch.num_sources, patch->mPatch.num_sinks);
+ }
+ *num_patches = size();
+
+ ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches);
+ return NO_ERROR;
+}
+
+status_t AudioPatchCollection::dump(int fd) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ snprintf(buffer, SIZE, "\nAudio Patches:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < size(); i++) {
+ valueAt(i)->dump(fd, 2, i);
+ }
+ return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
new file mode 100644
index 0000000..84a53ebd
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -0,0 +1,193 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioPolicyMix"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyMix.h"
+#include "HwModule.h"
+#include "AudioPort.h"
+#include "IOProfile.h"
+#include "AudioGain.h"
+#include <AudioOutputDescriptor.h>
+
+namespace android {
+
+void AudioPolicyMix::setOutput(sp<AudioOutputDescriptor> &output)
+{
+ mOutput = output;
+}
+
+const sp<AudioOutputDescriptor> &AudioPolicyMix::getOutput() const
+{
+ return mOutput;
+}
+
+void AudioPolicyMix::clearOutput()
+{
+ mOutput.clear();
+}
+
+void AudioPolicyMix::setMix(AudioMix &mix)
+{
+ mMix = mix;
+}
+
+android::AudioMix &AudioPolicyMix::getMix()
+{
+ return mMix;
+}
+
+status_t AudioPolicyMixCollection::registerMix(String8 address, AudioMix mix)
+{
+ ssize_t index = indexOfKey(address);
+ if (index >= 0) {
+ ALOGE("registerPolicyMixes(): mix for address %s already registered", address.string());
+ return BAD_VALUE;
+ }
+ sp<AudioPolicyMix> policyMix = new AudioPolicyMix();
+ policyMix->setMix(mix);
+ add(address, policyMix);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyMixCollection::unregisterMix(String8 address)
+{
+ ssize_t index = indexOfKey(address);
+ if (index < 0) {
+ ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string());
+ return BAD_VALUE;
+ }
+
+ removeItemsAt(index);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyMixCollection::getAudioPolicyMix(String8 address,
+ sp<AudioPolicyMix> &policyMix) const
+{
+ ssize_t index = indexOfKey(address);
+ if (index < 0) {
+ ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string());
+ return BAD_VALUE;
+ }
+ policyMix = valueAt(index);
+ return NO_ERROR;
+}
+
+void AudioPolicyMixCollection::closeOutput(sp<AudioOutputDescriptor> &desc)
+{
+ for (size_t i = 0; i < size(); i++) {
+ sp<AudioPolicyMix> policyMix = valueAt(i);
+ if (policyMix->getOutput() == desc) {
+ policyMix->clearOutput();
+ }
+ }
+}
+
+status_t AudioPolicyMixCollection::getOutputForAttr(audio_attributes_t attributes,
+ sp<AudioOutputDescriptor> &desc)
+{
+ for (size_t i = 0; i < size(); i++) {
+ sp<AudioPolicyMix> policyMix = valueAt(i);
+ AudioMix mix = policyMix->getMix();
+
+ if (mix.mMixType == MIX_TYPE_PLAYERS) {
+ for (size_t j = 0; j < mix.mCriteria.size(); j++) {
+ if ((RULE_MATCH_ATTRIBUTE_USAGE == mix.mCriteria[j].mRule &&
+ mix.mCriteria[j].mAttr.mUsage == attributes.usage) ||
+ (RULE_EXCLUDE_ATTRIBUTE_USAGE == mix.mCriteria[j].mRule &&
+ mix.mCriteria[j].mAttr.mUsage != attributes.usage)) {
+ desc = policyMix->getOutput();
+ break;
+ }
+ if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
+ strncmp(attributes.tags + strlen("addr="),
+ mix.mRegistrationId.string(),
+ AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
+ desc = policyMix->getOutput();
+ break;
+ }
+ }
+ } else if (mix.mMixType == MIX_TYPE_RECORDERS) {
+ if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE &&
+ strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
+ strncmp(attributes.tags + strlen("addr="),
+ mix.mRegistrationId.string(),
+ AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
+ desc = policyMix->getOutput();
+ }
+ }
+ if (desc != 0) {
+ desc->mPolicyMix = &mix;
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+audio_devices_t AudioPolicyMixCollection::getDeviceAndMixForInputSource(audio_source_t inputSource,
+ audio_devices_t availDevices,
+ AudioMix **policyMix)
+{
+ for (size_t i = 0; i < size(); i++) {
+ AudioMix mix = valueAt(i)->getMix();
+
+ if (mix.mMixType != MIX_TYPE_RECORDERS) {
+ continue;
+ }
+ for (size_t j = 0; j < mix.mCriteria.size(); j++) {
+ if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mix.mCriteria[j].mRule &&
+ mix.mCriteria[j].mAttr.mSource == inputSource) ||
+ (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mix.mCriteria[j].mRule &&
+ mix.mCriteria[j].mAttr.mSource != inputSource)) {
+ if (availDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ if (policyMix != NULL) {
+ *policyMix = &mix;
+ }
+ return AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ }
+ break;
+ }
+ }
+ }
+ return AUDIO_DEVICE_NONE;
+}
+
+status_t AudioPolicyMixCollection::getInputMixForAttr(audio_attributes_t attr, AudioMix *&policyMix)
+{
+ if (strncmp(attr.tags, "addr=", strlen("addr=")) != 0) {
+ return BAD_VALUE;
+ }
+ String8 address(attr.tags + strlen("addr="));
+
+ ssize_t index = indexOfKey(address);
+ if (index < 0) {
+ ALOGW("getInputForAttr() no policy for address %s", address.string());
+ return BAD_VALUE;
+ }
+ sp<AudioPolicyMix> audioPolicyMix = valueAt(index);
+ AudioMix mix = audioPolicyMix->getMix();
+
+ if (mix.mMixType != MIX_TYPE_PLAYERS) {
+ ALOGW("getInputForAttr() bad policy mix type for address %s", address.string());
+ return BAD_VALUE;
+ }
+ policyMix = &mix;
+ return NO_ERROR;
+}
+
+}; //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
new file mode 100644
index 0000000..46a119e
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -0,0 +1,804 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioPort"
+//#define LOG_NDEBUG 0
+
+#include "AudioPort.h"
+#include "HwModule.h"
+#include "AudioGain.h"
+#include "ConfigParsingUtils.h"
+#include "audio_policy_conf.h"
+#include <policy.h>
+
+namespace android {
+
+int32_t volatile AudioPort::mNextUniqueId = 1;
+
+// --- AudioPort class implementation
+
+AudioPort::AudioPort(const String8& name, audio_port_type_t type,
+ audio_port_role_t role, const sp<HwModule>& module) :
+ mName(name), mType(type), mRole(role), mModule(module), mFlags(0), mId(0)
+{
+ mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
+ ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
+}
+
+void AudioPort::attach(const sp<HwModule>& module)
+{
+ mId = getNextUniqueId();
+ mModule = module;
+}
+
+audio_port_handle_t AudioPort::getNextUniqueId()
+{
+ return static_cast<audio_port_handle_t>(android_atomic_inc(&mNextUniqueId));
+}
+
+audio_module_handle_t AudioPort::getModuleHandle() const
+{
+ return mModule->mHandle;
+}
+
+void AudioPort::toAudioPort(struct audio_port *port) const
+{
+ port->role = mRole;
+ port->type = mType;
+ strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
+ unsigned int i;
+ for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
+ if (mSamplingRates[i] != 0) {
+ port->sample_rates[i] = mSamplingRates[i];
+ }
+ }
+ port->num_sample_rates = i;
+ for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
+ if (mChannelMasks[i] != 0) {
+ port->channel_masks[i] = mChannelMasks[i];
+ }
+ }
+ port->num_channel_masks = i;
+ for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
+ if (mFormats[i] != 0) {
+ port->formats[i] = mFormats[i];
+ }
+ }
+ port->num_formats = i;
+
+ ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
+
+ for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
+ port->gains[i] = mGains[i]->mGain;
+ }
+ port->num_gains = i;
+}
+
+void AudioPort::importAudioPort(const sp<AudioPort> port) {
+ for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
+ const uint32_t rate = port->mSamplingRates.itemAt(k);
+ if (rate != 0) { // skip "dynamic" rates
+ bool hasRate = false;
+ for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
+ if (rate == mSamplingRates.itemAt(l)) {
+ hasRate = true;
+ break;
+ }
+ }
+ if (!hasRate) { // never import a sampling rate twice
+ mSamplingRates.add(rate);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
+ const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
+ if (mask != 0) { // skip "dynamic" masks
+ bool hasMask = false;
+ for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
+ if (mask == mChannelMasks.itemAt(l)) {
+ hasMask = true;
+ break;
+ }
+ }
+ if (!hasMask) { // never import a channel mask twice
+ mChannelMasks.add(mask);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
+ const audio_format_t format = port->mFormats.itemAt(k);
+ if (format != 0) { // skip "dynamic" formats
+ bool hasFormat = false;
+ for (size_t l = 0 ; l < mFormats.size() ; l++) {
+ if (format == mFormats.itemAt(l)) {
+ hasFormat = true;
+ break;
+ }
+ }
+ if (!hasFormat) { // never import a channel mask twice
+ mFormats.add(format);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mGains.size() ; k++) {
+ sp<AudioGain> gain = port->mGains.itemAt(k);
+ if (gain != 0) {
+ bool hasGain = false;
+ for (size_t l = 0 ; l < mGains.size() ; l++) {
+ if (gain == mGains.itemAt(l)) {
+ hasGain = true;
+ break;
+ }
+ }
+ if (!hasGain) { // never import a gain twice
+ mGains.add(gain);
+ }
+ }
+ }
+}
+
+void AudioPort::clearCapabilities() {
+ mChannelMasks.clear();
+ mFormats.clear();
+ mSamplingRates.clear();
+ mGains.clear();
+}
+
+void AudioPort::loadSamplingRates(char *name)
+{
+ char *str = strtok(name, "|");
+
+ // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
+ // rates should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mSamplingRates.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ uint32_t rate = atoi(str);
+ if (rate != 0) {
+ ALOGV("loadSamplingRates() adding rate %d", rate);
+ mSamplingRates.add(rate);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadFormats(char *name)
+{
+ char *str = strtok(name, "|");
+
+ // by convention, "0' in the first entry in mFormats indicates the supported formats
+ // should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mFormats.add(AUDIO_FORMAT_DEFAULT);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_format_t format = (audio_format_t)ConfigParsingUtils::stringToEnum(sFormatNameToEnumTable,
+ ARRAY_SIZE(sFormatNameToEnumTable),
+ str);
+ if (format != AUDIO_FORMAT_DEFAULT) {
+ mFormats.add(format);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadInChannels(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadInChannels() %s", name);
+
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mChannelMasks.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_channel_mask_t channelMask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+ ARRAY_SIZE(sInChannelsNameToEnumTable),
+ str);
+ if (channelMask != 0) {
+ ALOGV("loadInChannels() adding channelMask %04x", channelMask);
+ mChannelMasks.add(channelMask);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadOutChannels(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadOutChannels() %s", name);
+
+ // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
+ // masks should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mChannelMasks.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_channel_mask_t channelMask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+ ARRAY_SIZE(sOutChannelsNameToEnumTable),
+ str);
+ if (channelMask != 0) {
+ mChannelMasks.add(channelMask);
+ }
+ str = strtok(NULL, "|");
+ }
+ return;
+}
+
+audio_gain_mode_t AudioPort::loadGainMode(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadGainMode() %s", name);
+ audio_gain_mode_t mode = 0;
+ while (str != NULL) {
+ mode |= (audio_gain_mode_t)ConfigParsingUtils::stringToEnum(sGainModeNameToEnumTable,
+ ARRAY_SIZE(sGainModeNameToEnumTable),
+ str);
+ str = strtok(NULL, "|");
+ }
+ return mode;
+}
+
+void AudioPort::loadGain(cnode *root, int index)
+{
+ cnode *node = root->first_child;
+
+ sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
+
+ while (node) {
+ if (strcmp(node->name, GAIN_MODE) == 0) {
+ gain->mGain.mode = loadGainMode((char *)node->value);
+ } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
+ if (mUseInChannelMask) {
+ gain->mGain.channel_mask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+ ARRAY_SIZE(sInChannelsNameToEnumTable),
+ (char *)node->value);
+ } else {
+ gain->mGain.channel_mask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+ ARRAY_SIZE(sOutChannelsNameToEnumTable),
+ (char *)node->value);
+ }
+ } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
+ gain->mGain.min_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
+ gain->mGain.max_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
+ gain->mGain.default_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
+ gain->mGain.step_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
+ gain->mGain.min_ramp_ms = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
+ gain->mGain.max_ramp_ms = atoi((char *)node->value);
+ }
+ node = node->next;
+ }
+
+ ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
+ gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
+
+ if (gain->mGain.mode == 0) {
+ return;
+ }
+ mGains.add(gain);
+}
+
+void AudioPort::loadGains(cnode *root)
+{
+ cnode *node = root->first_child;
+ int index = 0;
+ while (node) {
+ ALOGV("loadGains() loading gain %s", node->name);
+ loadGain(node, index++);
+ node = node->next;
+ }
+}
+
+status_t AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
+{
+ if (mSamplingRates.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if (mSamplingRates[i] == samplingRate) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
+ uint32_t *updatedSamplingRate) const
+{
+ if (mSamplingRates.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ // Search for the closest supported sampling rate that is above (preferred)
+ // or below (acceptable) the desired sampling rate, within a permitted ratio.
+ // The sampling rates do not need to be sorted in ascending order.
+ ssize_t maxBelow = -1;
+ ssize_t minAbove = -1;
+ uint32_t candidate;
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ candidate = mSamplingRates[i];
+ if (candidate == samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ // candidate < desired
+ if (candidate < samplingRate) {
+ if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
+ maxBelow = i;
+ }
+ // candidate > desired
+ } else {
+ if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
+ minAbove = i;
+ }
+ }
+ }
+ // This uses hard-coded knowledge about AudioFlinger resampling ratios.
+ // TODO Move these assumptions out.
+ static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs
+ static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur
+ // due to approximation by an int32_t of the
+ // phase increments
+ // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
+ if (minAbove >= 0) {
+ candidate = mSamplingRates[minAbove];
+ if (candidate / kMaxDownSampleRatio <= samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ }
+ // But if we have to up-sample from a lower sampling rate, that's OK.
+ if (maxBelow >= 0) {
+ candidate = mSamplingRates[maxBelow];
+ if (candidate * kMaxUpSampleRatio >= samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ }
+ // leave updatedSamplingRate unmodified
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
+{
+ if (mChannelMasks.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ if (mChannelMasks[i] == channelMask) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
+ const
+{
+ if (mChannelMasks.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ // FIXME Does not handle multi-channel automatic conversions yet
+ audio_channel_mask_t supported = mChannelMasks[i];
+ if (supported == channelMask) {
+ return NO_ERROR;
+ }
+ if (isRecordThread) {
+ // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
+ // FIXME Abstract this out to a table.
+ if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
+ && channelMask == AUDIO_CHANNEL_IN_MONO) ||
+ (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
+ || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
+ return NO_ERROR;
+ }
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkFormat(audio_format_t format) const
+{
+ if (mFormats.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mFormats.size(); i ++) {
+ if (mFormats[i] == format) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+
+uint32_t AudioPort::pickSamplingRate() const
+{
+ // special case for uninitialized dynamic profile
+ if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
+ return 0;
+ }
+
+ // For direct outputs, pick minimum sampling rate: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ uint32_t samplingRate = UINT_MAX;
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
+ samplingRate = mSamplingRates[i];
+ }
+ }
+ return (samplingRate == UINT_MAX) ? 0 : samplingRate;
+ }
+
+ uint32_t samplingRate = 0;
+ uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
+
+ // For mixed output and inputs, use max mixer sampling rates. Do not
+ // limit sampling rate otherwise
+ if (mType != AUDIO_PORT_TYPE_MIX) {
+ maxRate = UINT_MAX;
+ }
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
+ samplingRate = mSamplingRates[i];
+ }
+ }
+ return samplingRate;
+}
+
+audio_channel_mask_t AudioPort::pickChannelMask() const
+{
+ // special case for uninitialized dynamic profile
+ if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
+ return AUDIO_CHANNEL_NONE;
+ }
+ audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
+
+ // For direct outputs, pick minimum channel count: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ uint32_t channelCount = UINT_MAX;
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ uint32_t cnlCount;
+ if (mUseInChannelMask) {
+ cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ }
+ if ((cnlCount < channelCount) && (cnlCount > 0)) {
+ channelMask = mChannelMasks[i];
+ channelCount = cnlCount;
+ }
+ }
+ return channelMask;
+ }
+
+ uint32_t channelCount = 0;
+ uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
+
+ // For mixed output and inputs, use max mixer channel count. Do not
+ // limit channel count otherwise
+ if (mType != AUDIO_PORT_TYPE_MIX) {
+ maxCount = UINT_MAX;
+ }
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ uint32_t cnlCount;
+ if (mUseInChannelMask) {
+ cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ }
+ if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
+ channelMask = mChannelMasks[i];
+ channelCount = cnlCount;
+ }
+ }
+ return channelMask;
+}
+
+/* format in order of increasing preference */
+const audio_format_t AudioPort::sPcmFormatCompareTable[] = {
+ AUDIO_FORMAT_DEFAULT,
+ AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_FORMAT_PCM_8_24_BIT,
+ AUDIO_FORMAT_PCM_24_BIT_PACKED,
+ AUDIO_FORMAT_PCM_32_BIT,
+ AUDIO_FORMAT_PCM_FLOAT,
+};
+
+int AudioPort::compareFormats(audio_format_t format1,
+ audio_format_t format2)
+{
+ // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
+ // compressed format and better than any PCM format. This is by design of pickFormat()
+ if (!audio_is_linear_pcm(format1)) {
+ if (!audio_is_linear_pcm(format2)) {
+ return 0;
+ }
+ return 1;
+ }
+ if (!audio_is_linear_pcm(format2)) {
+ return -1;
+ }
+
+ int index1 = -1, index2 = -1;
+ for (size_t i = 0;
+ (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
+ i ++) {
+ if (sPcmFormatCompareTable[i] == format1) {
+ index1 = i;
+ }
+ if (sPcmFormatCompareTable[i] == format2) {
+ index2 = i;
+ }
+ }
+ // format1 not found => index1 < 0 => format2 > format1
+ // format2 not found => index2 < 0 => format2 < format1
+ return index1 - index2;
+}
+
+audio_format_t AudioPort::pickFormat() const
+{
+ // special case for uninitialized dynamic profile
+ if (mFormats.size() == 1 && mFormats[0] == 0) {
+ return AUDIO_FORMAT_DEFAULT;
+ }
+
+ audio_format_t format = AUDIO_FORMAT_DEFAULT;
+ audio_format_t bestFormat =
+ AudioPort::sPcmFormatCompareTable[
+ ARRAY_SIZE(AudioPort::sPcmFormatCompareTable) - 1];
+ // For mixed output and inputs, use best mixer output format. Do not
+ // limit format otherwise
+ if ((mType != AUDIO_PORT_TYPE_MIX) ||
+ ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
+ bestFormat = AUDIO_FORMAT_INVALID;
+ }
+
+ for (size_t i = 0; i < mFormats.size(); i ++) {
+ if ((compareFormats(mFormats[i], format) > 0) &&
+ (compareFormats(mFormats[i], bestFormat) <= 0)) {
+ format = mFormats[i];
+ }
+ }
+ return format;
+}
+
+status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig,
+ int index) const
+{
+ if (index < 0 || (size_t)index >= mGains.size()) {
+ return BAD_VALUE;
+ }
+ return mGains[index]->checkConfig(gainConfig);
+}
+
+void AudioPort::dump(int fd, int spaces) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ if (mName.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
+ result.append(buffer);
+ }
+
+ if (mSamplingRates.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ if (i == 0 && mSamplingRates[i] == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+ }
+ result.append(buffer);
+ result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+
+ if (mChannelMasks.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
+
+ if (i == 0 && mChannelMasks[i] == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+ }
+ result.append(buffer);
+ result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+
+ if (mFormats.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mFormats.size(); i++) {
+ const char *formatStr = ConfigParsingUtils::enumToString(sFormatNameToEnumTable,
+ ARRAY_SIZE(sFormatNameToEnumTable),
+ mFormats[i]);
+ if (i == 0 && strcmp(formatStr, "") == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "%s", formatStr);
+ }
+ result.append(buffer);
+ result.append(i == (mFormats.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+ write(fd, result.string(), result.size());
+ if (mGains.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
+ write(fd, buffer, strlen(buffer) + 1);
+ result.append(buffer);
+ for (size_t i = 0; i < mGains.size(); i++) {
+ mGains[i]->dump(fd, spaces + 2, i);
+ }
+ }
+}
+
+
+// --- AudioPortConfig class implementation
+
+AudioPortConfig::AudioPortConfig()
+{
+ mSamplingRate = 0;
+ mChannelMask = AUDIO_CHANNEL_NONE;
+ mFormat = AUDIO_FORMAT_INVALID;
+ mGain.index = -1;
+}
+
+status_t AudioPortConfig::applyAudioPortConfig(
+ const struct audio_port_config *config,
+ struct audio_port_config *backupConfig)
+{
+ struct audio_port_config localBackupConfig;
+ status_t status = NO_ERROR;
+
+ localBackupConfig.config_mask = config->config_mask;
+ toAudioPortConfig(&localBackupConfig);
+
+ sp<AudioPort> audioport = getAudioPort();
+ if (audioport == 0) {
+ status = NO_INIT;
+ goto exit;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ status = audioport->checkExactSamplingRate(config->sample_rate);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mSamplingRate = config->sample_rate;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ status = audioport->checkExactChannelMask(config->channel_mask);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mChannelMask = config->channel_mask;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ status = audioport->checkFormat(config->format);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mFormat = config->format;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ status = audioport->checkGain(&config->gain, config->gain.index);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mGain = config->gain;
+ }
+
+exit:
+ if (status != NO_ERROR) {
+ applyAudioPortConfig(&localBackupConfig);
+ }
+ if (backupConfig != NULL) {
+ *backupConfig = localBackupConfig;
+ }
+ return status;
+}
+
+void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ dstConfig->sample_rate = mSamplingRate;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
+ dstConfig->sample_rate = srcConfig->sample_rate;
+ }
+ } else {
+ dstConfig->sample_rate = 0;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ dstConfig->channel_mask = mChannelMask;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
+ dstConfig->channel_mask = srcConfig->channel_mask;
+ }
+ } else {
+ dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ dstConfig->format = mFormat;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
+ dstConfig->format = srcConfig->format;
+ }
+ } else {
+ dstConfig->format = AUDIO_FORMAT_INVALID;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ dstConfig->gain = mGain;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
+ dstConfig->gain = srcConfig->gain;
+ }
+ } else {
+ dstConfig->gain.index = -1;
+ }
+ if (dstConfig->gain.index != -1) {
+ dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
+ } else {
+ dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
+ }
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp
new file mode 100644
index 0000000..fe5bc5f
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp
@@ -0,0 +1,288 @@
+/*
+ * Copyright (C) 2015 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 "APM::ConfigParsingUtils"
+//#define LOG_NDEBUG 0
+
+#include "ConfigParsingUtils.h"
+#include "AudioGain.h"
+#include <hardware/audio.h>
+#include <utils/Log.h>
+#include <cutils/misc.h>
+
+namespace android {
+
+//static
+uint32_t ConfigParsingUtils::stringToEnum(const struct StringToEnum *table,
+ size_t size,
+ const char *name)
+{
+ for (size_t i = 0; i < size; i++) {
+ if (strcmp(table[i].name, name) == 0) {
+ ALOGV("stringToEnum() found %s", table[i].name);
+ return table[i].value;
+ }
+ }
+ return 0;
+}
+
+//static
+const char *ConfigParsingUtils::enumToString(const struct StringToEnum *table,
+ size_t size,
+ uint32_t value)
+{
+ for (size_t i = 0; i < size; i++) {
+ if (table[i].value == value) {
+ return table[i].name;
+ }
+ }
+ return "";
+}
+
+//static
+bool ConfigParsingUtils::stringToBool(const char *value)
+{
+ return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
+}
+
+
+// --- audio_policy.conf file parsing
+//static
+uint32_t ConfigParsingUtils::parseOutputFlagNames(char *name)
+{
+ uint32_t flag = 0;
+
+ // it is OK to cast name to non const here as we are not going to use it after
+ // strtok() modifies it
+ char *flagName = strtok(name, "|");
+ while (flagName != NULL) {
+ if (strlen(flagName) != 0) {
+ flag |= ConfigParsingUtils::stringToEnum(sOutputFlagNameToEnumTable,
+ ARRAY_SIZE(sOutputFlagNameToEnumTable),
+ flagName);
+ }
+ flagName = strtok(NULL, "|");
+ }
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flag |= AUDIO_OUTPUT_FLAG_DIRECT;
+ }
+
+ return flag;
+}
+
+//static
+uint32_t ConfigParsingUtils::parseInputFlagNames(char *name)
+{
+ uint32_t flag = 0;
+
+ // it is OK to cast name to non const here as we are not going to use it after
+ // strtok() modifies it
+ char *flagName = strtok(name, "|");
+ while (flagName != NULL) {
+ if (strlen(flagName) != 0) {
+ flag |= stringToEnum(sInputFlagNameToEnumTable,
+ ARRAY_SIZE(sInputFlagNameToEnumTable),
+ flagName);
+ }
+ flagName = strtok(NULL, "|");
+ }
+ return flag;
+}
+
+//static
+audio_devices_t ConfigParsingUtils::parseDeviceNames(char *name)
+{
+ uint32_t device = 0;
+
+ char *devName = strtok(name, "|");
+ while (devName != NULL) {
+ if (strlen(devName) != 0) {
+ device |= stringToEnum(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ devName);
+ }
+ devName = strtok(NULL, "|");
+ }
+ return device;
+}
+
+//static
+void ConfigParsingUtils::loadHwModule(cnode *root, HwModuleCollection &hwModules,
+ DeviceVector &availableInputDevices,
+ DeviceVector &availableOutputDevices,
+ sp<DeviceDescriptor> &defaultOutputDevices,
+ bool &isSpeakerDrcEnable)
+{
+ status_t status = NAME_NOT_FOUND;
+ cnode *node;
+ sp<HwModule> module = new HwModule(root->name);
+
+ node = config_find(root, DEVICES_TAG);
+ if (node != NULL) {
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModule() loading device %s", node->name);
+ status_t tmpStatus = module->loadDevice(node);
+ if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+ status = tmpStatus;
+ }
+ node = node->next;
+ }
+ }
+ node = config_find(root, OUTPUTS_TAG);
+ if (node != NULL) {
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModule() loading output %s", node->name);
+ status_t tmpStatus = module->loadOutput(node);
+ if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+ status = tmpStatus;
+ }
+ node = node->next;
+ }
+ }
+ node = config_find(root, INPUTS_TAG);
+ if (node != NULL) {
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModule() loading input %s", node->name);
+ status_t tmpStatus = module->loadInput(node);
+ if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+ status = tmpStatus;
+ }
+ node = node->next;
+ }
+ }
+ loadGlobalConfig(root, module, availableInputDevices, availableOutputDevices,
+ defaultOutputDevices, isSpeakerDrcEnable);
+
+ if (status == NO_ERROR) {
+ hwModules.add(module);
+ }
+}
+
+//static
+void ConfigParsingUtils::loadHwModules(cnode *root, HwModuleCollection &hwModules,
+ DeviceVector &availableInputDevices,
+ DeviceVector &availableOutputDevices,
+ sp<DeviceDescriptor> &defaultOutputDevices,
+ bool &isSpeakerDrcEnabled)
+{
+ cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
+ if (node == NULL) {
+ return;
+ }
+
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModules() loading module %s", node->name);
+ loadHwModule(node, hwModules, availableInputDevices, availableOutputDevices,
+ defaultOutputDevices, isSpeakerDrcEnabled);
+ node = node->next;
+ }
+}
+
+//static
+void ConfigParsingUtils::loadGlobalConfig(cnode *root, const sp<HwModule>& module,
+ DeviceVector &availableInputDevices,
+ DeviceVector &availableOutputDevices,
+ sp<DeviceDescriptor> &defaultOutputDevice,
+ bool &speakerDrcEnabled)
+{
+ cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
+
+ if (node == NULL) {
+ return;
+ }
+ DeviceVector declaredDevices;
+ if (module != NULL) {
+ declaredDevices = module->mDeclaredDevices;
+ }
+
+ node = node->first_child;
+ while (node) {
+ if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
+ availableOutputDevices.loadDevicesFromName((char *)node->value,
+ declaredDevices);
+ ALOGV("loadGlobalConfig() Attached Output Devices %08x",
+ availableOutputDevices.types());
+ } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
+ audio_devices_t device = (audio_devices_t)stringToEnum(
+ sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ (char *)node->value);
+ if (device != AUDIO_DEVICE_NONE) {
+ defaultOutputDevice = new DeviceDescriptor(String8("default-output"), device);
+ } else {
+ ALOGW("loadGlobalConfig() default device not specified");
+ }
+ ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", defaultOutputDevice->type());
+ } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
+ availableInputDevices.loadDevicesFromName((char *)node->value,
+ declaredDevices);
+ ALOGV("loadGlobalConfig() Available InputDevices %08x", availableInputDevices.types());
+ } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
+ speakerDrcEnabled = stringToBool((char *)node->value);
+ ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", speakerDrcEnabled);
+ } else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) {
+ uint32_t major, minor;
+ sscanf((char *)node->value, "%u.%u", &major, &minor);
+ module->mHalVersion = HARDWARE_DEVICE_API_VERSION(major, minor);
+ ALOGV("loadGlobalConfig() mHalVersion = %04x major %u minor %u",
+ module->mHalVersion, major, minor);
+ }
+ node = node->next;
+ }
+}
+
+//static
+status_t ConfigParsingUtils::loadAudioPolicyConfig(const char *path,
+ HwModuleCollection &hwModules,
+ DeviceVector &availableInputDevices,
+ DeviceVector &availableOutputDevices,
+ sp<DeviceDescriptor> &defaultOutputDevices,
+ bool &isSpeakerDrcEnabled)
+{
+ cnode *root;
+ char *data;
+
+ data = (char *)load_file(path, NULL);
+ if (data == NULL) {
+ return -ENODEV;
+ }
+ root = config_node("", "");
+ config_load(root, data);
+
+ loadHwModules(root, hwModules,
+ availableInputDevices, availableOutputDevices,
+ defaultOutputDevices, isSpeakerDrcEnabled);
+ // legacy audio_policy.conf files have one global_configuration section
+ loadGlobalConfig(root, hwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY),
+ availableInputDevices, availableOutputDevices,
+ defaultOutputDevices, isSpeakerDrcEnabled);
+ config_free(root);
+ free(root);
+ free(data);
+
+ ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
+
+ return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
new file mode 100644
index 0000000..7df7d75
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -0,0 +1,320 @@
+/*
+ * Copyright (C) 2015 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 "APM::Devices"
+//#define LOG_NDEBUG 0
+
+#include "DeviceDescriptor.h"
+#include "AudioGain.h"
+#include "HwModule.h"
+#include "ConfigParsingUtils.h"
+
+namespace android {
+
+String8 DeviceDescriptor::emptyNameStr = String8("");
+
+DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) :
+ AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
+ audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
+ AUDIO_PORT_ROLE_SOURCE,
+ NULL),
+ mAddress(""), mDeviceType(type)
+{
+
+}
+
+bool DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
+{
+ // Devices are considered equal if they:
+ // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
+ // - have the same address or one device does not specify the address
+ // - have the same channel mask or one device does not specify the channel mask
+ return (mDeviceType == other->mDeviceType) &&
+ (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
+ (mChannelMask == 0 || other->mChannelMask == 0 ||
+ mChannelMask == other->mChannelMask);
+}
+
+void DeviceDescriptor::loadGains(cnode *root)
+{
+ AudioPort::loadGains(root);
+ if (mGains.size() > 0) {
+ mGains[0]->getDefaultConfig(&mGain);
+ }
+}
+
+void DeviceVector::refreshTypes()
+{
+ mDeviceTypes = AUDIO_DEVICE_NONE;
+ for(size_t i = 0; i < size(); i++) {
+ mDeviceTypes |= itemAt(i)->type();
+ }
+ ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
+}
+
+ssize_t DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
+{
+ for(size_t i = 0; i < size(); i++) {
+ if (item->equals(itemAt(i))) {
+ return i;
+ }
+ }
+ return -1;
+}
+
+ssize_t DeviceVector::add(const sp<DeviceDescriptor>& item)
+{
+ ssize_t ret = indexOf(item);
+
+ if (ret < 0) {
+ ret = SortedVector::add(item);
+ if (ret >= 0) {
+ refreshTypes();
+ }
+ } else {
+ ALOGW("DeviceVector::add device %08x already in", item->type());
+ ret = -1;
+ }
+ return ret;
+}
+
+ssize_t DeviceVector::remove(const sp<DeviceDescriptor>& item)
+{
+ size_t i;
+ ssize_t ret = indexOf(item);
+
+ if (ret < 0) {
+ ALOGW("DeviceVector::remove device %08x not in", item->type());
+ } else {
+ ret = SortedVector::removeAt(ret);
+ if (ret >= 0) {
+ refreshTypes();
+ }
+ }
+ return ret;
+}
+
+audio_devices_t DeviceVector::getDevicesFromHwModule(audio_module_handle_t moduleHandle) const
+{
+ audio_devices_t devices = AUDIO_DEVICE_NONE;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->getModuleHandle() == moduleHandle) {
+ devices |= itemAt(i)->type();
+ }
+ }
+ return devices;
+}
+
+void DeviceVector::loadDevicesFromType(audio_devices_t types)
+{
+ DeviceVector deviceList;
+
+ uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
+ types &= ~role_bit;
+
+ while (types) {
+ uint32_t i = 31 - __builtin_clz(types);
+ uint32_t type = 1 << i;
+ types &= ~type;
+ add(new DeviceDescriptor(String8("device_type"), type | role_bit));
+ }
+}
+
+void DeviceVector::loadDevicesFromName(char *name,
+ const DeviceVector& declaredDevices)
+{
+ char *devName = strtok(name, "|");
+ while (devName != NULL) {
+ if (strlen(devName) != 0) {
+ audio_devices_t type = ConfigParsingUtils::stringToEnum(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ devName);
+ if (type != AUDIO_DEVICE_NONE) {
+ sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(name), type);
+ if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
+ type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
+ dev->mAddress = String8("0");
+ }
+ add(dev);
+ } else {
+ sp<DeviceDescriptor> deviceDesc =
+ declaredDevices.getDeviceFromName(String8(devName));
+ if (deviceDesc != 0) {
+ add(deviceDesc);
+ }
+ }
+ }
+ devName = strtok(NULL, "|");
+ }
+}
+
+sp<DeviceDescriptor> DeviceVector::getDevice(audio_devices_t type, String8 address) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->type() == type) {
+ if (address == "" || itemAt(i)->mAddress == address) {
+ device = itemAt(i);
+ if (itemAt(i)->mAddress == address) {
+ break;
+ }
+ }
+ }
+ }
+ ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
+ type, address.string(), device.get());
+ return device;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromId(audio_port_handle_t id) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->getHandle() == id) {
+ device = itemAt(i);
+ break;
+ }
+ }
+ return device;
+}
+
+DeviceVector DeviceVector::getDevicesFromType(audio_devices_t type) const
+{
+ DeviceVector devices;
+ bool isOutput = audio_is_output_devices(type);
+ type &= ~AUDIO_DEVICE_BIT_IN;
+ for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
+ bool curIsOutput = audio_is_output_devices(itemAt(i)->mDeviceType);
+ audio_devices_t curType = itemAt(i)->mDeviceType & ~AUDIO_DEVICE_BIT_IN;
+ if ((isOutput == curIsOutput) && ((type & curType) != 0)) {
+ devices.add(itemAt(i));
+ type &= ~curType;
+ ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
+ itemAt(i)->type(), itemAt(i).get());
+ }
+ }
+ return devices;
+}
+
+DeviceVector DeviceVector::getDevicesFromTypeAddr(
+ audio_devices_t type, String8 address) const
+{
+ DeviceVector devices;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->type() == type) {
+ if (itemAt(i)->mAddress == address) {
+ devices.add(itemAt(i));
+ }
+ }
+ }
+ return devices;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromName(const String8& name) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->mName == name) {
+ device = itemAt(i);
+ break;
+ }
+ }
+ return device;
+}
+
+
+status_t DeviceVector::dump(int fd, const String8 &direction) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "\n Available %s devices:\n", direction.string());
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < size(); i++) {
+ itemAt(i)->dump(fd, 2, i);
+ }
+ return NO_ERROR;
+}
+
+audio_policy_dev_state_t DeviceVector::getDeviceConnectionState(const sp<DeviceDescriptor> &devDesc) const
+{
+ ssize_t index = indexOf(devDesc);
+ return index >= 0 ? AUDIO_POLICY_DEVICE_STATE_AVAILABLE : AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+}
+
+void DeviceDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = audio_is_output_device(mDeviceType) ?
+ AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
+ dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
+ dstConfig->ext.device.type = mDeviceType;
+
+ //TODO Understand why this test is necessary. i.e. why at boot time does it crash
+ // without the test?
+ // This has been demonstrated to NOT be true (at start up)
+ // ALOG_ASSERT(mModule != NULL);
+ dstConfig->ext.device.hw_module = mModule != 0 ? mModule->mHandle : AUDIO_IO_HANDLE_NONE;
+ strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+void DeviceDescriptor::toAudioPort(struct audio_port *port) const
+{
+ ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
+ AudioPort::toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.device.type = mDeviceType;
+ port->ext.device.hw_module = mModule->mHandle;
+ strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+status_t DeviceDescriptor::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ if (mId != 0) {
+ snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
+ result.append(buffer);
+ }
+ snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
+ ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mDeviceType));
+ result.append(buffer);
+ if (mAddress.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
+ result.append(buffer);
+ }
+ write(fd, result.string(), result.size());
+ AudioPort::dump(fd, spaces);
+
+ return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/EffectDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/EffectDescriptor.cpp
new file mode 100644
index 0000000..33d838d
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/EffectDescriptor.cpp
@@ -0,0 +1,192 @@
+/*
+ * Copyright (C) 2015 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 "APM::EffectDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "EffectDescriptor.h"
+#include <utils/String8.h>
+
+namespace android {
+
+status_t EffectDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " I/O: %d\n", mIo);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Session: %d\n", mSession);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Name: %s\n", mDesc.name);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled");
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+EffectDescriptorCollection::EffectDescriptorCollection() :
+ mTotalEffectsCpuLoad(0),
+ mTotalEffectsMemory(0)
+{
+
+}
+
+status_t EffectDescriptorCollection::registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id)
+{
+ if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
+ ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
+ desc->name, desc->memoryUsage);
+ return INVALID_OPERATION;
+ }
+ mTotalEffectsMemory += desc->memoryUsage;
+ ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
+ desc->name, io, strategy, session, id);
+ ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
+
+ sp<EffectDescriptor> effectDesc = new EffectDescriptor();
+ memcpy (&effectDesc->mDesc, desc, sizeof(effect_descriptor_t));
+ effectDesc->mIo = io;
+ effectDesc->mStrategy = static_cast<routing_strategy>(strategy);
+ effectDesc->mSession = session;
+ effectDesc->mEnabled = false;
+
+ add(id, effectDesc);
+
+ return NO_ERROR;
+}
+
+status_t EffectDescriptorCollection::unregisterEffect(int id)
+{
+ ssize_t index = indexOfKey(id);
+ if (index < 0) {
+ ALOGW("unregisterEffect() unknown effect ID %d", id);
+ return INVALID_OPERATION;
+ }
+
+ sp<EffectDescriptor> effectDesc = valueAt(index);
+
+ setEffectEnabled(effectDesc, false);
+
+ if (mTotalEffectsMemory < effectDesc->mDesc.memoryUsage) {
+ ALOGW("unregisterEffect() memory %d too big for total %d",
+ effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+ effectDesc->mDesc.memoryUsage = mTotalEffectsMemory;
+ }
+ mTotalEffectsMemory -= effectDesc->mDesc.memoryUsage;
+ ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
+ effectDesc->mDesc.name, id, effectDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+
+ removeItem(id);
+
+ return NO_ERROR;
+}
+
+status_t EffectDescriptorCollection::setEffectEnabled(int id, bool enabled)
+{
+ ssize_t index = indexOfKey(id);
+ if (index < 0) {
+ ALOGW("unregisterEffect() unknown effect ID %d", id);
+ return INVALID_OPERATION;
+ }
+
+ return setEffectEnabled(valueAt(index), enabled);
+}
+
+
+status_t EffectDescriptorCollection::setEffectEnabled(const sp<EffectDescriptor> &effectDesc,
+ bool enabled)
+{
+ if (enabled == effectDesc->mEnabled) {
+ ALOGV("setEffectEnabled(%s) effect already %s",
+ enabled?"true":"false", enabled?"enabled":"disabled");
+ return INVALID_OPERATION;
+ }
+
+ if (enabled) {
+ if (mTotalEffectsCpuLoad + effectDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
+ ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
+ effectDesc->mDesc.name, (float)effectDesc->mDesc.cpuLoad/10);
+ return INVALID_OPERATION;
+ }
+ mTotalEffectsCpuLoad += effectDesc->mDesc.cpuLoad;
+ ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
+ } else {
+ if (mTotalEffectsCpuLoad < effectDesc->mDesc.cpuLoad) {
+ ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
+ effectDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
+ effectDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
+ }
+ mTotalEffectsCpuLoad -= effectDesc->mDesc.cpuLoad;
+ ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
+ }
+ effectDesc->mEnabled = enabled;
+ return NO_ERROR;
+}
+
+bool EffectDescriptorCollection::isNonOffloadableEffectEnabled()
+{
+ for (size_t i = 0; i < size(); i++) {
+ sp<EffectDescriptor> effectDesc = valueAt(i);
+ if (effectDesc->mEnabled && (effectDesc->mStrategy == STRATEGY_MEDIA) &&
+ ((effectDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
+ ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
+ effectDesc->mDesc.name, effectDesc->mSession);
+ return true;
+ }
+ }
+ return false;
+}
+
+uint32_t EffectDescriptorCollection::getMaxEffectsCpuLoad() const
+{
+ return MAX_EFFECTS_CPU_LOAD;
+}
+
+uint32_t EffectDescriptorCollection::getMaxEffectsMemory() const
+{
+ return MAX_EFFECTS_MEMORY;
+}
+
+status_t EffectDescriptorCollection::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
+ (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
+ write(fd, buffer, strlen(buffer));
+
+ snprintf(buffer, SIZE, "Registered effects:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < size(); i++) {
+ snprintf(buffer, SIZE, "- Effect %d dump:\n", keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ valueAt(i)->dump(fd);
+ }
+ return NO_ERROR;
+}
+
+}; //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
new file mode 100644
index 0000000..0097d69
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -0,0 +1,371 @@
+/*
+ * Copyright (C) 2015 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 "APM::HwModule"
+//#define LOG_NDEBUG 0
+
+#include "HwModule.h"
+#include "IOProfile.h"
+#include "AudioGain.h"
+#include "ConfigParsingUtils.h"
+#include "audio_policy_conf.h"
+#include <hardware/audio.h>
+#include <policy.h>
+
+namespace android {
+
+HwModule::HwModule(const char *name)
+ : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
+ mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
+{
+}
+
+HwModule::~HwModule()
+{
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ mOutputProfiles[i]->mSupportedDevices.clear();
+ }
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ mInputProfiles[i]->mSupportedDevices.clear();
+ }
+ free((void *)mName);
+}
+
+status_t HwModule::loadInput(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
+
+ while (node) {
+ if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+ profile->loadSamplingRates((char *)node->value);
+ } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+ profile->loadFormats((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ profile->loadInChannels((char *)node->value);
+ } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+ profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+ mDeclaredDevices);
+ } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+ profile->mFlags = ConfigParsingUtils::parseInputFlagNames((char *)node->value);
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ profile->loadGains(node);
+ }
+ node = node->next;
+ }
+ ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+ "loadInput() invalid supported devices");
+ ALOGW_IF(profile->mChannelMasks.size() == 0,
+ "loadInput() invalid supported channel masks");
+ ALOGW_IF(profile->mSamplingRates.size() == 0,
+ "loadInput() invalid supported sampling rates");
+ ALOGW_IF(profile->mFormats.size() == 0,
+ "loadInput() invalid supported formats");
+ if (!profile->mSupportedDevices.isEmpty() &&
+ (profile->mChannelMasks.size() != 0) &&
+ (profile->mSamplingRates.size() != 0) &&
+ (profile->mFormats.size() != 0)) {
+
+ ALOGV("loadInput() adding input Supported Devices %04x",
+ profile->mSupportedDevices.types());
+
+ mInputProfiles.add(profile);
+ return NO_ERROR;
+ } else {
+ return BAD_VALUE;
+ }
+}
+
+status_t HwModule::loadOutput(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
+
+ while (node) {
+ if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+ profile->loadSamplingRates((char *)node->value);
+ } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+ profile->loadFormats((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ profile->loadOutChannels((char *)node->value);
+ } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+ profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+ mDeclaredDevices);
+ } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+ profile->mFlags = ConfigParsingUtils::parseOutputFlagNames((char *)node->value);
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ profile->loadGains(node);
+ }
+ node = node->next;
+ }
+ ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+ "loadOutput() invalid supported devices");
+ ALOGW_IF(profile->mChannelMasks.size() == 0,
+ "loadOutput() invalid supported channel masks");
+ ALOGW_IF(profile->mSamplingRates.size() == 0,
+ "loadOutput() invalid supported sampling rates");
+ ALOGW_IF(profile->mFormats.size() == 0,
+ "loadOutput() invalid supported formats");
+ if (!profile->mSupportedDevices.isEmpty() &&
+ (profile->mChannelMasks.size() != 0) &&
+ (profile->mSamplingRates.size() != 0) &&
+ (profile->mFormats.size() != 0)) {
+
+ ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
+ profile->mSupportedDevices.types(), profile->mFlags);
+
+ mOutputProfiles.add(profile);
+ return NO_ERROR;
+ } else {
+ return BAD_VALUE;
+ }
+}
+
+status_t HwModule::loadDevice(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ audio_devices_t type = AUDIO_DEVICE_NONE;
+ while (node) {
+ if (strcmp(node->name, APM_DEVICE_TYPE) == 0) {
+ type = ConfigParsingUtils::parseDeviceNames((char *)node->value);
+ break;
+ }
+ node = node->next;
+ }
+ if (type == AUDIO_DEVICE_NONE ||
+ (!audio_is_input_device(type) && !audio_is_output_device(type))) {
+ ALOGW("loadDevice() bad type %08x", type);
+ return BAD_VALUE;
+ }
+ sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
+ deviceDesc->mModule = this;
+
+ node = root->first_child;
+ while (node) {
+ if (strcmp(node->name, APM_DEVICE_ADDRESS) == 0) {
+ deviceDesc->mAddress = String8((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ if (audio_is_input_device(type)) {
+ deviceDesc->loadInChannels((char *)node->value);
+ } else {
+ deviceDesc->loadOutChannels((char *)node->value);
+ }
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ deviceDesc->loadGains(node);
+ }
+ node = node->next;
+ }
+
+ ALOGV("loadDevice() adding device name %s type %08x address %s",
+ deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
+
+ mDeclaredDevices.add(deviceDesc);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::addOutputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address)
+{
+ sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this);
+
+ profile->mSamplingRates.add(config->sample_rate);
+ profile->mChannelMasks.add(config->channel_mask);
+ profile->mFormats.add(config->format);
+
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+ devDesc->mAddress = address;
+ profile->mSupportedDevices.add(devDesc);
+
+ mOutputProfiles.add(profile);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::removeOutputProfile(String8 name)
+{
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ if (mOutputProfiles[i]->mName == name) {
+ mOutputProfiles.removeAt(i);
+ break;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+status_t HwModule::addInputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address)
+{
+ sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this);
+
+ profile->mSamplingRates.add(config->sample_rate);
+ profile->mChannelMasks.add(config->channel_mask);
+ profile->mFormats.add(config->format);
+
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+ devDesc->mAddress = address;
+ profile->mSupportedDevices.add(devDesc);
+
+ ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
+
+ mInputProfiles.add(profile);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::removeInputProfile(String8 name)
+{
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ if (mInputProfiles[i]->mName == name) {
+ mInputProfiles.removeAt(i);
+ break;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+
+void HwModule::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " - name: %s\n", mName);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ if (mOutputProfiles.size()) {
+ write(fd, " - outputs:\n", strlen(" - outputs:\n"));
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ snprintf(buffer, SIZE, " output %zu:\n", i);
+ write(fd, buffer, strlen(buffer));
+ mOutputProfiles[i]->dump(fd);
+ }
+ }
+ if (mInputProfiles.size()) {
+ write(fd, " - inputs:\n", strlen(" - inputs:\n"));
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ snprintf(buffer, SIZE, " input %zu:\n", i);
+ write(fd, buffer, strlen(buffer));
+ mInputProfiles[i]->dump(fd);
+ }
+ }
+ if (mDeclaredDevices.size()) {
+ write(fd, " - devices:\n", strlen(" - devices:\n"));
+ for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
+ mDeclaredDevices[i]->dump(fd, 4, i);
+ }
+ }
+}
+
+sp <HwModule> HwModuleCollection::getModuleFromName(const char *name) const
+{
+ sp <HwModule> module;
+
+ for (size_t i = 0; i < size(); i++)
+ {
+ if (strcmp(itemAt(i)->mName, name) == 0) {
+ return itemAt(i);
+ }
+ }
+ return module;
+}
+
+
+sp <HwModule> HwModuleCollection::getModuleForDevice(audio_devices_t device) const
+{
+ sp <HwModule> module;
+
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->mHandle == 0) {
+ continue;
+ }
+ if (audio_is_output_device(device)) {
+ for (size_t j = 0; j < itemAt(i)->mOutputProfiles.size(); j++)
+ {
+ if (itemAt(i)->mOutputProfiles[j]->mSupportedDevices.types() & device) {
+ return itemAt(i);
+ }
+ }
+ } else {
+ for (size_t j = 0; j < itemAt(i)->mInputProfiles.size(); j++) {
+ if (itemAt(i)->mInputProfiles[j]->mSupportedDevices.types() &
+ device & ~AUDIO_DEVICE_BIT_IN) {
+ return itemAt(i);
+ }
+ }
+ }
+ }
+ return module;
+}
+
+sp<DeviceDescriptor> HwModuleCollection::getDeviceDescriptor(const audio_devices_t device,
+ const char *device_address,
+ const char *device_name) const
+{
+ String8 address = (device_address == NULL) ? String8("") : String8(device_address);
+ // handle legacy remote submix case where the address was not always specified
+ if (device_distinguishes_on_address(device) && (address.length() == 0)) {
+ address = String8("0");
+ }
+
+ for (size_t i = 0; i < size(); i++) {
+ const sp<HwModule> hwModule = itemAt(i);
+ if (hwModule->mHandle == 0) {
+ continue;
+ }
+ DeviceVector deviceList =
+ hwModule->mDeclaredDevices.getDevicesFromTypeAddr(device, address);
+ if (!deviceList.isEmpty()) {
+ return deviceList.itemAt(0);
+ }
+ deviceList = hwModule->mDeclaredDevices.getDevicesFromType(device);
+ if (!deviceList.isEmpty()) {
+ return deviceList.itemAt(0);
+ }
+ }
+
+ sp<DeviceDescriptor> devDesc =
+ new DeviceDescriptor(String8(device_name != NULL ? device_name : ""), device);
+ devDesc->mAddress = address;
+ return devDesc;
+}
+
+status_t HwModuleCollection::dump(int fd) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "\nHW Modules dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < size(); i++) {
+ snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1);
+ write(fd, buffer, strlen(buffer));
+ itemAt(i)->dump(fd);
+ }
+ return NO_ERROR;
+}
+
+} //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
new file mode 100644
index 0000000..376dd22
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
@@ -0,0 +1,149 @@
+/*
+ * Copyright (C) 2015 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 "APM::IOProfile"
+//#define LOG_NDEBUG 0
+
+#include "IOProfile.h"
+#include "HwModule.h"
+#include "AudioGain.h"
+
+namespace android {
+
+IOProfile::IOProfile(const String8& name, audio_port_role_t role,
+ const sp<HwModule>& module)
+ : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
+{
+}
+
+IOProfile::~IOProfile()
+{
+}
+
+// checks if the IO profile is compatible with specified parameters.
+// Sampling rate, format and channel mask must be specified in order to
+// get a valid a match
+bool IOProfile::isCompatibleProfile(audio_devices_t device,
+ String8 address,
+ uint32_t samplingRate,
+ uint32_t *updatedSamplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ uint32_t flags) const
+{
+ const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
+ const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+ ALOG_ASSERT(isPlaybackThread != isRecordThread);
+
+
+ if (device != AUDIO_DEVICE_NONE) {
+ // just check types if multiple devices are selected
+ if (popcount(device & ~AUDIO_DEVICE_BIT_IN) > 1) {
+ if ((mSupportedDevices.types() & device) != device) {
+ return false;
+ }
+ } else if (mSupportedDevices.getDevice(device, address) == 0) {
+ return false;
+ }
+ }
+
+ if (samplingRate == 0) {
+ return false;
+ }
+ uint32_t myUpdatedSamplingRate = samplingRate;
+ if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
+ return false;
+ }
+ if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
+ NO_ERROR) {
+ return false;
+ }
+
+ if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
+ return false;
+ }
+
+ if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
+ checkExactChannelMask(channelMask) != NO_ERROR)) {
+ return false;
+ }
+ if (isRecordThread && (!audio_is_input_channel(channelMask) ||
+ checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
+ return false;
+ }
+
+ if (isPlaybackThread && (mFlags & flags) != flags) {
+ return false;
+ }
+ // The only input flag that is allowed to be different is the fast flag.
+ // An existing fast stream is compatible with a normal track request.
+ // An existing normal stream is compatible with a fast track request,
+ // but the fast request will be denied by AudioFlinger and converted to normal track.
+ if (isRecordThread && ((mFlags ^ flags) &
+ ~AUDIO_INPUT_FLAG_FAST)) {
+ return false;
+ }
+
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = myUpdatedSamplingRate;
+ }
+ return true;
+}
+
+void IOProfile::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ AudioPort::dump(fd, 4);
+
+ snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - devices:\n");
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ for (size_t i = 0; i < mSupportedDevices.size(); i++) {
+ mSupportedDevices[i]->dump(fd, 6, i);
+ }
+}
+
+void IOProfile::log()
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ ALOGV(" - sampling rates: ");
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ ALOGV(" %d", mSamplingRates[i]);
+ }
+
+ ALOGV(" - channel masks: ");
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ ALOGV(" 0x%04x", mChannelMasks[i]);
+ }
+
+ ALOGV(" - formats: ");
+ for (size_t i = 0; i < mFormats.size(); i++) {
+ ALOGV(" 0x%08x", mFormats[i]);
+ }
+
+ ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types());
+ ALOGV(" - flags: 0x%04x\n", mFlags);
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/SoundTriggerSession.cpp b/services/audiopolicy/common/managerdefinitions/src/SoundTriggerSession.cpp
new file mode 100644
index 0000000..8ca3ae0
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/SoundTriggerSession.cpp
@@ -0,0 +1,45 @@
+/*
+ * Copyright (C) 2015 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 "APM::SoundTriggerSession"
+//#define LOG_NDEBUG 0
+
+#include "SoundTriggerSession.h"
+
+
+namespace android {
+
+status_t SoundTriggerSessionCollection::acquireSession(audio_session_t session,
+ audio_io_handle_t ioHandle)
+{
+ add(session, ioHandle);
+
+ return NO_ERROR;
+}
+
+status_t SoundTriggerSessionCollection::releaseSession(audio_session_t session)
+{
+ ssize_t index = indexOfKey(session);
+ if (index < 0) {
+ ALOGW("acquireSoundTriggerSession() session %d not registered", session);
+ return BAD_VALUE;
+ }
+
+ removeItem(session);
+ return NO_ERROR;
+}
+
+}; //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/StreamDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/StreamDescriptor.cpp
new file mode 100644
index 0000000..b682e2c
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/StreamDescriptor.cpp
@@ -0,0 +1,162 @@
+/*
+ * Copyright (C) 2015 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 "APM::Volumes"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include "StreamDescriptor.h"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+namespace android {
+
+// --- StreamDescriptor class implementation
+
+StreamDescriptor::StreamDescriptor()
+ : mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
+{
+ mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
+}
+
+int StreamDescriptor::getVolumeIndex(audio_devices_t device) const
+{
+ device = Volume::getDeviceForVolume(device);
+ // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
+ if (mIndexCur.indexOfKey(device) < 0) {
+ device = AUDIO_DEVICE_OUT_DEFAULT;
+ }
+ return mIndexCur.valueFor(device);
+}
+
+void StreamDescriptor::clearCurrentVolumeIndex()
+{
+ mIndexCur.clear();
+}
+
+void StreamDescriptor::addCurrentVolumeIndex(audio_devices_t device, int index)
+{
+ mIndexCur.add(device, index);
+}
+
+void StreamDescriptor::setVolumeIndexMin(int volIndexMin)
+{
+ mIndexMin = volIndexMin;
+}
+
+void StreamDescriptor::setVolumeIndexMax(int volIndexMax)
+{
+ mIndexMax = volIndexMax;
+}
+
+void StreamDescriptor::setVolumeCurvePoint(Volume::device_category deviceCategory,
+ const VolumeCurvePoint *point)
+{
+ mVolumeCurve[deviceCategory] = point;
+}
+
+void StreamDescriptor::dump(int fd) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%s %02d %02d ",
+ mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
+ result.append(buffer);
+ for (size_t i = 0; i < mIndexCur.size(); i++) {
+ snprintf(buffer, SIZE, "%04x : %02d, ",
+ mIndexCur.keyAt(i),
+ mIndexCur.valueAt(i));
+ result.append(buffer);
+ }
+ result.append("\n");
+
+ write(fd, result.string(), result.size());
+}
+
+StreamDescriptorCollection::StreamDescriptorCollection()
+{
+ for (size_t stream = 0 ; stream < AUDIO_STREAM_CNT; stream++) {
+ add(static_cast<audio_stream_type_t>(stream), StreamDescriptor());
+ }
+}
+
+bool StreamDescriptorCollection::canBeMuted(audio_stream_type_t stream)
+{
+ return valueAt(stream).canBeMuted();
+}
+
+void StreamDescriptorCollection::clearCurrentVolumeIndex(audio_stream_type_t stream)
+{
+ editValueAt(stream).clearCurrentVolumeIndex();
+}
+
+void StreamDescriptorCollection::addCurrentVolumeIndex(audio_stream_type_t stream,
+ audio_devices_t device, int index)
+{
+ editValueAt(stream).addCurrentVolumeIndex(device, index);
+}
+
+void StreamDescriptorCollection::setVolumeCurvePoint(audio_stream_type_t stream,
+ Volume::device_category deviceCategory,
+ const VolumeCurvePoint *point)
+{
+ editValueAt(stream).setVolumeCurvePoint(deviceCategory, point);
+}
+
+const VolumeCurvePoint *StreamDescriptorCollection::getVolumeCurvePoint(audio_stream_type_t stream,
+ Volume::device_category deviceCategory) const
+{
+ return valueAt(stream).getVolumeCurvePoint(deviceCategory);
+}
+
+void StreamDescriptorCollection::setVolumeIndexMin(audio_stream_type_t stream,int volIndexMin)
+{
+ return editValueAt(stream).setVolumeIndexMin(volIndexMin);
+}
+
+void StreamDescriptorCollection::setVolumeIndexMax(audio_stream_type_t stream,int volIndexMax)
+{
+ return editValueAt(stream).setVolumeIndexMax(volIndexMax);
+}
+
+status_t StreamDescriptorCollection::dump(int fd) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "\nStreams dump:\n");
+ write(fd, buffer, strlen(buffer));
+ snprintf(buffer, SIZE,
+ " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < size(); i++) {
+ snprintf(buffer, SIZE, " %02zu ", i);
+ write(fd, buffer, strlen(buffer));
+ valueAt(i).dump(fd);
+ }
+
+ return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h b/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h
new file mode 100755
index 0000000..eadaa77
--- /dev/null
+++ b/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h
@@ -0,0 +1,171 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <AudioPolicyManagerObserver.h>
+#include <RoutingStrategy.h>
+#include <Volume.h>
+#include <HwModule.h>
+#include <DeviceDescriptor.h>
+#include <system/audio.h>
+#include <system/audio_policy.h>
+#include <utils/Errors.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+/**
+ * This interface is dedicated to the policy manager that a Policy Engine shall implement.
+ */
+class AudioPolicyManagerInterface
+{
+public:
+ /**
+ * Checks if the engine was correctly initialized.
+ *
+ * @return NO_ERROR if initialization has been done correctly, error code otherwise..
+ */
+ virtual status_t initCheck() = 0;
+
+ /**
+ * Sets the Manager observer that allows the engine to retrieve information on collection
+ * of devices, streams, HwModules, ...
+ *
+ * @param[in] observer handle on the manager.
+ */
+ virtual void setObserver(AudioPolicyManagerObserver *observer) = 0;
+
+ /**
+ * Get the input device selected for a given input source.
+ *
+ * @param[in] inputSource to get the selected input device associated to
+ *
+ * @return selected input device for the given input source, may be none if error.
+ */
+ virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const = 0;
+
+ /**
+ * Get the output device associated to a given strategy.
+ *
+ * @param[in] stream type for which the selected ouput device is requested.
+ *
+ * @return selected ouput device for the given strategy, may be none if error.
+ */
+ virtual audio_devices_t getDeviceForStrategy(routing_strategy stategy) const = 0;
+
+ /**
+ * Get the strategy selected for a given stream type.
+ *
+ * @param[in] stream: for which the selected strategy followed by is requested.
+ *
+ * @return strategy to be followed.
+ */
+ virtual routing_strategy getStrategyForStream(audio_stream_type_t stream) = 0;
+
+ /**
+ * Get the strategy selected for a given usage.
+ *
+ * @param[in] usage to get the selected strategy followed by.
+ *
+ * @return strategy to be followed.
+ */
+ virtual routing_strategy getStrategyForUsage(audio_usage_t usage) = 0;
+
+ /**
+ * Set the Telephony Mode.
+ *
+ * @param[in] mode: Android Phone state (normal, ringtone, csv, in communication)
+ *
+ * @return NO_ERROR if Telephony Mode set correctly, error code otherwise.
+ */
+ virtual status_t setPhoneState(audio_mode_t mode) = 0;
+
+ /**
+ * Get the telephony Mode
+ *
+ * @return the current telephony mode
+ */
+ virtual audio_mode_t getPhoneState() const = 0;
+
+ /**
+ * Set Force Use config for a given usage.
+ *
+ * @param[in] usage for which a configuration shall be forced.
+ * @param[in] config wished to be forced for the given usage.
+ *
+ * @return NO_ERROR if the Force Use config was set correctly, error code otherwise (e.g. config not
+ * allowed a given usage...)
+ */
+ virtual status_t setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config) = 0;
+
+ /**
+ * Get Force Use config for a given usage.
+ *
+ * @param[in] usage for which a configuration shall be forced.
+ *
+ * @return config wished to be forced for the given usage.
+ */
+ virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) const = 0;
+
+ /**
+ * Set the connection state of device(s).
+ *
+ * @param[in] devDesc for which the state has changed.
+ * @param[in] state of availability of this(these) device(s).
+ *
+ * @return NO_ERROR if devices criterion updated correctly, error code otherwise.
+ */
+ virtual status_t setDeviceConnectionState(const android::sp<android::DeviceDescriptor> devDesc,
+ audio_policy_dev_state_t state) = 0;
+
+ /**
+ * Translate a volume index given by the UI to an amplification value for a stream type
+ * and a device category.
+ *
+ * @param[in] deviceCategory for which the conversion is requested.
+ * @param[in] stream type for which the conversion is requested.
+ * @param[in] indexInUi index received from the UI to be translated.
+ *
+ * @return amplification value matching the UI index for this given device and stream.
+ */
+ virtual float volIndexToAmpl(Volume::device_category deviceCategory, audio_stream_type_t stream,
+ int indexInUi) = 0;
+
+ /**
+ * Initialize the min / max index of volume applicable for a given stream type. These indexes
+ * will be used upon conversion of UI index to volume amplification.
+ *
+ * @param[in] stream type for which the indexes need to be set
+ * @param[in] indexMin Minimum index allowed for this stream.
+ * @param[in] indexMax Maximum index allowed for this stream.
+ */
+ virtual status_t initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax) = 0;
+
+ /**
+ * Initialize volume curves for each strategy and device category
+ *
+ * @param[in] isSpeakerDrcEnabled true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER
+ path to boost soft sounds, used to adjust volume curves accordingly
+ */
+ virtual void initializeVolumeCurves(bool isSpeakerDrcEnabled) = 0;
+
+protected:
+ virtual ~AudioPolicyManagerInterface() {}
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
new file mode 100755
index 0000000..4f5427e
--- /dev/null
+++ b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
@@ -0,0 +1,62 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <AudioGain.h>
+#include <AudioPort.h>
+#include <AudioPatch.h>
+#include <IOProfile.h>
+#include <DeviceDescriptor.h>
+#include <AudioInputDescriptor.h>
+#include <AudioOutputDescriptor.h>
+#include <AudioPolicyMix.h>
+#include <SoundTriggerSession.h>
+#include <StreamDescriptor.h>
+
+namespace android {
+
+/**
+ * This interface is an observer that the manager shall implement to allows e.g. the engine
+ * to access to policy pillars elements (like output / input descritors collections,
+ * HwModule collections, AudioMix, ...
+ */
+class AudioPolicyManagerObserver
+{
+public:
+ virtual const AudioPatchCollection &getAudioPatches() const = 0;
+
+ virtual const SoundTriggerSessionCollection &getSoundTriggerSessionCollection() const = 0;
+
+ virtual const AudioPolicyMixCollection &getAudioPolicyMixCollection() const = 0;
+
+ virtual const AudioOutputCollection &getOutputs() const = 0;
+
+ virtual const AudioInputCollection &getInputs() const = 0;
+
+ virtual const DeviceVector &getAvailableOutputDevices() const = 0;
+
+ virtual const DeviceVector &getAvailableInputDevices() const = 0;
+
+ virtual StreamDescriptorCollection &getStreamDescriptors() = 0;
+
+ virtual const sp<DeviceDescriptor> &getDefaultOutputDevice() const = 0;
+
+protected:
+ virtual ~AudioPolicyManagerObserver() {}
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/enginedefault/Android.mk b/services/audiopolicy/enginedefault/Android.mk
new file mode 100755
index 0000000..b0ae835
--- /dev/null
+++ b/services/audiopolicy/enginedefault/Android.mk
@@ -0,0 +1,48 @@
+LOCAL_PATH := $(call my-dir)
+
+# Component build
+#######################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ src/Engine.cpp \
+ src/EngineInstance.cpp \
+ src/Gains.cpp \
+
+
+audio_policy_engine_includes_common := \
+ $(LOCAL_PATH)/include \
+ $(TOPDIR)frameworks/av/services/audiopolicy/engine/interface
+
+LOCAL_CFLAGS += \
+ -Wall \
+ -Werror \
+ -Wextra \
+
+LOCAL_EXPORT_C_INCLUDE_DIRS := \
+ $(audio_policy_engine_includes_common)
+
+LOCAL_C_INCLUDES := \
+ $(audio_policy_engine_includes_common) \
+ $(TARGET_OUT_HEADERS)/hw \
+ $(call include-path-for, frameworks-av) \
+ $(call include-path-for, audio-utils) \
+ $(call include-path-for, bionic) \
+ $(TOPDIR)frameworks/av/services/audiopolicy/common/include
+
+
+LOCAL_MODULE := libaudiopolicyenginedefault
+LOCAL_MODULE_TAGS := optional
+LOCAL_STATIC_LIBRARIES := \
+ libmedia_helper \
+ libaudiopolicycomponents
+
+LOCAL_SHARED_LIBRARIES += \
+ libcutils \
+ libutils \
+ libaudioutils \
+
+include external/stlport/libstlport.mk
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h b/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h
new file mode 100755
index 0000000..1e329f0
--- /dev/null
+++ b/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+class AudioPolicyManagerInterface;
+
+namespace android
+{
+namespace audio_policy
+{
+
+class Engine;
+
+class EngineInstance
+{
+protected:
+ EngineInstance();
+
+public:
+ virtual ~EngineInstance();
+
+ /**
+ * Get Audio Policy Engine instance.
+ *
+ * @return pointer to Route Manager Instance object.
+ */
+ static EngineInstance *getInstance();
+
+ /**
+ * Interface query.
+ * The first client of an interface of the policy engine will start the singleton.
+ *
+ * @tparam RequestedInterface: interface that the client is wishing to retrieve.
+ *
+ * @return interface handle.
+ */
+ template <class RequestedInterface>
+ RequestedInterface *queryInterface() const;
+
+protected:
+ /**
+ * Get Audio Policy Engine instance.
+ *
+ * @return Audio Policy Engine singleton.
+ */
+ Engine *getEngine() const;
+
+private:
+ /* Copy facilities are put private to disable copy. */
+ EngineInstance(const EngineInstance &object);
+ EngineInstance &operator=(const EngineInstance &object);
+};
+
+/**
+ * Limit template instantation to supported type interfaces.
+ * Compile time error will claim if invalid interface is requested.
+ */
+template <>
+AudioPolicyManagerInterface *EngineInstance::queryInterface() const;
+
+} // namespace audio_policy
+} // namespace android
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
new file mode 100755
index 0000000..1fd3341
--- /dev/null
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -0,0 +1,707 @@
+/*
+ * Copyright (C) 2015 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 "APM::AudioPolicyEngine"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include "Engine.h"
+#include "Gains.h"
+#include <AudioPolicyManagerObserver.h>
+#include <AudioPort.h>
+#include <IOProfile.h>
+#include <policy.h>
+#include <utils/String8.h>
+#include <utils/Log.h>
+
+namespace android
+{
+namespace audio_policy
+{
+
+Engine::Engine()
+ : mManagerInterface(this),
+ mPhoneState(AUDIO_MODE_NORMAL),
+ mApmObserver(NULL)
+{
+ for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) {
+ mForceUse[i] = AUDIO_POLICY_FORCE_NONE;
+ }
+}
+
+Engine::~Engine()
+{
+}
+
+void Engine::setObserver(AudioPolicyManagerObserver *observer)
+{
+ ALOG_ASSERT(observer != NULL, "Invalid Audio Policy Manager observer");
+ mApmObserver = observer;
+}
+
+status_t Engine::initCheck()
+{
+ return (mApmObserver != NULL) ? NO_ERROR : NO_INIT;
+}
+
+float Engine::volIndexToAmpl(Volume::device_category category, audio_stream_type_t streamType,
+ int indexInUi)
+{
+ const StreamDescriptor &streamDesc = mApmObserver->getStreamDescriptors().valueAt(streamType);
+ return Gains::volIndexToAmpl(category, streamDesc, indexInUi);
+}
+
+status_t Engine::initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax)
+{
+ ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+ if (indexMin < 0 || indexMin >= indexMax) {
+ ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d",
+ stream , indexMin, indexMax);
+ return BAD_VALUE;
+ }
+ mApmObserver->getStreamDescriptors().setVolumeIndexMin(stream, indexMin);
+ mApmObserver->getStreamDescriptors().setVolumeIndexMax(stream, indexMax);
+ return NO_ERROR;
+}
+
+void Engine::initializeVolumeCurves(bool isSpeakerDrcEnabled)
+{
+ StreamDescriptorCollection &streams = mApmObserver->getStreamDescriptors();
+
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ for (int j = 0; j < Volume::DEVICE_CATEGORY_CNT; j++) {
+ streams.setVolumeCurvePoint(static_cast<audio_stream_type_t>(i),
+ static_cast<Volume::device_category>(j),
+ Gains::sVolumeProfiles[i][j]);
+ }
+ }
+
+ // Check availability of DRC on speaker path: if available, override some of the speaker curves
+ if (isSpeakerDrcEnabled) {
+ streams.setVolumeCurvePoint(AUDIO_STREAM_SYSTEM, Volume::DEVICE_CATEGORY_SPEAKER,
+ Gains::sDefaultSystemVolumeCurveDrc);
+ streams.setVolumeCurvePoint(AUDIO_STREAM_RING, Volume::DEVICE_CATEGORY_SPEAKER,
+ Gains::sSpeakerSonificationVolumeCurveDrc);
+ streams.setVolumeCurvePoint(AUDIO_STREAM_ALARM, Volume::DEVICE_CATEGORY_SPEAKER,
+ Gains::sSpeakerSonificationVolumeCurveDrc);
+ streams.setVolumeCurvePoint(AUDIO_STREAM_NOTIFICATION, Volume::DEVICE_CATEGORY_SPEAKER,
+ Gains::sSpeakerSonificationVolumeCurveDrc);
+ streams.setVolumeCurvePoint(AUDIO_STREAM_MUSIC, Volume::DEVICE_CATEGORY_SPEAKER,
+ Gains::sSpeakerMediaVolumeCurveDrc);
+ streams.setVolumeCurvePoint(AUDIO_STREAM_ACCESSIBILITY, Volume::DEVICE_CATEGORY_SPEAKER,
+ Gains::sSpeakerMediaVolumeCurveDrc);
+ }
+}
+
+
+status_t Engine::setPhoneState(audio_mode_t state)
+{
+ ALOGV("setPhoneState() state %d", state);
+
+ if (state < 0 || state >= AUDIO_MODE_CNT) {
+ ALOGW("setPhoneState() invalid state %d", state);
+ return BAD_VALUE;
+ }
+
+ if (state == mPhoneState ) {
+ ALOGW("setPhoneState() setting same state %d", state);
+ return BAD_VALUE;
+ }
+
+ // store previous phone state for management of sonification strategy below
+ int oldState = mPhoneState;
+ mPhoneState = state;
+ StreamDescriptorCollection &streams = mApmObserver->getStreamDescriptors();
+ // are we entering or starting a call
+ if (!is_state_in_call(oldState) && is_state_in_call(state)) {
+ ALOGV(" Entering call in setPhoneState()");
+ for (int j = 0; j < Volume::DEVICE_CATEGORY_CNT; j++) {
+ streams.setVolumeCurvePoint(AUDIO_STREAM_DTMF, static_cast<Volume::device_category>(j),
+ Gains::sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j]);
+ }
+ } else if (is_state_in_call(oldState) && !is_state_in_call(state)) {
+ ALOGV(" Exiting call in setPhoneState()");
+ for (int j = 0; j < Volume::DEVICE_CATEGORY_CNT; j++) {
+ streams.setVolumeCurvePoint(AUDIO_STREAM_DTMF, static_cast<Volume::device_category>(j),
+ Gains::sVolumeProfiles[AUDIO_STREAM_DTMF][j]);
+ }
+ }
+ return NO_ERROR;
+}
+
+status_t Engine::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
+{
+ switch(usage) {
+ case AUDIO_POLICY_FORCE_FOR_COMMUNICATION:
+ if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO &&
+ config != AUDIO_POLICY_FORCE_NONE) {
+ ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
+ return BAD_VALUE;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_MEDIA:
+ if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP &&
+ config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
+ config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE &&
+ config != AUDIO_POLICY_FORCE_NO_BT_A2DP && config != AUDIO_POLICY_FORCE_SPEAKER ) {
+ ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
+ return BAD_VALUE;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_RECORD:
+ if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_NONE) {
+ ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
+ return BAD_VALUE;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_DOCK:
+ if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK &&
+ config != AUDIO_POLICY_FORCE_BT_DESK_DOCK &&
+ config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
+ config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) {
+ ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
+ }
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_SYSTEM:
+ if (config != AUDIO_POLICY_FORCE_NONE &&
+ config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+ ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
+ }
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO:
+ if (config != AUDIO_POLICY_FORCE_NONE &&
+ config != AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED) {
+ ALOGW("setForceUse() invalid config %d forHDMI_SYSTEM_AUDIO", config);
+ }
+ mForceUse[usage] = config;
+ break;
+ default:
+ ALOGW("setForceUse() invalid usage %d", usage);
+ break;
+ }
+ return NO_ERROR;
+}
+
+routing_strategy Engine::getStrategyForStream(audio_stream_type_t stream)
+{
+ // stream to strategy mapping
+ switch (stream) {
+ case AUDIO_STREAM_VOICE_CALL:
+ case AUDIO_STREAM_BLUETOOTH_SCO:
+ return STRATEGY_PHONE;
+ case AUDIO_STREAM_RING:
+ case AUDIO_STREAM_ALARM:
+ return STRATEGY_SONIFICATION;
+ case AUDIO_STREAM_NOTIFICATION:
+ return STRATEGY_SONIFICATION_RESPECTFUL;
+ case AUDIO_STREAM_DTMF:
+ return STRATEGY_DTMF;
+ default:
+ ALOGE("unknown stream type %d", stream);
+ case AUDIO_STREAM_SYSTEM:
+ // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+ // while key clicks are played produces a poor result
+ case AUDIO_STREAM_MUSIC:
+ return STRATEGY_MEDIA;
+ case AUDIO_STREAM_ENFORCED_AUDIBLE:
+ return STRATEGY_ENFORCED_AUDIBLE;
+ case AUDIO_STREAM_TTS:
+ return STRATEGY_TRANSMITTED_THROUGH_SPEAKER;
+ case AUDIO_STREAM_ACCESSIBILITY:
+ return STRATEGY_ACCESSIBILITY;
+ case AUDIO_STREAM_REROUTING:
+ return STRATEGY_REROUTING;
+ }
+}
+
+routing_strategy Engine::getStrategyForUsage(audio_usage_t usage)
+{
+ const AudioOutputCollection &outputs = mApmObserver->getOutputs();
+
+ // usage to strategy mapping
+ switch (usage) {
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ if (outputs.isStreamActive(AUDIO_STREAM_RING) ||
+ outputs.isStreamActive(AUDIO_STREAM_ALARM)) {
+ return STRATEGY_SONIFICATION;
+ }
+ if (isInCall()) {
+ return STRATEGY_PHONE;
+ }
+ return STRATEGY_ACCESSIBILITY;
+
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ return STRATEGY_MEDIA;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ return STRATEGY_PHONE;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ return STRATEGY_DTMF;
+
+ case AUDIO_USAGE_ALARM:
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ return STRATEGY_SONIFICATION;
+
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ return STRATEGY_SONIFICATION_RESPECTFUL;
+
+ case AUDIO_USAGE_UNKNOWN:
+ default:
+ return STRATEGY_MEDIA;
+ }
+}
+
+audio_devices_t Engine::getDeviceForStrategy(routing_strategy strategy) const
+{
+ const DeviceVector &availableOutputDevices = mApmObserver->getAvailableOutputDevices();
+ const DeviceVector &availableInputDevices = mApmObserver->getAvailableInputDevices();
+
+ const AudioOutputCollection &outputs = mApmObserver->getOutputs();
+
+ uint32_t device = AUDIO_DEVICE_NONE;
+ uint32_t availableOutputDevicesType = availableOutputDevices.types();
+
+ switch (strategy) {
+
+ case STRATEGY_TRANSMITTED_THROUGH_SPEAKER:
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+ if (!device) {
+ ALOGE("getDeviceForStrategy() no device found for "\
+ "STRATEGY_TRANSMITTED_THROUGH_SPEAKER");
+ }
+ break;
+
+ case STRATEGY_SONIFICATION_RESPECTFUL:
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION);
+ } else if (outputs.isStreamActiveRemotely(AUDIO_STREAM_MUSIC,
+ SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing on a remote device, use the the sonification behavior.
+ // Note that we test this usecase before testing if media is playing because
+ // the isStreamActive() method only informs about the activity of a stream, not
+ // if it's for local playback. Note also that we use the same delay between both tests
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION);
+ //user "safe" speaker if available instead of normal speaker to avoid triggering
+ //other acoustic safety mechanisms for notification
+ if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
+ device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ } else if (outputs.isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing (or has recently played), use the same device
+ device = getDeviceForStrategy(STRATEGY_MEDIA);
+ } else {
+ // when media is not playing anymore, fall back on the sonification behavior
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION);
+ //user "safe" speaker if available instead of normal speaker to avoid triggering
+ //other acoustic safety mechanisms for notification
+ if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE))
+ device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ }
+ break;
+
+ case STRATEGY_DTMF:
+ if (!isInCall()) {
+ // when off call, DTMF strategy follows the same rules as MEDIA strategy
+ device = getDeviceForStrategy(STRATEGY_MEDIA);
+ break;
+ }
+ // when in call, DTMF and PHONE strategies follow the same rules
+ // FALL THROUGH
+
+ case STRATEGY_PHONE:
+ // Force use of only devices on primary output if:
+ // - in call AND
+ // - cannot route from voice call RX OR
+ // - audio HAL version is < 3.0 and TX device is on the primary HW module
+ if (getPhoneState() == AUDIO_MODE_IN_CALL) {
+ audio_devices_t txDevice = getDeviceForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
+ sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput();
+ audio_devices_t availPrimaryInputDevices =
+ availableInputDevices.getDevicesFromHwModule(primaryOutput->getModuleHandle());
+ audio_devices_t availPrimaryOutputDevices =
+ primaryOutput->supportedDevices() & availableOutputDevices.types();
+
+ if (((availableInputDevices.types() &
+ AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) ||
+ (((txDevice & availPrimaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+ (primaryOutput->getAudioPort()->mModule->mHalVersion <
+ AUDIO_DEVICE_API_VERSION_3_0))) {
+ availableOutputDevicesType = availPrimaryOutputDevices;
+ }
+ }
+ // for phone strategy, we first consider the forced use and then the available devices by order
+ // of priority
+ switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
+ case AUDIO_POLICY_FORCE_BT_SCO:
+ if (!isInCall() || strategy != STRATEGY_DTMF) {
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+ if (device) break;
+ }
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
+ if (device) break;
+ // if SCO device is requested but no SCO device is available, fall back to default case
+ // FALL THROUGH
+
+ default: // FORCE_NONE
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
+ if (!isInCall() &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (outputs.getA2dpOutput() != 0)) {
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ if (device) break;
+ }
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ if (!isInCall()) {
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+ }
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_EARPIECE;
+ if (device) break;
+ device = mApmObserver->getDefaultOutputDevice()->type();
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
+ }
+ break;
+
+ case AUDIO_POLICY_FORCE_SPEAKER:
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
+ // A2DP speaker when forcing to speaker output
+ if (!isInCall() &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (outputs.getA2dpOutput() != 0)) {
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ if (device) break;
+ }
+ if (!isInCall()) {
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+ }
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_LINE;
+ if (device) break;
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device) break;
+ device = mApmObserver->getDefaultOutputDevice()->type();
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
+ }
+ break;
+ }
+ break;
+
+ case STRATEGY_SONIFICATION:
+
+ // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
+ // handleIncallSonification().
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_PHONE);
+ break;
+ }
+ // FALL THROUGH
+
+ case STRATEGY_ENFORCED_AUDIBLE:
+ // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
+ // except:
+ // - when in call where it doesn't default to STRATEGY_PHONE behavior
+ // - in countries where not enforced in which case it follows STRATEGY_MEDIA
+
+ if ((strategy == STRATEGY_SONIFICATION) ||
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
+ device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
+ }
+ }
+ // The second device used for sonification is the same as the device used by media strategy
+ // FALL THROUGH
+
+ // FIXME: STRATEGY_ACCESSIBILITY and STRATEGY_REROUTING follow STRATEGY_MEDIA for now
+ case STRATEGY_ACCESSIBILITY:
+ if (strategy == STRATEGY_ACCESSIBILITY) {
+ // do not route accessibility prompts to a digital output currently configured with a
+ // compressed format as they would likely not be mixed and dropped.
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = outputs.valueAt(i);
+ audio_devices_t devices = desc->device() &
+ (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC);
+ if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) &&
+ devices != AUDIO_DEVICE_NONE) {
+ availableOutputDevicesType = availableOutputDevices.types() & ~devices;
+ }
+ }
+ }
+ // FALL THROUGH
+
+ case STRATEGY_REROUTING:
+ case STRATEGY_MEDIA: {
+ uint32_t device2 = AUDIO_DEVICE_NONE;
+ if (strategy != STRATEGY_SONIFICATION) {
+ // no sonification on remote submix (e.g. WFD)
+ if (availableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0")) != 0) {
+ device2 = availableOutputDevices.types() & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ }
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (outputs.getA2dpOutput() != 0)) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ }
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] == AUDIO_POLICY_FORCE_SPEAKER)) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_LINE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+ // no sonification on aux digital (e.g. HDMI)
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+ }
+ int device3 = AUDIO_DEVICE_NONE;
+ if (strategy == STRATEGY_MEDIA) {
+ // ARC, SPDIF and AUX_LINE can co-exist with others.
+ device3 = availableOutputDevicesType & AUDIO_DEVICE_OUT_HDMI_ARC;
+ device3 |= (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPDIF);
+ device3 |= (availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_LINE);
+ }
+
+ device2 |= device3;
+ // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
+ // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
+ device |= device2;
+
+ // If hdmi system audio mode is on, remove speaker out of output list.
+ if ((strategy == STRATEGY_MEDIA) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] ==
+ AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) {
+ device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+ }
+
+ if (device) break;
+ device = mApmObserver->getDefaultOutputDevice()->type();
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
+ }
+ } break;
+
+ default:
+ ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+ break;
+ }
+
+ ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
+ return device;
+}
+
+
+audio_devices_t Engine::getDeviceForInputSource(audio_source_t inputSource) const
+{
+ const DeviceVector &availableOutputDevices = mApmObserver->getAvailableOutputDevices();
+ const DeviceVector &availableInputDevices = mApmObserver->getAvailableInputDevices();
+ const AudioOutputCollection &outputs = mApmObserver->getOutputs();
+ audio_devices_t availableDeviceTypes = availableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
+
+ uint32_t device = AUDIO_DEVICE_NONE;
+
+ switch (inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ break;
+ }
+ break;
+
+ case AUDIO_SOURCE_DEFAULT:
+ case AUDIO_SOURCE_MIC:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
+ } else if ((mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO) &&
+ (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
+ device = AUDIO_DEVICE_IN_USB_DEVICE;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ // Allow only use of devices on primary input if in call and HAL does not support routing
+ // to voice call path.
+ if ((getPhoneState() == AUDIO_MODE_IN_CALL) &&
+ (availableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) {
+ sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput();
+ availableDeviceTypes =
+ availableInputDevices.getDevicesFromHwModule(primaryOutput->getModuleHandle())
+ & ~AUDIO_DEVICE_BIT_IN;
+ }
+
+ switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
+ case AUDIO_POLICY_FORCE_BT_SCO:
+ // if SCO device is requested but no SCO device is available, fall back to default case
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ break;
+ }
+ // FALL THROUGH
+
+ default: // FORCE_NONE
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
+ device = AUDIO_DEVICE_IN_USB_DEVICE;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+
+ case AUDIO_POLICY_FORCE_SPEAKER:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
+ device = AUDIO_DEVICE_IN_BACK_MIC;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ }
+ break;
+
+ case AUDIO_SOURCE_VOICE_RECOGNITION:
+ case AUDIO_SOURCE_HOTWORD:
+ if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO &&
+ availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
+ device = AUDIO_DEVICE_IN_USB_DEVICE;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_CAMCORDER:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
+ device = AUDIO_DEVICE_IN_BACK_MIC;
+ } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ }
+ break;
+ case AUDIO_SOURCE_REMOTE_SUBMIX:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ }
+ break;
+ case AUDIO_SOURCE_FM_TUNER:
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) {
+ device = AUDIO_DEVICE_IN_FM_TUNER;
+ }
+ break;
+ default:
+ ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
+ break;
+ }
+ ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+ return device;
+}
+
+template <>
+AudioPolicyManagerInterface *Engine::queryInterface()
+{
+ return &mManagerInterface;
+}
+
+} // namespace audio_policy
+} // namespace android
+
+
diff --git a/services/audiopolicy/enginedefault/src/Engine.h b/services/audiopolicy/enginedefault/src/Engine.h
new file mode 100755
index 0000000..f44556c
--- /dev/null
+++ b/services/audiopolicy/enginedefault/src/Engine.h
@@ -0,0 +1,158 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+
+#include "AudioPolicyManagerInterface.h"
+#include "Gains.h"
+#include <AudioGain.h>
+#include <policy.h>
+
+namespace android
+{
+
+class AudioPolicyManagerObserver;
+
+namespace audio_policy
+{
+
+class Engine
+{
+public:
+ Engine();
+ virtual ~Engine();
+
+ template <class RequestedInterface>
+ RequestedInterface *queryInterface();
+
+private:
+ /// Interface members
+ class ManagerInterfaceImpl : public AudioPolicyManagerInterface
+ {
+ public:
+ ManagerInterfaceImpl(Engine *policyEngine)
+ : mPolicyEngine(policyEngine) {}
+
+ virtual void setObserver(AudioPolicyManagerObserver *observer)
+ {
+ mPolicyEngine->setObserver(observer);
+ }
+ virtual status_t initCheck()
+ {
+ return mPolicyEngine->initCheck();
+ }
+ virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const
+ {
+ return mPolicyEngine->getDeviceForInputSource(inputSource);
+ }
+ virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy) const
+ {
+ return mPolicyEngine->getDeviceForStrategy(strategy);
+ }
+ virtual routing_strategy getStrategyForStream(audio_stream_type_t stream)
+ {
+ return mPolicyEngine->getStrategyForStream(stream);
+ }
+ virtual routing_strategy getStrategyForUsage(audio_usage_t usage)
+ {
+ return mPolicyEngine->getStrategyForUsage(usage);
+ }
+ virtual status_t setPhoneState(audio_mode_t mode)
+ {
+ return mPolicyEngine->setPhoneState(mode);
+ }
+ virtual audio_mode_t getPhoneState() const
+ {
+ return mPolicyEngine->getPhoneState();
+ }
+ virtual status_t setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config)
+ {
+ return mPolicyEngine->setForceUse(usage, config);
+ }
+ virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) const
+ {
+ return mPolicyEngine->getForceUse(usage);
+ }
+ virtual status_t setDeviceConnectionState(const sp<DeviceDescriptor> /*devDesc*/,
+ audio_policy_dev_state_t /*state*/)
+ {
+ return NO_ERROR;
+ }
+ virtual status_t initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax)
+ {
+ return mPolicyEngine->initStreamVolume(stream, indexMin, indexMax);
+ }
+ virtual void initializeVolumeCurves(bool isSpeakerDrcEnabled)
+ {
+ return mPolicyEngine->initializeVolumeCurves(isSpeakerDrcEnabled);
+ }
+ virtual float volIndexToAmpl(Volume::device_category deviceCategory,
+ audio_stream_type_t stream,int indexInUi)
+ {
+ return mPolicyEngine->volIndexToAmpl(deviceCategory, stream, indexInUi);
+ }
+ private:
+ Engine *mPolicyEngine;
+ } mManagerInterface;
+
+private:
+ /* Copy facilities are put private to disable copy. */
+ Engine(const Engine &object);
+ Engine &operator=(const Engine &object);
+
+ void setObserver(AudioPolicyManagerObserver *observer);
+
+ status_t initCheck();
+
+ inline bool isInCall() const
+ {
+ return is_state_in_call(mPhoneState);
+ }
+
+ status_t setPhoneState(audio_mode_t mode);
+ audio_mode_t getPhoneState() const
+ {
+ return mPhoneState;
+ }
+ status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
+ audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) const
+ {
+ return mForceUse[usage];
+ }
+ status_t setDefaultDevice(audio_devices_t device);
+
+ routing_strategy getStrategyForStream(audio_stream_type_t stream);
+ routing_strategy getStrategyForUsage(audio_usage_t usage);
+ audio_devices_t getDeviceForStrategy(routing_strategy strategy) const;
+ audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const;
+
+ float volIndexToAmpl(Volume::device_category category,
+ audio_stream_type_t stream, int indexInUi);
+ status_t initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax);
+ void initializeVolumeCurves(bool isSpeakerDrcEnabled);
+
+ audio_mode_t mPhoneState; /**< current phone state. */
+
+ /** current forced use configuration. */
+ audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT];
+
+ AudioPolicyManagerObserver *mApmObserver;
+};
+} // namespace audio_policy
+} // namespace android
+
diff --git a/services/audiopolicy/enginedefault/src/EngineInstance.cpp b/services/audiopolicy/enginedefault/src/EngineInstance.cpp
new file mode 100755
index 0000000..17e9832
--- /dev/null
+++ b/services/audiopolicy/enginedefault/src/EngineInstance.cpp
@@ -0,0 +1,54 @@
+/*
+ * Copyright (C) 2015 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 <AudioPolicyManagerInterface.h>
+#include "AudioPolicyEngineInstance.h"
+#include "Engine.h"
+
+namespace android
+{
+namespace audio_policy
+{
+
+EngineInstance::EngineInstance()
+{
+}
+
+EngineInstance *EngineInstance::getInstance()
+{
+ static EngineInstance instance;
+ return &instance;
+}
+
+EngineInstance::~EngineInstance()
+{
+}
+
+Engine *EngineInstance::getEngine() const
+{
+ static Engine engine;
+ return &engine;
+}
+
+template <>
+AudioPolicyManagerInterface *EngineInstance::queryInterface() const
+{
+ return getEngine()->queryInterface<AudioPolicyManagerInterface>();
+}
+
+} // namespace audio_policy
+} // namespace android
+
diff --git a/services/audiopolicy/enginedefault/src/Gains.cpp b/services/audiopolicy/enginedefault/src/Gains.cpp
new file mode 100644
index 0000000..a684fdd
--- /dev/null
+++ b/services/audiopolicy/enginedefault/src/Gains.cpp
@@ -0,0 +1,247 @@
+/*
+ * Copyright (C) 2015 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 "APM::Gains"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include "Gains.h"
+#include <Volume.h>
+#include <math.h>
+#include <utils/String8.h>
+
+namespace android {
+
+// Enginedefault
+const VolumeCurvePoint
+Gains::sDefaultVolumeCurve[Volume::VOLCNT] = {
+ {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
+};
+
+
+const VolumeCurvePoint
+Gains::sDefaultMediaVolumeCurve[Volume::VOLCNT] = {
+ {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+Gains::sExtMediaSystemVolumeCurve[Volume::VOLCNT] = {
+ {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
+};
+
+const VolumeCurvePoint
+Gains::sSpeakerMediaVolumeCurve[Volume::VOLCNT] = {
+ {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+Gains::sSpeakerMediaVolumeCurveDrc[Volume::VOLCNT] = {
+ {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+Gains::sSpeakerSonificationVolumeCurve[Volume::VOLCNT] = {
+ {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+Gains::sSpeakerSonificationVolumeCurveDrc[Volume::VOLCNT] = {
+ {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
+};
+
+// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
+// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
+// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
+// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
+
+const VolumeCurvePoint
+Gains::sDefaultSystemVolumeCurve[Volume::VOLCNT] = {
+ {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+Gains::sDefaultSystemVolumeCurveDrc[Volume::VOLCNT] = {
+ {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+Gains::sHeadsetSystemVolumeCurve[Volume::VOLCNT] = {
+ {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
+};
+
+const VolumeCurvePoint
+Gains::sDefaultVoiceVolumeCurve[Volume::VOLCNT] = {
+ {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+Gains::sSpeakerVoiceVolumeCurve[Volume::VOLCNT] = {
+ {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+Gains::sLinearVolumeCurve[Volume::VOLCNT] = {
+ {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+Gains::sSilentVolumeCurve[Volume::VOLCNT] = {
+ {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f}
+};
+
+const VolumeCurvePoint
+Gains::sFullScaleVolumeCurve[Volume::VOLCNT] = {
+ {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint *Gains::sVolumeProfiles[AUDIO_STREAM_CNT]
+ [Volume::DEVICE_CATEGORY_CNT] = {
+ { // AUDIO_STREAM_VOICE_CALL
+ Gains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_SYSTEM
+ Gains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_RING
+ Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_MUSIC
+ Gains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ALARM
+ Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_NOTIFICATION
+ Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_BLUETOOTH_SCO
+ Gains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ENFORCED_AUDIBLE
+ Gains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_DTMF
+ Gains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_TTS
+ // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER
+ Gains::sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ACCESSIBILITY
+ Gains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_REROUTING
+ Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_PATCH
+ Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ Gains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+};
+
+//static
+float Gains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+ int indexInUi)
+{
+ Volume::device_category deviceCategory = Volume::getDeviceCategory(device);
+ const VolumeCurvePoint *curve = streamDesc.getVolumeCurvePoint(deviceCategory);
+
+ // the volume index in the UI is relative to the min and max volume indices for this stream type
+ int nbSteps = 1 + curve[Volume::VOLMAX].mIndex -
+ curve[Volume::VOLMIN].mIndex;
+ int volIdx = (nbSteps * (indexInUi - streamDesc.getVolumeIndexMin())) /
+ (streamDesc.getVolumeIndexMax() - streamDesc.getVolumeIndexMin());
+
+ // find what part of the curve this index volume belongs to, or if it's out of bounds
+ int segment = 0;
+ if (volIdx < curve[Volume::VOLMIN].mIndex) { // out of bounds
+ return 0.0f;
+ } else if (volIdx < curve[Volume::VOLKNEE1].mIndex) {
+ segment = 0;
+ } else if (volIdx < curve[Volume::VOLKNEE2].mIndex) {
+ segment = 1;
+ } else if (volIdx <= curve[Volume::VOLMAX].mIndex) {
+ segment = 2;
+ } else { // out of bounds
+ return 1.0f;
+ }
+
+ // linear interpolation in the attenuation table in dB
+ float decibels = curve[segment].mDBAttenuation +
+ ((float)(volIdx - curve[segment].mIndex)) *
+ ( (curve[segment+1].mDBAttenuation -
+ curve[segment].mDBAttenuation) /
+ ((float)(curve[segment+1].mIndex -
+ curve[segment].mIndex)) );
+
+ float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
+
+ ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
+ curve[segment].mIndex, volIdx,
+ curve[segment+1].mIndex,
+ curve[segment].mDBAttenuation,
+ decibels,
+ curve[segment+1].mDBAttenuation,
+ amplification);
+
+ return amplification;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/enginedefault/src/Gains.h b/services/audiopolicy/enginedefault/src/Gains.h
new file mode 100644
index 0000000..b5601ca
--- /dev/null
+++ b/services/audiopolicy/enginedefault/src/Gains.h
@@ -0,0 +1,59 @@
+/*
+ * Copyright (C) 2015 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.
+ */
+
+#pragma once
+
+#include <StreamDescriptor.h>
+#include <utils/KeyedVector.h>
+#include <system/audio.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+class StreamDescriptor;
+
+class Gains
+{
+public :
+ static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+ int indexInUi);
+
+ // default volume curve
+ static const VolumeCurvePoint sDefaultVolumeCurve[Volume::VOLCNT];
+ // default volume curve for media strategy
+ static const VolumeCurvePoint sDefaultMediaVolumeCurve[Volume::VOLCNT];
+ // volume curve for non-media audio on ext media outputs (HDMI, Line, etc)
+ static const VolumeCurvePoint sExtMediaSystemVolumeCurve[Volume::VOLCNT];
+ // volume curve for media strategy on speakers
+ static const VolumeCurvePoint sSpeakerMediaVolumeCurve[Volume::VOLCNT];
+ static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[Volume::VOLCNT];
+ // volume curve for sonification strategy on speakers
+ static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[Volume::VOLCNT];
+ static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[Volume::VOLCNT];
+ static const VolumeCurvePoint sDefaultSystemVolumeCurve[Volume::VOLCNT];
+ static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[Volume::VOLCNT];
+ static const VolumeCurvePoint sHeadsetSystemVolumeCurve[Volume::VOLCNT];
+ static const VolumeCurvePoint sDefaultVoiceVolumeCurve[Volume::VOLCNT];
+ static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[Volume::VOLCNT];
+ static const VolumeCurvePoint sLinearVolumeCurve[Volume::VOLCNT];
+ static const VolumeCurvePoint sSilentVolumeCurve[Volume::VOLCNT];
+ static const VolumeCurvePoint sFullScaleVolumeCurve[Volume::VOLCNT];
+ // default volume curves per stream and device category. See initializeVolumeCurves()
+ static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][Volume::DEVICE_CATEGORY_CNT];
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audiopolicy/manager/AudioPolicyFactory.cpp
similarity index 94%
rename from services/audiopolicy/AudioPolicyFactory.cpp
rename to services/audiopolicy/manager/AudioPolicyFactory.cpp
index 2ae7bc1..9910a1f 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/services/audiopolicy/manager/AudioPolicyFactory.cpp
@@ -14,7 +14,7 @@
* limitations under the License.
*/
-#include "AudioPolicyManager.h"
+#include "managerdefault/AudioPolicyManager.h"
namespace android {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
new file mode 100644
index 0000000..797a2b4
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -0,0 +1,4561 @@
+/*
+ * 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 "APM::AudioPolicyManager"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include <inttypes.h>
+#include <math.h>
+
+#include <AudioPolicyManagerInterface.h>
+#include <AudioPolicyEngineInstance.h>
+#include <cutils/properties.h>
+#include <utils/Log.h>
+#include <hardware/audio.h>
+#include <hardware/audio_effect.h>
+#include <media/AudioParameter.h>
+#include <media/AudioPolicyHelper.h>
+#include <soundtrigger/SoundTrigger.h>
+#include "AudioPolicyManager.h"
+#include "audio_policy_conf.h"
+#include <ConfigParsingUtils.h>
+#include <policy.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+
+status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name)
+{
+ return setDeviceConnectionStateInt(device, state, device_address, device_name);
+}
+
+status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name)
+{
+ ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
+- device, state, device_address, device_name);
+
+ // connect/disconnect only 1 device at a time
+ if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
+
+ sp<DeviceDescriptor> devDesc =
+ mHwModules.getDeviceDescriptor(device, device_address, device_name);
+
+ // handle output devices
+ if (audio_is_output_device(device)) {
+ SortedVector <audio_io_handle_t> outputs;
+
+ ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
+
+ // save a copy of the opened output descriptors before any output is opened or closed
+ // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
+ mPreviousOutputs = mOutputs;
+ switch (state)
+ {
+ // handle output device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+ if (index >= 0) {
+ ALOGW("setDeviceConnectionState() device already connected: %x", device);
+ return INVALID_OPERATION;
+ }
+ ALOGV("setDeviceConnectionState() connecting device %x", device);
+
+ // register new device as available
+ index = mAvailableOutputDevices.add(devDesc);
+ if (index >= 0) {
+ sp<HwModule> module = mHwModules.getModuleForDevice(device);
+ if (module == 0) {
+ ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
+ device);
+ mAvailableOutputDevices.remove(devDesc);
+ return INVALID_OPERATION;
+ }
+ mAvailableOutputDevices[index]->attach(module);
+ } else {
+ return NO_MEMORY;
+ }
+
+ if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
+ mAvailableOutputDevices.remove(devDesc);
+ return INVALID_OPERATION;
+ }
+ // Propagate device availability to Engine
+ mEngine->setDeviceConnectionState(devDesc, state);
+
+ // outputs should never be empty here
+ ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
+ "checkOutputsForDevice() returned no outputs but status OK");
+ ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
+ outputs.size());
+
+ // Send connect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ } break;
+ // handle output device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (index < 0) {
+ ALOGW("setDeviceConnectionState() device not connected: %x", device);
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
+
+ // Send Disconnect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ // remove device from available output devices
+ mAvailableOutputDevices.remove(devDesc);
+
+ checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
+
+ // Propagate device availability to Engine
+ mEngine->setDeviceConnectionState(devDesc, state);
+ } break;
+
+ default:
+ ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
+ // output is suspended before any tracks are moved to it
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ // outputs must be closed after checkOutputForAllStrategies() is executed
+ if (!outputs.isEmpty()) {
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+ // close unused outputs after device disconnection or direct outputs that have been
+ // opened by checkOutputsForDevice() to query dynamic parameters
+ if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
+ (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+ (desc->mDirectOpenCount == 0))) {
+ closeOutput(outputs[i]);
+ }
+ }
+ // check again after closing A2DP output to reset mA2dpSuspended if needed
+ checkA2dpSuspend();
+ }
+
+ updateDevicesAndOutputs();
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t output = mOutputs.keyAt(i);
+ if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
+ audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i),
+ true /*fromCache*/);
+ // do not force device change on duplicated output because if device is 0, it will
+ // also force a device 0 for the two outputs it is duplicated to which may override
+ // a valid device selection on those outputs.
+ bool force = !mOutputs.valueAt(i)->isDuplicated()
+ && (!device_distinguishes_on_address(device)
+ // always force when disconnecting (a non-duplicated device)
+ || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
+ setOutputDevice(output, newDevice, force, 0);
+ }
+ }
+
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+ } // end if is output device
+
+ // handle input devices
+ if (audio_is_input_device(device)) {
+ SortedVector <audio_io_handle_t> inputs;
+
+ ssize_t index = mAvailableInputDevices.indexOf(devDesc);
+ switch (state)
+ {
+ // handle input device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+ if (index >= 0) {
+ ALOGW("setDeviceConnectionState() device already connected: %d", device);
+ return INVALID_OPERATION;
+ }
+ sp<HwModule> module = mHwModules.getModuleForDevice(device);
+ if (module == NULL) {
+ ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
+ device);
+ return INVALID_OPERATION;
+ }
+ if (checkInputsForDevice(device, state, inputs, devDesc->mAddress) != NO_ERROR) {
+ return INVALID_OPERATION;
+ }
+
+ index = mAvailableInputDevices.add(devDesc);
+ if (index >= 0) {
+ mAvailableInputDevices[index]->attach(module);
+ } else {
+ return NO_MEMORY;
+ }
+
+ // Set connect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ // Propagate device availability to Engine
+ mEngine->setDeviceConnectionState(devDesc, state);
+ } break;
+
+ // handle input device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (index < 0) {
+ ALOGW("setDeviceConnectionState() device not connected: %d", device);
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
+
+ // Set Disconnect to HALs
+ AudioParameter param = AudioParameter(devDesc->mAddress);
+ param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+
+ checkInputsForDevice(device, state, inputs, devDesc->mAddress);
+ mAvailableInputDevices.remove(devDesc);
+
+ // Propagate device availability to Engine
+ mEngine->setDeviceConnectionState(devDesc, state);
+ } break;
+
+ default:
+ ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ closeAllInputs();
+
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+ } // end if is input device
+
+ ALOGW("setDeviceConnectionState() invalid device: %x", device);
+ return BAD_VALUE;
+}
+
+audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
+ const char *device_address)
+{
+ sp<DeviceDescriptor> devDesc = mHwModules.getDeviceDescriptor(device, device_address, "");
+
+ DeviceVector *deviceVector;
+
+ if (audio_is_output_device(device)) {
+ deviceVector = &mAvailableOutputDevices;
+ } else if (audio_is_input_device(device)) {
+ deviceVector = &mAvailableInputDevices;
+ } else {
+ ALOGW("getDeviceConnectionState() invalid device type %08x", device);
+ return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+ }
+ return deviceVector->getDeviceConnectionState(devDesc);
+}
+
+void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs)
+{
+ bool createTxPatch = false;
+ struct audio_patch patch;
+ patch.num_sources = 1;
+ patch.num_sinks = 1;
+ status_t status;
+ audio_patch_handle_t afPatchHandle;
+ DeviceVector deviceList;
+
+ audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
+ ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice);
+
+ // release existing RX patch if any
+ if (mCallRxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+ mCallRxPatch.clear();
+ }
+ // release TX patch if any
+ if (mCallTxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+ mCallTxPatch.clear();
+ }
+
+ // If the RX device is on the primary HW module, then use legacy routing method for voice calls
+ // via setOutputDevice() on primary output.
+ // Otherwise, create two audio patches for TX and RX path.
+ if (availablePrimaryOutputDevices() & rxDevice) {
+ setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs);
+ // If the TX device is also on the primary HW module, setOutputDevice() will take care
+ // of it due to legacy implementation. If not, create a patch.
+ if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN)
+ == AUDIO_DEVICE_NONE) {
+ createTxPatch = true;
+ }
+ } else {
+ // create RX path audio patch
+ deviceList = mAvailableOutputDevices.getDevicesFromType(rxDevice);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() selected device not in output device list");
+ sp<DeviceDescriptor> rxSinkDeviceDesc = deviceList.itemAt(0);
+ deviceList = mAvailableInputDevices.getDevicesFromType(AUDIO_DEVICE_IN_TELEPHONY_RX);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() no telephony RX device");
+ sp<DeviceDescriptor> rxSourceDeviceDesc = deviceList.itemAt(0);
+
+ rxSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
+ rxSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
+
+ // request to reuse existing output stream if one is already opened to reach the RX device
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevice(rxDevice, mOutputs);
+ audio_io_handle_t output = selectOutput(outputs,
+ AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_INVALID);
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ ALOG_ASSERT(!outputDesc->isDuplicated(),
+ "updateCallRouting() RX device output is duplicated");
+ outputDesc->toAudioPortConfig(&patch.sources[1]);
+ patch.num_sources = 2;
+ }
+
+ afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0);
+ ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch",
+ status);
+ if (status == NO_ERROR) {
+ mCallRxPatch = new AudioPatch(&patch, mUidCached);
+ mCallRxPatch->mAfPatchHandle = afPatchHandle;
+ mCallRxPatch->mUid = mUidCached;
+ }
+ createTxPatch = true;
+ }
+ if (createTxPatch) {
+
+ struct audio_patch patch;
+ patch.num_sources = 1;
+ patch.num_sinks = 1;
+ deviceList = mAvailableInputDevices.getDevicesFromType(txDevice);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() selected device not in input device list");
+ sp<DeviceDescriptor> txSourceDeviceDesc = deviceList.itemAt(0);
+ txSourceDeviceDesc->toAudioPortConfig(&patch.sources[0]);
+ deviceList = mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX);
+ ALOG_ASSERT(!deviceList.isEmpty(),
+ "updateCallRouting() no telephony TX device");
+ sp<DeviceDescriptor> txSinkDeviceDesc = deviceList.itemAt(0);
+ txSinkDeviceDesc->toAudioPortConfig(&patch.sinks[0]);
+
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX, mOutputs);
+ audio_io_handle_t output = selectOutput(outputs,
+ AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_INVALID);
+ // request to reuse existing output stream if one is already opened to reach the TX
+ // path output device
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ ALOG_ASSERT(!outputDesc->isDuplicated(),
+ "updateCallRouting() RX device output is duplicated");
+ outputDesc->toAudioPortConfig(&patch.sources[1]);
+ patch.num_sources = 2;
+ }
+
+ afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, 0);
+ ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch",
+ status);
+ if (status == NO_ERROR) {
+ mCallTxPatch = new AudioPatch(&patch, mUidCached);
+ mCallTxPatch->mAfPatchHandle = afPatchHandle;
+ mCallTxPatch->mUid = mUidCached;
+ }
+ }
+}
+
+void AudioPolicyManager::setPhoneState(audio_mode_t state)
+{
+ ALOGV("setPhoneState() state %d", state);
+ // store previous phone state for management of sonification strategy below
+ int oldState = mEngine->getPhoneState();
+
+ if (mEngine->setPhoneState(state) != NO_ERROR) {
+ ALOGW("setPhoneState() invalid or same state %d", state);
+ return;
+ }
+ /// Opens: can these line be executed after the switch of volume curves???
+ // if leaving call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isInCall()) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ handleIncallSonification((audio_stream_type_t)stream, false, true);
+ }
+
+ // force reevaluating accessibility routing when call starts
+ mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+ }
+
+ /**
+ * Switching to or from incall state or switching between telephony and VoIP lead to force
+ * routing command.
+ */
+ bool force = ((is_state_in_call(oldState) != is_state_in_call(state))
+ || (is_state_in_call(state) && (state != oldState)));
+
+ // check for device and output changes triggered by new phone state
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+
+ sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+
+ int delayMs = 0;
+ if (isStateInCall(state)) {
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ // mute media and sonification strategies and delay device switch by the largest
+ // latency of any output where either strategy is active.
+ // This avoid sending the ring tone or music tail into the earpiece or headset.
+ if ((isStrategyActive(desc, STRATEGY_MEDIA,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime) ||
+ isStrategyActive(desc, STRATEGY_SONIFICATION,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime)) &&
+ (delayMs < (int)desc->mLatency*2)) {
+ delayMs = desc->mLatency*2;
+ }
+ setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+ setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+ }
+ }
+
+ // Note that despite the fact that getNewOutputDevice() is called on the primary output,
+ // the device returned is not necessarily reachable via this output
+ audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ // force routing command to audio hardware when ending call
+ // even if no device change is needed
+ if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
+ rxDevice = hwOutputDesc->device();
+ }
+
+ if (state == AUDIO_MODE_IN_CALL) {
+ updateCallRouting(rxDevice, delayMs);
+ } else if (oldState == AUDIO_MODE_IN_CALL) {
+ if (mCallRxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+ mCallRxPatch.clear();
+ }
+ if (mCallTxPatch != 0) {
+ mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+ mCallTxPatch.clear();
+ }
+ setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+ } else {
+ setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+ }
+ // if entering in call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isStateInCall(state)) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ handleIncallSonification((audio_stream_type_t)stream, true, true);
+ }
+ }
+
+ // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+ if (state == AUDIO_MODE_RINGTONE &&
+ isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+ mLimitRingtoneVolume = true;
+ } else {
+ mLimitRingtoneVolume = false;
+ }
+}
+
+audio_mode_t AudioPolicyManager::getPhoneState() {
+ return mEngine->getPhoneState();
+}
+
+void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config)
+{
+ ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
+
+ if (mEngine->setForceUse(usage, config) != NO_ERROR) {
+ ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
+ return;
+ }
+ bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
+ (usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
+ (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
+
+ // check for device and output changes triggered by new force usage
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) {
+ audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
+ updateCallRouting(newDevice);
+ }
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t output = mOutputs.keyAt(i);
+ audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/);
+ if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) {
+ setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+ }
+ if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+ applyStreamVolumes(output, newDevice, 0, true);
+ }
+ }
+
+ audio_io_handle_t activeInput = mInputs.getActiveInput();
+ if (activeInput != 0) {
+ setInputDevice(activeInput, getNewInputDevice(activeInput));
+ }
+
+}
+
+void AudioPolicyManager::setSystemProperty(const char* property, const char* value)
+{
+ ALOGV("setSystemProperty() property %s, value %s", property, value);
+}
+
+// Find a direct output profile compatible with the parameters passed, even if the input flags do
+// not explicitly request a direct output
+sp<IOProfile> AudioPolicyManager::getProfileForDirectOutput(
+ audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags)
+{
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
+ sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ bool found = profile->isCompatibleProfile(device, String8(""), samplingRate,
+ NULL /*updatedSamplingRate*/, format, channelMask,
+ flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ?
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT);
+ if (found && (mAvailableOutputDevices.types() & profile->mSupportedDevices.types())) {
+ return profile;
+ }
+ }
+ }
+ return 0;
+}
+
+audio_io_handle_t AudioPolicyManager::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)
+{
+ routing_strategy strategy = getStrategy(stream);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
+ device, stream, samplingRate, format, channelMask, flags);
+
+ return getOutputForDevice(device, AUDIO_SESSION_ALLOCATE,
+ stream, samplingRate,format, channelMask,
+ flags, offloadInfo);
+}
+
+status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ 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_attributes_t attributes;
+ if (attr != NULL) {
+ if (!isValidAttributes(attr)) {
+ ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
+ attr->usage, attr->content_type, attr->flags,
+ attr->tags);
+ return BAD_VALUE;
+ }
+ attributes = *attr;
+ } else {
+ if (*stream < AUDIO_STREAM_MIN || *stream >= AUDIO_STREAM_PUBLIC_CNT) {
+ ALOGE("getOutputForAttr(): invalid stream type");
+ return BAD_VALUE;
+ }
+ stream_type_to_audio_attributes(*stream, &attributes);
+ }
+ sp<AudioOutputDescriptor> desc;
+ if (mPolicyMixes.getOutputForAttr(attributes, desc) == NO_ERROR) {
+ ALOG_ASSERT(desc != 0, "Invalid desc returned by getOutputForAttr");
+ if (!audio_is_linear_pcm(format)) {
+ return BAD_VALUE;
+ }
+ *stream = streamTypefromAttributesInt(&attributes);
+ *output = desc->mIoHandle;
+ ALOGV("getOutputForAttr() returns output %d", *output);
+ return NO_ERROR;
+ }
+ if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
+ ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
+ return BAD_VALUE;
+ }
+
+ ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x",
+ attributes.usage, attributes.content_type, attributes.tags, attributes.flags);
+
+ routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+
+ if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
+ }
+
+ ALOGV("getOutputForAttr() device 0x%x, samplingRate %d, format %x, channelMask %x, flags %x",
+ device, samplingRate, format, channelMask, flags);
+
+ *stream = streamTypefromAttributesInt(&attributes);
+ *output = getOutputForDevice(device, session, *stream,
+ samplingRate, format, channelMask,
+ flags, offloadInfo);
+ if (*output == AUDIO_IO_HANDLE_NONE) {
+ return INVALID_OPERATION;
+ }
+ return NO_ERROR;
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForDevice(
+ audio_devices_t device,
+ audio_session_t session __unused,
+ 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 output = AUDIO_IO_HANDLE_NONE;
+ uint32_t latency = 0;
+ status_t status;
+
+#ifdef AUDIO_POLICY_TEST
+ if (mCurOutput != 0) {
+ ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+ mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+ if (mTestOutputs[mCurOutput] == 0) {
+ ALOGV("getOutput() opening test output");
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
+ outputDesc->mDevice = mTestDevice;
+ outputDesc->mLatency = mTestLatencyMs;
+ outputDesc->mFlags =
+ (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
+ outputDesc->mRefCount[stream] = 0;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = mTestSamplingRate;
+ config.channel_mask = mTestChannels;
+ config.format = mTestFormat;
+ if (offloadInfo != NULL) {
+ config.offload_info = *offloadInfo;
+ }
+ status = mpClientInterface->openOutput(0,
+ &mTestOutputs[mCurOutput],
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (status == NO_ERROR) {
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mFormat = config.format;
+ outputDesc->mChannelMask = config.channel_mask;
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"),mCurOutput);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+ addOutput(mTestOutputs[mCurOutput], outputDesc);
+ }
+ }
+ return mTestOutputs[mCurOutput];
+ }
+#endif //AUDIO_POLICY_TEST
+
+ // open a direct output if required by specified parameters
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+ if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+ // only allow deep buffering for music stream type
+ if (stream != AUDIO_STREAM_MUSIC) {
+ flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+ }
+
+ sp<IOProfile> profile;
+
+ // skip direct output selection if the request can obviously be attached to a mixed output
+ // and not explicitly requested
+ if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
+ audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE &&
+ audio_channel_count_from_out_mask(channelMask) <= 2) {
+ goto non_direct_output;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+
+ if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
+ !mEffects.isNonOffloadableEffectEnabled()) {
+ profile = getProfileForDirectOutput(device,
+ samplingRate,
+ format,
+ channelMask,
+ (audio_output_flags_t)flags);
+ }
+
+ if (profile != 0) {
+ sp<AudioOutputDescriptor> outputDesc = NULL;
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+ outputDesc = desc;
+ // reuse direct output if currently open and configured with same parameters
+ if ((samplingRate == outputDesc->mSamplingRate) &&
+ (format == outputDesc->mFormat) &&
+ (channelMask == outputDesc->mChannelMask)) {
+ outputDesc->mDirectOpenCount++;
+ ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
+ return mOutputs.keyAt(i);
+ }
+ }
+ }
+ // close direct output if currently open and configured with different parameters
+ if (outputDesc != NULL) {
+ closeOutput(outputDesc->mIoHandle);
+ }
+ outputDesc = new AudioOutputDescriptor(profile);
+ outputDesc->mDevice = device;
+ outputDesc->mLatency = 0;
+ outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = samplingRate;
+ config.channel_mask = channelMask;
+ config.format = format;
+ if (offloadInfo != NULL) {
+ config.offload_info = *offloadInfo;
+ }
+ status = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &output,
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+
+ // only accept an output with the requested parameters
+ if (status != NO_ERROR ||
+ (samplingRate != 0 && samplingRate != config.sample_rate) ||
+ (format != AUDIO_FORMAT_DEFAULT && format != config.format) ||
+ (channelMask != 0 && channelMask != config.channel_mask)) {
+ ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+ "format %d %d, channelMask %04x %04x", output, samplingRate,
+ outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+ outputDesc->mChannelMask);
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ mpClientInterface->closeOutput(output);
+ }
+ // fall back to mixer output if possible when the direct output could not be open
+ if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
+ goto non_direct_output;
+ }
+ // fall back to mixer output if possible when the direct output could not be open
+ if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
+ goto non_direct_output;
+ }
+ return AUDIO_IO_HANDLE_NONE;
+ }
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mChannelMask = config.channel_mask;
+ outputDesc->mFormat = config.format;
+ outputDesc->mRefCount[stream] = 0;
+ outputDesc->mStopTime[stream] = 0;
+ outputDesc->mDirectOpenCount = 1;
+
+ audio_io_handle_t srcOutput = getOutputForEffect();
+ addOutput(output, outputDesc);
+ audio_io_handle_t dstOutput = getOutputForEffect();
+ if (dstOutput == output) {
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+ }
+ mPreviousOutputs = mOutputs;
+ ALOGV("getOutput() returns new direct output %d", output);
+ mpClientInterface->onAudioPortListUpdate();
+ return output;
+ }
+
+non_direct_output:
+
+ // ignoring channel mask due to downmix capability in mixer
+
+ // open a non direct output
+
+ // for non direct outputs, only PCM is supported
+ if (audio_is_linear_pcm(format)) {
+ // get which output is suitable for the specified stream. The actual
+ // routing change will happen when startOutput() will be called
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+ // at this stage we should ignore the DIRECT flag as no direct output could be found earlier
+ flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
+ output = selectOutput(outputs, flags, format);
+ }
+ ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
+ "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+ ALOGV("getOutput() returns output %d", output);
+
+ return output;
+}
+
+audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+ audio_output_flags_t flags,
+ audio_format_t format)
+{
+ // select one output among several that provide a path to a particular device or set of
+ // devices (the list was previously build by getOutputsForDevice()).
+ // The priority is as follows:
+ // 1: the output with the highest number of requested policy flags
+ // 2: the primary output
+ // 3: the first output in the list
+
+ if (outputs.size() == 0) {
+ return 0;
+ }
+ if (outputs.size() == 1) {
+ return outputs[0];
+ }
+
+ int maxCommonFlags = 0;
+ audio_io_handle_t outputFlags = 0;
+ audio_io_handle_t outputPrimary = 0;
+
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
+ if (!outputDesc->isDuplicated()) {
+ // if a valid format is specified, skip output if not compatible
+ if (format != AUDIO_FORMAT_INVALID) {
+ if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ if (format != outputDesc->mFormat) {
+ continue;
+ }
+ } else if (!audio_is_linear_pcm(format)) {
+ continue;
+ }
+ }
+
+ int commonFlags = popcount(outputDesc->mProfile->mFlags & flags);
+ if (commonFlags > maxCommonFlags) {
+ outputFlags = outputs[i];
+ maxCommonFlags = commonFlags;
+ ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
+ }
+ if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ outputPrimary = outputs[i];
+ }
+ }
+ }
+
+ if (outputFlags != 0) {
+ return outputFlags;
+ }
+ if (outputPrimary != 0) {
+ return outputPrimary;
+ }
+
+ return outputs[0];
+}
+
+status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session)
+{
+ ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("startOutput() unknown output %d", output);
+ return BAD_VALUE;
+ }
+
+ // cannot start playback of STREAM_TTS if any other output is being used
+ uint32_t beaconMuteLatency = 0;
+ if (stream == AUDIO_STREAM_TTS) {
+ ALOGV("\t found BEACON stream");
+ if (mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {
+ return INVALID_OPERATION;
+ } else {
+ beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
+ }
+ } else {
+ // some playback other than beacon starts
+ beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
+ }
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+
+ // increment usage count for this stream on the requested output:
+ // NOTE that the usage count is the same for duplicated output and hardware output which is
+ // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
+ outputDesc->changeRefCount(stream, 1);
+
+ if (outputDesc->mRefCount[stream] == 1) {
+ // starting an output being rerouted?
+ audio_devices_t newDevice;
+ if (outputDesc->mPolicyMix != NULL) {
+ newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ } else {
+ newDevice = getNewOutputDevice(output, false /*fromCache*/);
+ }
+ routing_strategy strategy = getStrategy(stream);
+ bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
+ (strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||
+ (beaconMuteLatency > 0);
+ uint32_t waitMs = beaconMuteLatency;
+ bool force = false;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != outputDesc) {
+ // force a device change if any other output is managed by the same hw
+ // module and has a current device selection that differs from selected device.
+ // In this case, the audio HAL must receive the new device selection so that it can
+ // change the device currently selected by the other active output.
+ if (outputDesc->sharesHwModuleWith(desc) &&
+ desc->device() != newDevice) {
+ force = true;
+ }
+ // wait for audio on other active outputs to be presented when starting
+ // a notification so that audio focus effect can propagate, or that a mute/unmute
+ // event occurred for beacon
+ uint32_t latency = desc->latency();
+ if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
+ waitMs = latency;
+ }
+ }
+ }
+ uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, true, false);
+ }
+
+ // apply volume rules for current stream and device if necessary
+ checkAndSetVolume(stream,
+ mStreams[stream].getVolumeIndex(newDevice),
+ output,
+ newDevice);
+
+ // update the outputs if starting an output with a stream that can affect notification
+ // routing
+ handleNotificationRoutingForStream(stream);
+
+ // Automatically enable the remote submix input when output is started on a re routing mix
+ // of type MIX_TYPE_RECORDERS
+ if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL &&
+ outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ outputDesc->mPolicyMix->mRegistrationId,
+ "remote-submix");
+ }
+
+ // force reevaluating accessibility routing when ringtone or alarm starts
+ if (strategy == STRATEGY_SONIFICATION) {
+ mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
+ }
+
+ if (waitMs > muteWaitMs) {
+ usleep((waitMs - muteWaitMs) * 2 * 1000);
+ }
+ }
+ return NO_ERROR;
+}
+
+
+status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session)
+{
+ ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("stopOutput() unknown output %d", output);
+ return BAD_VALUE;
+ }
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+
+ // always handle stream stop, check which stream type is stopping
+ handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, false, false);
+ }
+
+ if (outputDesc->mRefCount[stream] > 0) {
+ // decrement usage count of this stream on the output
+ outputDesc->changeRefCount(stream, -1);
+ // store time at which the stream was stopped - see isStreamActive()
+ if (outputDesc->mRefCount[stream] == 0) {
+ // Automatically disable the remote submix input when output is stopped on a
+ // re routing mix of type MIX_TYPE_RECORDERS
+ if (audio_is_remote_submix_device(outputDesc->mDevice) &&
+ outputDesc->mPolicyMix != NULL &&
+ outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ outputDesc->mPolicyMix->mRegistrationId,
+ "remote-submix");
+ }
+
+ outputDesc->mStopTime[stream] = systemTime();
+ audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
+ // delay the device switch by twice the latency because stopOutput() is executed when
+ // the track stop() command is received and at that time the audio track buffer can
+ // still contain data that needs to be drained. The latency only covers the audio HAL
+ // and kernel buffers. Also the latency does not always include additional delay in the
+ // audio path (audio DSP, CODEC ...)
+ setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
+
+ // force restoring the device selection on other active outputs if it differs from the
+ // one being selected for this output
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(i);
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (curOutput != output &&
+ desc->isActive() &&
+ outputDesc->sharesHwModuleWith(desc) &&
+ (newDevice != desc->device())) {
+ setOutputDevice(curOutput,
+ getNewOutputDevice(curOutput, false /*fromCache*/),
+ true,
+ outputDesc->mLatency*2);
+ }
+ }
+ // update the outputs if stopping one with a stream that can affect notification routing
+ handleNotificationRoutingForStream(stream);
+ }
+ return NO_ERROR;
+ } else {
+ ALOGW("stopOutput() refcount is already 0 for output %d", output);
+ return INVALID_OPERATION;
+ }
+}
+
+void AudioPolicyManager::releaseOutput(audio_io_handle_t output,
+ audio_stream_type_t stream __unused,
+ audio_session_t session __unused)
+{
+ ALOGV("releaseOutput() %d", output);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("releaseOutput() releasing unknown output %d", output);
+ return;
+ }
+
+#ifdef AUDIO_POLICY_TEST
+ int testIndex = testOutputIndex(output);
+ if (testIndex != 0) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+ if (outputDesc->isActive()) {
+ mpClientInterface->closeOutput(output);
+ removeOutput(output);
+ mTestOutputs[testIndex] = 0;
+ }
+ return;
+ }
+#endif //AUDIO_POLICY_TEST
+
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(index);
+ if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ if (desc->mDirectOpenCount <= 0) {
+ ALOGW("releaseOutput() invalid open count %d for output %d",
+ desc->mDirectOpenCount, output);
+ return;
+ }
+ if (--desc->mDirectOpenCount == 0) {
+ closeOutput(output);
+ // If effects where present on the output, audioflinger moved them to the primary
+ // output by default: move them back to the appropriate output.
+ audio_io_handle_t dstOutput = getOutputForEffect();
+ if (dstOutput != mPrimaryOutput) {
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput);
+ }
+ mpClientInterface->onAudioPortListUpdate();
+ }
+ }
+}
+
+
+status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ input_type_t *inputType)
+{
+ ALOGV("getInputForAttr() source %d, samplingRate %d, format %d, channelMask %x,"
+ "session %d, flags %#x",
+ attr->source, samplingRate, format, channelMask, session, flags);
+
+ *input = AUDIO_IO_HANDLE_NONE;
+ *inputType = API_INPUT_INVALID;
+ audio_devices_t device;
+ // handle legacy remote submix case where the address was not always specified
+ String8 address = String8("");
+ bool isSoundTrigger = false;
+ audio_source_t inputSource = attr->source;
+ audio_source_t halInputSource;
+ AudioMix *policyMix = NULL;
+
+ if (inputSource == AUDIO_SOURCE_DEFAULT) {
+ inputSource = AUDIO_SOURCE_MIC;
+ }
+ halInputSource = inputSource;
+
+ if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX &&
+ strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {
+ status_t ret = mPolicyMixes.getInputMixForAttr(*attr, policyMix);
+ if (ret != NO_ERROR) {
+ return ret;
+ }
+ *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
+ device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ address = String8(attr->tags + strlen("addr="));
+ } else {
+ device = getDeviceAndMixForInputSource(inputSource, &policyMix);
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGW("getInputForAttr() could not find device for source %d", inputSource);
+ return BAD_VALUE;
+ }
+ if (policyMix != NULL) {
+ address = policyMix->mRegistrationId;
+ if (policyMix->mMixType == MIX_TYPE_RECORDERS) {
+ // there is an external policy, but this input is attached to a mix of recorders,
+ // meaning it receives audio injected into the framework, so the recorder doesn't
+ // know about it and is therefore considered "legacy"
+ *inputType = API_INPUT_LEGACY;
+ } else {
+ // recording a mix of players defined by an external policy, we're rerouting for
+ // an external policy
+ *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
+ }
+ } else if (audio_is_remote_submix_device(device)) {
+ address = String8("0");
+ *inputType = API_INPUT_MIX_CAPTURE;
+ } else {
+ *inputType = API_INPUT_LEGACY;
+ }
+ // adapt channel selection to input source
+ switch (inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_CALL:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
+ break;
+ default:
+ break;
+ }
+ if (inputSource == AUDIO_SOURCE_HOTWORD) {
+ ssize_t index = mSoundTriggerSessions.indexOfKey(session);
+ if (index >= 0) {
+ *input = mSoundTriggerSessions.valueFor(session);
+ isSoundTrigger = true;
+ flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD);
+ ALOGV("SoundTrigger capture on session %d input %d", session, *input);
+ } else {
+ halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION;
+ }
+ }
+ }
+
+ sp<IOProfile> profile = getInputProfile(device, address,
+ samplingRate, format, channelMask,
+ flags);
+ if (profile == 0) {
+ //retry without flags
+ audio_input_flags_t log_flags = flags;
+ flags = AUDIO_INPUT_FLAG_NONE;
+ profile = getInputProfile(device, address,
+ samplingRate, format, channelMask,
+ flags);
+ if (profile == 0) {
+ ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u,"
+ "format %#x, channelMask 0x%X, flags %#x",
+ device, samplingRate, format, channelMask, log_flags);
+ return BAD_VALUE;
+ }
+ }
+
+ if (profile->mModule->mHandle == 0) {
+ ALOGE("getInputForAttr(): HW module %s not opened", profile->mModule->mName);
+ return NO_INIT;
+ }
+
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = samplingRate;
+ config.channel_mask = channelMask;
+ config.format = format;
+
+ status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
+ input,
+ &config,
+ &device,
+ address,
+ halInputSource,
+ flags);
+
+ // only accept input with the exact requested set of parameters
+ if (status != NO_ERROR || *input == AUDIO_IO_HANDLE_NONE ||
+ (samplingRate != config.sample_rate) ||
+ (format != config.format) ||
+ (channelMask != config.channel_mask)) {
+ ALOGW("getInputForAttr() failed opening input: samplingRate %d, format %d, channelMask %x",
+ samplingRate, format, channelMask);
+ if (*input != AUDIO_IO_HANDLE_NONE) {
+ mpClientInterface->closeInput(*input);
+ }
+ return BAD_VALUE;
+ }
+
+ sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile);
+ inputDesc->mInputSource = inputSource;
+ inputDesc->mRefCount = 0;
+ inputDesc->mOpenRefCount = 1;
+ inputDesc->mSamplingRate = samplingRate;
+ inputDesc->mFormat = format;
+ inputDesc->mChannelMask = channelMask;
+ inputDesc->mDevice = device;
+ inputDesc->mSessions.add(session);
+ inputDesc->mIsSoundTrigger = isSoundTrigger;
+ inputDesc->mPolicyMix = policyMix;
+
+ ALOGV("getInputForAttr() returns input type = %d", inputType);
+
+ addInput(*input, inputDesc);
+ mpClientInterface->onAudioPortListUpdate();
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::startInput(audio_io_handle_t input,
+ audio_session_t session)
+{
+ ALOGV("startInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("startInput() unknown input %d", input);
+ return BAD_VALUE;
+ }
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+
+ index = inputDesc->mSessions.indexOf(session);
+ if (index < 0) {
+ ALOGW("startInput() unknown session %d on input %d", session, input);
+ return BAD_VALUE;
+ }
+
+ // virtual input devices are compatible with other input devices
+ if (!is_virtual_input_device(inputDesc->mDevice)) {
+
+ // for a non-virtual input device, check if there is another (non-virtual) active input
+ audio_io_handle_t activeInput = mInputs.getActiveInput();
+ if (activeInput != 0 && activeInput != input) {
+
+ // If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed,
+ // otherwise the active input continues and the new input cannot be started.
+ sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
+ if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
+ ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput);
+ stopInput(activeInput, activeDesc->mSessions.itemAt(0));
+ releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
+ } else {
+ ALOGE("startInput(%d) failed: other input %d already started", input, activeInput);
+ return INVALID_OPERATION;
+ }
+ }
+ }
+
+ if (inputDesc->mRefCount == 0) {
+ if (mInputs.activeInputsCount() == 0) {
+ SoundTrigger::setCaptureState(true);
+ }
+ setInputDevice(input, getNewInputDevice(input), true /* force */);
+
+ // automatically enable the remote submix output when input is started if not
+ // used by a policy mix of type MIX_TYPE_RECORDERS
+ // For remote submix (a virtual device), we open only one input per capture request.
+ if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+ String8 address = String8("");
+ if (inputDesc->mPolicyMix == NULL) {
+ address = String8("0");
+ } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
+ address = inputDesc->mPolicyMix->mRegistrationId;
+ }
+ if (address != "") {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address, "remote-submix");
+ }
+ }
+ }
+
+ ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
+
+ inputDesc->mRefCount++;
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::stopInput(audio_io_handle_t input,
+ audio_session_t session)
+{
+ ALOGV("stopInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("stopInput() unknown input %d", input);
+ return BAD_VALUE;
+ }
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+
+ index = inputDesc->mSessions.indexOf(session);
+ if (index < 0) {
+ ALOGW("stopInput() unknown session %d on input %d", session, input);
+ return BAD_VALUE;
+ }
+
+ if (inputDesc->mRefCount == 0) {
+ ALOGW("stopInput() input %d already stopped", input);
+ return INVALID_OPERATION;
+ }
+
+ inputDesc->mRefCount--;
+ if (inputDesc->mRefCount == 0) {
+
+ // automatically disable the remote submix output when input is stopped if not
+ // used by a policy mix of type MIX_TYPE_RECORDERS
+ if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+ String8 address = String8("");
+ if (inputDesc->mPolicyMix == NULL) {
+ address = String8("0");
+ } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
+ address = inputDesc->mPolicyMix->mRegistrationId;
+ }
+ if (address != "") {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address, "remote-submix");
+ }
+ }
+
+ resetInputDevice(input);
+
+ if (mInputs.activeInputsCount() == 0) {
+ SoundTrigger::setCaptureState(false);
+ }
+ }
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::releaseInput(audio_io_handle_t input,
+ audio_session_t session)
+{
+ ALOGV("releaseInput() %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("releaseInput() releasing unknown input %d", input);
+ return;
+ }
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+ ALOG_ASSERT(inputDesc != 0);
+
+ index = inputDesc->mSessions.indexOf(session);
+ if (index < 0) {
+ ALOGW("releaseInput() unknown session %d on input %d", session, input);
+ return;
+ }
+ inputDesc->mSessions.remove(session);
+ if (inputDesc->mOpenRefCount == 0) {
+ ALOGW("releaseInput() invalid open ref count %d", inputDesc->mOpenRefCount);
+ return;
+ }
+ inputDesc->mOpenRefCount--;
+ if (inputDesc->mOpenRefCount > 0) {
+ ALOGV("releaseInput() exit > 0");
+ return;
+ }
+
+ closeInput(input);
+ mpClientInterface->onAudioPortListUpdate();
+ ALOGV("releaseInput() exit");
+}
+
+void AudioPolicyManager::closeAllInputs() {
+ bool patchRemoved = false;
+
+ for(size_t input_index = 0; input_index < mInputs.size(); input_index++) {
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index);
+ ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ if (patch_index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patch_index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ mAudioPatches.removeItemsAt(patch_index);
+ patchRemoved = true;
+ }
+ mpClientInterface->closeInput(mInputs.keyAt(input_index));
+ }
+ mInputs.clear();
+ nextAudioPortGeneration();
+
+ if (patchRemoved) {
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+}
+
+void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax)
+{
+ ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+ mEngine->initStreamVolume(stream, indexMin, indexMax);
+ //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now
+ if (stream == AUDIO_STREAM_MUSIC) {
+ mEngine->initStreamVolume(AUDIO_STREAM_ACCESSIBILITY, indexMin, indexMax);
+ }
+}
+
+status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
+{
+
+ if ((index < mStreams[stream].getVolumeIndexMin()) ||
+ (index > mStreams[stream].getVolumeIndexMax())) {
+ return BAD_VALUE;
+ }
+ if (!audio_is_output_device(device)) {
+ return BAD_VALUE;
+ }
+
+ // Force max volume if stream cannot be muted
+ if (!mStreams.canBeMuted(stream)) index = mStreams[stream].getVolumeIndexMax();
+
+ ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
+ stream, device, index);
+
+ // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
+ // clear all device specific values
+ if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+ mStreams.clearCurrentVolumeIndex(stream);
+ }
+ mStreams.addCurrentVolumeIndex(stream, device, index);
+
+ // update volume on all outputs whose current device is also selected by the same
+ // strategy as the device specified by the caller
+ audio_devices_t strategyDevice = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
+
+
+ //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now
+ audio_devices_t accessibilityDevice = AUDIO_DEVICE_NONE;
+ if (stream == AUDIO_STREAM_MUSIC) {
+ mStreams.addCurrentVolumeIndex(AUDIO_STREAM_ACCESSIBILITY, device, index);
+ accessibilityDevice = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, true /*fromCache*/);
+ }
+ if ((device != AUDIO_DEVICE_OUT_DEFAULT) &&
+ (device & (strategyDevice | accessibilityDevice)) == 0) {
+ return NO_ERROR;
+ }
+ status_t status = NO_ERROR;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_devices_t curDevice = Volume::getDeviceForVolume(mOutputs.valueAt(i)->device());
+ if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) {
+ status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
+ if (volStatus != NO_ERROR) {
+ status = volStatus;
+ }
+ }
+ if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) {
+ status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY,
+ index, mOutputs.keyAt(i), curDevice);
+ }
+ }
+ return status;
+}
+
+status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
+{
+ if (index == NULL) {
+ return BAD_VALUE;
+ }
+ if (!audio_is_output_device(device)) {
+ return BAD_VALUE;
+ }
+ // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to
+ // the strategy the stream belongs to.
+ if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+ device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
+ }
+ device = Volume::getDeviceForVolume(device);
+
+ *index = mStreams[stream].getVolumeIndex(device);
+ ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
+ return NO_ERROR;
+}
+
+audio_io_handle_t AudioPolicyManager::selectOutputForEffects(
+ const SortedVector<audio_io_handle_t>& outputs)
+{
+ // select one output among several suitable for global effects.
+ // The priority is as follows:
+ // 1: An offloaded output. If the effect ends up not being offloadable,
+ // AudioFlinger will invalidate the track and the offloaded output
+ // will be closed causing the effect to be moved to a PCM output.
+ // 2: A deep buffer output
+ // 3: the first output in the list
+
+ if (outputs.size() == 0) {
+ return 0;
+ }
+
+ audio_io_handle_t outputOffloaded = 0;
+ audio_io_handle_t outputDeepBuffer = 0;
+
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
+ ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags);
+ if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ outputOffloaded = outputs[i];
+ }
+ if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
+ outputDeepBuffer = outputs[i];
+ }
+ }
+
+ ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d",
+ outputOffloaded, outputDeepBuffer);
+ if (outputOffloaded != 0) {
+ return outputOffloaded;
+ }
+ if (outputDeepBuffer != 0) {
+ return outputDeepBuffer;
+ }
+
+ return outputs[0];
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc)
+{
+ // apply simple rule where global effects are attached to the same output as MUSIC streams
+
+ routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
+
+ audio_io_handle_t output = selectOutputForEffects(dstOutputs);
+ ALOGV("getOutputForEffect() got output %d for fx %s flags %x",
+ output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags);
+
+ return output;
+}
+
+status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id)
+{
+ ssize_t index = mOutputs.indexOfKey(io);
+ if (index < 0) {
+ index = mInputs.indexOfKey(io);
+ if (index < 0) {
+ ALOGW("registerEffect() unknown io %d", io);
+ return INVALID_OPERATION;
+ }
+ }
+ return mEffects.registerEffect(desc, io, strategy, session, id);
+}
+
+bool AudioPolicyManager::isSourceActive(audio_source_t source) const
+{
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
+ if (inputDescriptor->mRefCount == 0) {
+ continue;
+ }
+ if (inputDescriptor->mInputSource == (int)source) {
+ return true;
+ }
+ // AUDIO_SOURCE_HOTWORD is equivalent to AUDIO_SOURCE_VOICE_RECOGNITION only if it
+ // corresponds to an active capture triggered by a hardware hotword recognition
+ if ((source == AUDIO_SOURCE_VOICE_RECOGNITION) &&
+ (inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD)) {
+ // FIXME: we should not assume that the first session is the active one and keep
+ // activity count per session. Same in startInput().
+ ssize_t index = mSoundTriggerSessions.indexOfKey(inputDescriptor->mSessions.itemAt(0));
+ if (index >= 0) {
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+// Register a list of custom mixes with their attributes and format.
+// When a mix is registered, corresponding input and output profiles are
+// added to the remote submix hw module. The profile contains only the
+// parameters (sampling rate, format...) specified by the mix.
+// The corresponding input remote submix device is also connected.
+//
+// When a remote submix device is connected, the address is checked to select the
+// appropriate profile and the corresponding input or output stream is opened.
+//
+// When capture starts, getInputForAttr() will:
+// - 1 look for a mix matching the address passed in attribtutes tags if any
+// - 2 if none found, getDeviceForInputSource() will:
+// - 2.1 look for a mix matching the attributes source
+// - 2.2 if none found, default to device selection by policy rules
+// At this time, the corresponding output remote submix device is also connected
+// and active playback use cases can be transferred to this mix if needed when reconnecting
+// after AudioTracks are invalidated
+//
+// When playback starts, getOutputForAttr() will:
+// - 1 look for a mix matching the address passed in attribtutes tags if any
+// - 2 if none found, look for a mix matching the attributes usage
+// - 3 if none found, default to device and output selection by policy rules.
+
+status_t AudioPolicyManager::registerPolicyMixes(Vector<AudioMix> mixes)
+{
+ sp<HwModule> module;
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
+ mHwModules[i]->mHandle != 0) {
+ module = mHwModules[i];
+ break;
+ }
+ }
+
+ if (module == 0) {
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("registerPolicyMixes() num mixes %d", mixes.size());
+
+ for (size_t i = 0; i < mixes.size(); i++) {
+ String8 address = mixes[i].mRegistrationId;
+
+ if (mPolicyMixes.registerMix(address, mixes[i]) != NO_ERROR) {
+ continue;
+ }
+ audio_config_t outputConfig = mixes[i].mFormat;
+ audio_config_t inputConfig = mixes[i].mFormat;
+ // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in
+ // stereo and let audio flinger do the channel conversion if needed.
+ outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ module->addOutputProfile(address, &outputConfig,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address);
+ module->addInputProfile(address, &inputConfig,
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX, address);
+
+ if (mixes[i].mMixType == MIX_TYPE_PLAYERS) {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address.string(), "remote-submix");
+ } else {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address.string(), "remote-submix");
+ }
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes)
+{
+ sp<HwModule> module;
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
+ mHwModules[i]->mHandle != 0) {
+ module = mHwModules[i];
+ break;
+ }
+ }
+
+ if (module == 0) {
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("unregisterPolicyMixes() num mixes %d", mixes.size());
+
+ for (size_t i = 0; i < mixes.size(); i++) {
+ String8 address = mixes[i].mRegistrationId;
+
+ if (mPolicyMixes.unregisterMix(address) != NO_ERROR) {
+ continue;
+ }
+
+ if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) ==
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
+ {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address.string(), "remote-submix");
+ }
+
+ if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
+ {
+ setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address.string(), "remote-submix");
+ }
+ module->removeOutputProfile(address);
+ module->removeInputProfile(address);
+ }
+ return NO_ERROR;
+}
+
+
+status_t AudioPolicyManager::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
+ result.append(buffer);
+
+ snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Phone state: %d\n", mEngine->getPhoneState());
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for communications %d\n",
+ mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION));
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for media %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA));
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for record %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD));
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for dock %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_DOCK));
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for system %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM));
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n",
+ mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO));
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+
+ mAvailableOutputDevices.dump(fd, String8("output"));
+ mAvailableInputDevices.dump(fd, String8("input"));
+ mHwModules.dump(fd);
+ mOutputs.dump(fd);
+ mInputs.dump(fd);
+ mStreams.dump(fd);
+ mEffects.dump(fd);
+ mAudioPatches.dump(fd);
+
+ return NO_ERROR;
+}
+
+// This function checks for the parameters which can be offloaded.
+// This can be enhanced depending on the capability of the DSP and policy
+// of the system.
+bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+ ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+ " BitRate=%u, duration=%" PRId64 " us, has_video=%d",
+ offloadInfo.sample_rate, offloadInfo.channel_mask,
+ offloadInfo.format,
+ offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+ offloadInfo.has_video);
+
+ // Check if offload has been disabled
+ char propValue[PROPERTY_VALUE_MAX];
+ if (property_get("audio.offload.disable", propValue, "0")) {
+ if (atoi(propValue) != 0) {
+ ALOGV("offload disabled by audio.offload.disable=%s", propValue );
+ return false;
+ }
+ }
+
+ // Check if stream type is music, then only allow offload as of now.
+ if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
+ {
+ ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
+ return false;
+ }
+
+ //TODO: enable audio offloading with video when ready
+ if (offloadInfo.has_video)
+ {
+ ALOGV("isOffloadSupported: has_video == true, returning false");
+ return false;
+ }
+
+ //If duration is less than minimum value defined in property, return false
+ if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
+ if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
+ ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
+ return false;
+ }
+ } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+ ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
+ return false;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+ if (mEffects.isNonOffloadableEffectEnabled()) {
+ return false;
+ }
+
+ // See if there is a profile to support this.
+ // AUDIO_DEVICE_NONE
+ sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+ offloadInfo.sample_rate,
+ offloadInfo.format,
+ offloadInfo.channel_mask,
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+ ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
+ return (profile != 0);
+}
+
+status_t AudioPolicyManager::listAudioPorts(audio_port_role_t role,
+ audio_port_type_t type,
+ unsigned int *num_ports,
+ struct audio_port *ports,
+ unsigned int *generation)
+{
+ if (num_ports == NULL || (*num_ports != 0 && ports == NULL) ||
+ generation == NULL) {
+ return BAD_VALUE;
+ }
+ ALOGV("listAudioPorts() role %d type %d num_ports %d ports %p", role, type, *num_ports, ports);
+ if (ports == NULL) {
+ *num_ports = 0;
+ }
+
+ size_t portsWritten = 0;
+ size_t portsMax = *num_ports;
+ *num_ports = 0;
+ if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_DEVICE) {
+ if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
+ for (size_t i = 0;
+ i < mAvailableOutputDevices.size() && portsWritten < portsMax; i++) {
+ mAvailableOutputDevices[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ *num_ports += mAvailableOutputDevices.size();
+ }
+ if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
+ for (size_t i = 0;
+ i < mAvailableInputDevices.size() && portsWritten < portsMax; i++) {
+ mAvailableInputDevices[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ *num_ports += mAvailableInputDevices.size();
+ }
+ }
+ if (type == AUDIO_PORT_TYPE_NONE || type == AUDIO_PORT_TYPE_MIX) {
+ if (role == AUDIO_PORT_ROLE_SINK || role == AUDIO_PORT_ROLE_NONE) {
+ for (size_t i = 0; i < mInputs.size() && portsWritten < portsMax; i++) {
+ mInputs[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ *num_ports += mInputs.size();
+ }
+ if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
+ size_t numOutputs = 0;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ if (!mOutputs[i]->isDuplicated()) {
+ numOutputs++;
+ if (portsWritten < portsMax) {
+ mOutputs[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ }
+ }
+ *num_ports += numOutputs;
+ }
+ }
+ *generation = curAudioPortGeneration();
+ ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused)
+{
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch,
+ audio_patch_handle_t *handle,
+ uid_t uid)
+{
+ ALOGV("createAudioPatch()");
+
+ if (handle == NULL || patch == NULL) {
+ return BAD_VALUE;
+ }
+ ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks);
+
+ if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX ||
+ patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) {
+ return BAD_VALUE;
+ }
+ // only one source per audio patch supported for now
+ if (patch->num_sources > 1) {
+ return INVALID_OPERATION;
+ }
+
+ if (patch->sources[0].role != AUDIO_PORT_ROLE_SOURCE) {
+ return INVALID_OPERATION;
+ }
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ if (patch->sinks[i].role != AUDIO_PORT_ROLE_SINK) {
+ return INVALID_OPERATION;
+ }
+ }
+
+ sp<AudioPatch> patchDesc;
+ ssize_t index = mAudioPatches.indexOfKey(*handle);
+
+ ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id,
+ patch->sources[0].role,
+ patch->sources[0].type);
+#if LOG_NDEBUG == 0
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ ALOGV("createAudioPatch sink %d: id %d role %d type %d", i, patch->sinks[i].id,
+ patch->sinks[i].role,
+ patch->sinks[i].type);
+ }
+#endif
+
+ if (index >= 0) {
+ patchDesc = mAudioPatches.valueAt(index);
+ ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
+ mUidCached, patchDesc->mUid, uid);
+ if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
+ return INVALID_OPERATION;
+ }
+ } else {
+ *handle = 0;
+ }
+
+ if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
+ if (outputDesc == NULL) {
+ ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
+ return BAD_VALUE;
+ }
+ ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports",
+ outputDesc->mIoHandle);
+ if (patchDesc != 0) {
+ if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
+ ALOGV("createAudioPatch() source id differs for patch current id %d new id %d",
+ patchDesc->mPatch.sources[0].id, patch->sources[0].id);
+ return BAD_VALUE;
+ }
+ }
+ DeviceVector devices;
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ // Only support mix to devices connection
+ // TODO add support for mix to mix connection
+ if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
+ ALOGV("createAudioPatch() source mix but sink is not a device");
+ return INVALID_OPERATION;
+ }
+ sp<DeviceDescriptor> devDesc =
+ mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
+ if (devDesc == 0) {
+ ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id);
+ return BAD_VALUE;
+ }
+
+ if (!outputDesc->mProfile->isCompatibleProfile(devDesc->type(),
+ devDesc->mAddress,
+ patch->sources[0].sample_rate,
+ NULL, // updatedSamplingRate
+ patch->sources[0].format,
+ patch->sources[0].channel_mask,
+ AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) {
+ ALOGV("createAudioPatch() profile not supported for device %08x", devDesc->type());
+ return INVALID_OPERATION;
+ }
+ devices.add(devDesc);
+ }
+ if (devices.size() == 0) {
+ return INVALID_OPERATION;
+ }
+
+ // TODO: reconfigure output format and channels here
+ ALOGV("createAudioPatch() setting device %08x on output %d",
+ devices.types(), outputDesc->mIoHandle);
+ setOutputDevice(outputDesc->mIoHandle, devices.types(), true, 0, handle);
+ index = mAudioPatches.indexOfKey(*handle);
+ if (index >= 0) {
+ if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
+ ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided");
+ }
+ patchDesc = mAudioPatches.valueAt(index);
+ patchDesc->mUid = uid;
+ ALOGV("createAudioPatch() success");
+ } else {
+ ALOGW("createAudioPatch() setOutputDevice() failed to create a patch");
+ return INVALID_OPERATION;
+ }
+ } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
+ // input device to input mix connection
+ // only one sink supported when connecting an input device to a mix
+ if (patch->num_sinks > 1) {
+ return INVALID_OPERATION;
+ }
+ sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id);
+ if (inputDesc == NULL) {
+ return BAD_VALUE;
+ }
+ if (patchDesc != 0) {
+ if (patchDesc->mPatch.sinks[0].id != patch->sinks[0].id) {
+ return BAD_VALUE;
+ }
+ }
+ sp<DeviceDescriptor> devDesc =
+ mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
+ if (devDesc == 0) {
+ return BAD_VALUE;
+ }
+
+ if (!inputDesc->mProfile->isCompatibleProfile(devDesc->type(),
+ devDesc->mAddress,
+ patch->sinks[0].sample_rate,
+ NULL, /*updatedSampleRate*/
+ patch->sinks[0].format,
+ patch->sinks[0].channel_mask,
+ // FIXME for the parameter type,
+ // and the NONE
+ (audio_output_flags_t)
+ AUDIO_INPUT_FLAG_NONE)) {
+ return INVALID_OPERATION;
+ }
+ // TODO: reconfigure output format and channels here
+ ALOGV("createAudioPatch() setting device %08x on output %d",
+ devDesc->type(), inputDesc->mIoHandle);
+ setInputDevice(inputDesc->mIoHandle, devDesc->type(), true, handle);
+ index = mAudioPatches.indexOfKey(*handle);
+ if (index >= 0) {
+ if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
+ ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided");
+ }
+ patchDesc = mAudioPatches.valueAt(index);
+ patchDesc->mUid = uid;
+ ALOGV("createAudioPatch() success");
+ } else {
+ ALOGW("createAudioPatch() setInputDevice() failed to create a patch");
+ return INVALID_OPERATION;
+ }
+ } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ // device to device connection
+ if (patchDesc != 0) {
+ if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
+ return BAD_VALUE;
+ }
+ }
+ sp<DeviceDescriptor> srcDeviceDesc =
+ mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
+ if (srcDeviceDesc == 0) {
+ return BAD_VALUE;
+ }
+
+ //update source and sink with our own data as the data passed in the patch may
+ // be incomplete.
+ struct audio_patch newPatch = *patch;
+ srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
+
+ for (size_t i = 0; i < patch->num_sinks; i++) {
+ if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
+ ALOGV("createAudioPatch() source device but one sink is not a device");
+ return INVALID_OPERATION;
+ }
+
+ sp<DeviceDescriptor> sinkDeviceDesc =
+ mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
+ if (sinkDeviceDesc == 0) {
+ return BAD_VALUE;
+ }
+ sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
+
+ if (srcDeviceDesc->mModule != sinkDeviceDesc->mModule) {
+ // only one sink supported when connected devices across HW modules
+ if (patch->num_sinks > 1) {
+ return INVALID_OPERATION;
+ }
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevice(sinkDeviceDesc->type(), mOutputs);
+ // if the sink device is reachable via an opened output stream, request to go via
+ // this output stream by adding a second source to the patch description
+ audio_io_handle_t output = selectOutput(outputs,
+ AUDIO_OUTPUT_FLAG_NONE,
+ AUDIO_FORMAT_INVALID);
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ if (outputDesc->isDuplicated()) {
+ return INVALID_OPERATION;
+ }
+ outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]);
+ newPatch.num_sources = 2;
+ }
+ }
+ }
+ // TODO: check from routing capabilities in config file and other conflicting patches
+
+ audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ if (index >= 0) {
+ afPatchHandle = patchDesc->mAfPatchHandle;
+ }
+
+ status_t status = mpClientInterface->createAudioPatch(&newPatch,
+ &afPatchHandle,
+ 0);
+ ALOGV("createAudioPatch() patch panel returned %d patchHandle %d",
+ status, afPatchHandle);
+ if (status == NO_ERROR) {
+ if (index < 0) {
+ patchDesc = new AudioPatch(&newPatch, uid);
+ addAudioPatch(patchDesc->mHandle, patchDesc);
+ } else {
+ patchDesc->mPatch = newPatch;
+ }
+ patchDesc->mAfPatchHandle = afPatchHandle;
+ *handle = patchDesc->mHandle;
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ } else {
+ ALOGW("createAudioPatch() patch panel could not connect device patch, error %d",
+ status);
+ return INVALID_OPERATION;
+ }
+ } else {
+ return BAD_VALUE;
+ }
+ } else {
+ return BAD_VALUE;
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle,
+ uid_t uid)
+{
+ ALOGV("releaseAudioPatch() patch %d", handle);
+
+ ssize_t index = mAudioPatches.indexOfKey(handle);
+
+ if (index < 0) {
+ return BAD_VALUE;
+ }
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
+ mUidCached, patchDesc->mUid, uid);
+ if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
+ return INVALID_OPERATION;
+ }
+
+ struct audio_patch *patch = &patchDesc->mPatch;
+ patchDesc->mUid = mUidCached;
+ if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
+ if (outputDesc == NULL) {
+ ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id);
+ return BAD_VALUE;
+ }
+
+ setOutputDevice(outputDesc->mIoHandle,
+ getNewOutputDevice(outputDesc->mIoHandle, true /*fromCache*/),
+ true,
+ 0,
+ NULL);
+ } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
+ sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id);
+ if (inputDesc == NULL) {
+ ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id);
+ return BAD_VALUE;
+ }
+ setInputDevice(inputDesc->mIoHandle,
+ getNewInputDevice(inputDesc->mIoHandle),
+ true,
+ NULL);
+ } else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
+ audio_patch_handle_t afPatchHandle = patchDesc->mAfPatchHandle;
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d",
+ status, patchDesc->mAfPatchHandle);
+ removeAudioPatch(patchDesc->mHandle);
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ } else {
+ return BAD_VALUE;
+ }
+ } else {
+ return BAD_VALUE;
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches,
+ struct audio_patch *patches,
+ unsigned int *generation)
+{
+ if (generation == NULL) {
+ return BAD_VALUE;
+ }
+ *generation = curAudioPortGeneration();
+ return mAudioPatches.listAudioPatches(num_patches, patches);
+}
+
+status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config)
+{
+ ALOGV("setAudioPortConfig()");
+
+ if (config == NULL) {
+ return BAD_VALUE;
+ }
+ ALOGV("setAudioPortConfig() on port handle %d", config->id);
+ // Only support gain configuration for now
+ if (config->config_mask != AUDIO_PORT_CONFIG_GAIN) {
+ return INVALID_OPERATION;
+ }
+
+ sp<AudioPortConfig> audioPortConfig;
+ if (config->type == AUDIO_PORT_TYPE_MIX) {
+ if (config->role == AUDIO_PORT_ROLE_SOURCE) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(config->id);
+ if (outputDesc == NULL) {
+ return BAD_VALUE;
+ }
+ ALOG_ASSERT(!outputDesc->isDuplicated(),
+ "setAudioPortConfig() called on duplicated output %d",
+ outputDesc->mIoHandle);
+ audioPortConfig = outputDesc;
+ } else if (config->role == AUDIO_PORT_ROLE_SINK) {
+ sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(config->id);
+ if (inputDesc == NULL) {
+ return BAD_VALUE;
+ }
+ audioPortConfig = inputDesc;
+ } else {
+ return BAD_VALUE;
+ }
+ } else if (config->type == AUDIO_PORT_TYPE_DEVICE) {
+ sp<DeviceDescriptor> deviceDesc;
+ if (config->role == AUDIO_PORT_ROLE_SOURCE) {
+ deviceDesc = mAvailableInputDevices.getDeviceFromId(config->id);
+ } else if (config->role == AUDIO_PORT_ROLE_SINK) {
+ deviceDesc = mAvailableOutputDevices.getDeviceFromId(config->id);
+ } else {
+ return BAD_VALUE;
+ }
+ if (deviceDesc == NULL) {
+ return BAD_VALUE;
+ }
+ audioPortConfig = deviceDesc;
+ } else {
+ return BAD_VALUE;
+ }
+
+ struct audio_port_config backupConfig;
+ status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig);
+ if (status == NO_ERROR) {
+ struct audio_port_config newConfig;
+ audioPortConfig->toAudioPortConfig(&newConfig, config);
+ status = mpClientInterface->setAudioPortConfig(&newConfig, 0);
+ }
+ if (status != NO_ERROR) {
+ audioPortConfig->applyAudioPortConfig(&backupConfig);
+ }
+
+ return status;
+}
+
+void AudioPolicyManager::clearAudioPatches(uid_t uid)
+{
+ for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i);
+ if (patchDesc->mUid == uid) {
+ releaseAudioPatch(mAudioPatches.keyAt(i), uid);
+ }
+ }
+}
+
+status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session,
+ audio_io_handle_t *ioHandle,
+ audio_devices_t *device)
+{
+ *session = (audio_session_t)mpClientInterface->newAudioUniqueId();
+ *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId();
+ *device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD);
+
+ return mSoundTriggerSessions.acquireSession(*session, *ioHandle);
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager
+// ----------------------------------------------------------------------------
+uint32_t AudioPolicyManager::nextAudioPortGeneration()
+{
+ return android_atomic_inc(&mAudioPortGeneration);
+}
+
+AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+ :
+#ifdef AUDIO_POLICY_TEST
+ Thread(false),
+#endif //AUDIO_POLICY_TEST
+ mPrimaryOutput((audio_io_handle_t)0),
+ mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
+ mA2dpSuspended(false),
+ mSpeakerDrcEnabled(false),
+ mAudioPortGeneration(1),
+ mBeaconMuteRefCount(0),
+ mBeaconPlayingRefCount(0),
+ mBeaconMuted(false)
+{
+ audio_policy::EngineInstance *engineInstance = audio_policy::EngineInstance::getInstance();
+ if (!engineInstance) {
+ ALOGE("%s: Could not get an instance of policy engine", __FUNCTION__);
+ return;
+ }
+ // Retrieve the Policy Manager Interface
+ mEngine = engineInstance->queryInterface<AudioPolicyManagerInterface>();
+ if (mEngine == NULL) {
+ ALOGE("%s: Failed to get Policy Engine Interface", __FUNCTION__);
+ return;
+ }
+ mEngine->setObserver(this);
+ status_t status = mEngine->initCheck();
+ ALOG_ASSERT(status == NO_ERROR, "Policy engine not initialized(err=%d)", status);
+
+ mUidCached = getuid();
+ mpClientInterface = clientInterface;
+
+ mDefaultOutputDevice = new DeviceDescriptor(String8("Speaker"), AUDIO_DEVICE_OUT_SPEAKER);
+ if (ConfigParsingUtils::loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE,
+ mHwModules, mAvailableInputDevices, mAvailableOutputDevices,
+ mDefaultOutputDevice, mSpeakerDrcEnabled) != NO_ERROR) {
+ if (ConfigParsingUtils::loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE,
+ mHwModules, mAvailableInputDevices, mAvailableOutputDevices,
+ mDefaultOutputDevice, mSpeakerDrcEnabled) != NO_ERROR) {
+ ALOGE("could not load audio policy configuration file, setting defaults");
+ defaultAudioPolicyConfig();
+ }
+ }
+ // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
+
+ // must be done after reading the policy (since conditionned by Speaker Drc Enabling)
+ mEngine->initializeVolumeCurves(mSpeakerDrcEnabled);
+
+ // open all output streams needed to access attached devices
+ audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
+ audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
+ if (mHwModules[i]->mHandle == 0) {
+ ALOGW("could not open HW module %s", mHwModules[i]->mName);
+ continue;
+ }
+ // open all output streams needed to access attached devices
+ // except for direct output streams that are only opened when they are actually
+ // required by an app.
+ // This also validates mAvailableOutputDevices list
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ const sp<IOProfile> outProfile = mHwModules[i]->mOutputProfiles[j];
+
+ if (outProfile->mSupportedDevices.isEmpty()) {
+ ALOGW("Output profile contains no device on module %s", mHwModules[i]->mName);
+ continue;
+ }
+
+ if ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
+ continue;
+ }
+ audio_devices_t profileType = outProfile->mSupportedDevices.types();
+ if ((profileType & mDefaultOutputDevice->type()) != AUDIO_DEVICE_NONE) {
+ profileType = mDefaultOutputDevice->type();
+ } else {
+ // chose first device present in mSupportedDevices also part of
+ // outputDeviceTypes
+ for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
+ profileType = outProfile->mSupportedDevices[k]->type();
+ if ((profileType & outputDeviceTypes) != 0) {
+ break;
+ }
+ }
+ }
+ if ((profileType & outputDeviceTypes) == 0) {
+ continue;
+ }
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(outProfile);
+
+ outputDesc->mDevice = profileType;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = outputDesc->mSamplingRate;
+ config.channel_mask = outputDesc->mChannelMask;
+ config.format = outputDesc->mFormat;
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle,
+ &output,
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+
+ if (status != NO_ERROR) {
+ ALOGW("Cannot open output stream for device %08x on hw module %s",
+ outputDesc->mDevice,
+ mHwModules[i]->mName);
+ } else {
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mChannelMask = config.channel_mask;
+ outputDesc->mFormat = config.format;
+
+ for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) {
+ audio_devices_t type = outProfile->mSupportedDevices[k]->type();
+ ssize_t index =
+ mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]);
+ // give a valid ID to an attached device once confirmed it is reachable
+ if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) {
+ mAvailableOutputDevices[index]->attach(mHwModules[i]);
+ }
+ }
+ if (mPrimaryOutput == 0 &&
+ outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ mPrimaryOutput = output;
+ }
+ addOutput(output, outputDesc);
+ setOutputDevice(output,
+ outputDesc->mDevice,
+ true);
+ }
+ }
+ // open input streams needed to access attached devices to validate
+ // mAvailableInputDevices list
+ for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
+ {
+ const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
+
+ if (inProfile->mSupportedDevices.isEmpty()) {
+ ALOGW("Input profile contains no device on module %s", mHwModules[i]->mName);
+ continue;
+ }
+ // chose first device present in mSupportedDevices also part of
+ // inputDeviceTypes
+ audio_devices_t profileType = AUDIO_DEVICE_NONE;
+ for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
+ profileType = inProfile->mSupportedDevices[k]->type();
+ if (profileType & inputDeviceTypes) {
+ break;
+ }
+ }
+ if ((profileType & inputDeviceTypes) == 0) {
+ continue;
+ }
+ sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(inProfile);
+
+ inputDesc->mInputSource = AUDIO_SOURCE_MIC;
+ inputDesc->mDevice = profileType;
+
+ // find the address
+ DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType);
+ // the inputs vector must be of size 1, but we don't want to crash here
+ String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress
+ : String8("");
+ ALOGV(" for input device 0x%x using address %s", profileType, address.string());
+ ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!");
+
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = inputDesc->mSamplingRate;
+ config.channel_mask = inputDesc->mChannelMask;
+ config.format = inputDesc->mFormat;
+ audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openInput(inProfile->mModule->mHandle,
+ &input,
+ &config,
+ &inputDesc->mDevice,
+ address,
+ AUDIO_SOURCE_MIC,
+ AUDIO_INPUT_FLAG_NONE);
+
+ if (status == NO_ERROR) {
+ for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) {
+ audio_devices_t type = inProfile->mSupportedDevices[k]->type();
+ ssize_t index =
+ mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]);
+ // give a valid ID to an attached device once confirmed it is reachable
+ if (index >= 0 && !mAvailableInputDevices[index]->isAttached()) {
+ mAvailableInputDevices[index]->attach(mHwModules[i]);
+ }
+ }
+ mpClientInterface->closeInput(input);
+ } else {
+ ALOGW("Cannot open input stream for device %08x on hw module %s",
+ inputDesc->mDevice,
+ mHwModules[i]->mName);
+ }
+ }
+ }
+ // make sure all attached devices have been allocated a unique ID
+ for (size_t i = 0; i < mAvailableOutputDevices.size();) {
+ if (!mAvailableOutputDevices[i]->isAttached()) {
+ ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->type());
+ mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
+ continue;
+ }
+ // The device is now validated and can be appended to the available devices of the engine
+ mEngine->setDeviceConnectionState(mAvailableOutputDevices[i],
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
+ i++;
+ }
+ for (size_t i = 0; i < mAvailableInputDevices.size();) {
+ if (!mAvailableInputDevices[i]->isAttached()) {
+ ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->type());
+ mAvailableInputDevices.remove(mAvailableInputDevices[i]);
+ continue;
+ }
+ // The device is now validated and can be appended to the available devices of the engine
+ mEngine->setDeviceConnectionState(mAvailableInputDevices[i],
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
+ i++;
+ }
+ // make sure default device is reachable
+ if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
+ ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->type());
+ }
+
+ ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
+
+ updateDevicesAndOutputs();
+
+#ifdef AUDIO_POLICY_TEST
+ if (mPrimaryOutput != 0) {
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"), 0);
+ mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+
+ mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
+ mTestSamplingRate = 44100;
+ mTestFormat = AUDIO_FORMAT_PCM_16_BIT;
+ mTestChannels = AUDIO_CHANNEL_OUT_STEREO;
+ mTestLatencyMs = 0;
+ mCurOutput = 0;
+ mDirectOutput = false;
+ for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+ mTestOutputs[i] = 0;
+ }
+
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ snprintf(buffer, SIZE, "AudioPolicyManagerTest");
+ run(buffer, ANDROID_PRIORITY_AUDIO);
+ }
+#endif //AUDIO_POLICY_TEST
+}
+
+AudioPolicyManager::~AudioPolicyManager()
+{
+#ifdef AUDIO_POLICY_TEST
+ exit();
+#endif //AUDIO_POLICY_TEST
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ mpClientInterface->closeOutput(mOutputs.keyAt(i));
+ }
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ mpClientInterface->closeInput(mInputs.keyAt(i));
+ }
+ mAvailableOutputDevices.clear();
+ mAvailableInputDevices.clear();
+ mOutputs.clear();
+ mInputs.clear();
+ mHwModules.clear();
+}
+
+status_t AudioPolicyManager::initCheck()
+{
+ return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
+}
+
+#ifdef AUDIO_POLICY_TEST
+bool AudioPolicyManager::threadLoop()
+{
+ ALOGV("entering threadLoop()");
+ while (!exitPending())
+ {
+ String8 command;
+ int valueInt;
+ String8 value;
+
+ Mutex::Autolock _l(mLock);
+ mWaitWorkCV.waitRelative(mLock, milliseconds(50));
+
+ command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
+ AudioParameter param = AudioParameter(command);
+
+ if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
+ valueInt != 0) {
+ ALOGV("Test command %s received", command.string());
+ String8 target;
+ if (param.get(String8("target"), target) != NO_ERROR) {
+ target = "Manager";
+ }
+ if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_output"));
+ mCurOutput = valueInt;
+ }
+ if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_direct"));
+ if (value == "false") {
+ mDirectOutput = false;
+ } else if (value == "true") {
+ mDirectOutput = true;
+ }
+ }
+ if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_input"));
+ mTestInput = valueInt;
+ }
+
+ if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_format"));
+ int format = AUDIO_FORMAT_INVALID;
+ if (value == "PCM 16 bits") {
+ format = AUDIO_FORMAT_PCM_16_BIT;
+ } else if (value == "PCM 8 bits") {
+ format = AUDIO_FORMAT_PCM_8_BIT;
+ } else if (value == "Compressed MP3") {
+ format = AUDIO_FORMAT_MP3;
+ }
+ if (format != AUDIO_FORMAT_INVALID) {
+ if (target == "Manager") {
+ mTestFormat = format;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("format"), format);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+ if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_channels"));
+ int channels = 0;
+
+ if (value == "Channels Stereo") {
+ channels = AUDIO_CHANNEL_OUT_STEREO;
+ } else if (value == "Channels Mono") {
+ channels = AUDIO_CHANNEL_OUT_MONO;
+ }
+ if (channels != 0) {
+ if (target == "Manager") {
+ mTestChannels = channels;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("channels"), channels);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+ if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_sampleRate"));
+ if (valueInt >= 0 && valueInt <= 96000) {
+ int samplingRate = valueInt;
+ if (target == "Manager") {
+ mTestSamplingRate = samplingRate;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("sampling_rate"), samplingRate);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+
+ if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_reopen"));
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ mpClientInterface->closeOutput(mPrimaryOutput);
+
+ audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
+
+ removeOutput(mPrimaryOutput);
+ sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL);
+ outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = outputDesc->mSamplingRate;
+ config.channel_mask = outputDesc->mChannelMask;
+ config.format = outputDesc->mFormat;
+ status_t status = mpClientInterface->openOutput(moduleHandle,
+ &mPrimaryOutput,
+ &config,
+ &outputDesc->mDevice,
+ String8(""),
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (status != NO_ERROR) {
+ ALOGE("Failed to reopen hardware output stream, "
+ "samplingRate: %d, format %d, channels %d",
+ outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
+ } else {
+ outputDesc->mSamplingRate = config.sample_rate;
+ outputDesc->mChannelMask = config.channel_mask;
+ outputDesc->mFormat = config.format;
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"), 0);
+ mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+ addOutput(mPrimaryOutput, outputDesc);
+ }
+ }
+
+
+ mpClientInterface->setParameters(0, String8("test_cmd_policy="));
+ }
+ }
+ return false;
+}
+
+void AudioPolicyManager::exit()
+{
+ {
+ AutoMutex _l(mLock);
+ requestExit();
+ mWaitWorkCV.signal();
+ }
+ requestExitAndWait();
+}
+
+int AudioPolicyManager::testOutputIndex(audio_io_handle_t output)
+{
+ for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+ if (output == mTestOutputs[i]) return i;
+ }
+ return 0;
+}
+#endif //AUDIO_POLICY_TEST
+
+// ---
+
+void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc)
+{
+ outputDesc->setIoHandle(output);
+ mOutputs.add(output, outputDesc);
+ nextAudioPortGeneration();
+}
+
+void AudioPolicyManager::removeOutput(audio_io_handle_t output)
+{
+ mOutputs.removeItem(output);
+}
+
+void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc)
+{
+ inputDesc->setIoHandle(input);
+ mInputs.add(input, inputDesc);
+ nextAudioPortGeneration();
+}
+
+void AudioPolicyManager::findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
+ const audio_devices_t device /*in*/,
+ const String8 address /*in*/,
+ SortedVector<audio_io_handle_t>& outputs /*out*/) {
+ sp<DeviceDescriptor> devDesc =
+ desc->mProfile->mSupportedDevices.getDevice(device, address);
+ if (devDesc != 0) {
+ ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s",
+ desc->mIoHandle, address.string());
+ outputs.add(desc->mIoHandle);
+ }
+}
+
+status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& outputs,
+ const String8 address)
+{
+ audio_devices_t device = devDesc->type();
+ sp<AudioOutputDescriptor> desc;
+ // erase all current sample rates, formats and channel masks
+ devDesc->clearCapabilities();
+
+ if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+ // first list already open outputs that can be routed to this device
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices.types() & device)) {
+ if (!device_distinguishes_on_address(device)) {
+ ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
+ outputs.add(mOutputs.keyAt(i));
+ } else {
+ ALOGV(" checking address match due to device 0x%x", device);
+ findIoHandlesByAddress(desc, device, address, outputs);
+ }
+ }
+ }
+ // then look for output profiles that can be routed to this device
+ SortedVector< sp<IOProfile> > profiles;
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ if (profile->mSupportedDevices.types() & device) {
+ if (!device_distinguishes_on_address(device) ||
+ address == profile->mSupportedDevices[0]->mAddress) {
+ profiles.add(profile);
+ ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
+ }
+ }
+ }
+ }
+
+ ALOGV(" found %d profiles, %d outputs", profiles.size(), outputs.size());
+
+ if (profiles.isEmpty() && outputs.isEmpty()) {
+ ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+ return BAD_VALUE;
+ }
+
+ // open outputs for matching profiles if needed. Direct outputs are also opened to
+ // query for dynamic parameters and will be closed later by setDeviceConnectionState()
+ for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
+ sp<IOProfile> profile = profiles[profile_index];
+
+ // nothing to do if one output is already opened for this profile
+ size_t j;
+ for (j = 0; j < outputs.size(); j++) {
+ desc = mOutputs.valueFor(outputs.itemAt(j));
+ if (!desc->isDuplicated() && desc->mProfile == profile) {
+ // matching profile: save the sample rates, format and channel masks supported
+ // by the profile in our device descriptor
+ devDesc->importAudioPort(profile);
+ break;
+ }
+ }
+ if (j != outputs.size()) {
+ continue;
+ }
+
+ ALOGV("opening output for device %08x with params %s profile %p",
+ device, address.string(), profile.get());
+ desc = new AudioOutputDescriptor(profile);
+ desc->mDevice = device;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = desc->mSamplingRate;
+ config.channel_mask = desc->mChannelMask;
+ config.format = desc->mFormat;
+ config.offload_info.sample_rate = desc->mSamplingRate;
+ config.offload_info.channel_mask = desc->mChannelMask;
+ config.offload_info.format = desc->mFormat;
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &output,
+ &config,
+ &desc->mDevice,
+ address,
+ &desc->mLatency,
+ desc->mFlags);
+ if (status == NO_ERROR) {
+ desc->mSamplingRate = config.sample_rate;
+ desc->mChannelMask = config.channel_mask;
+ desc->mFormat = config.format;
+
+ // Here is where the out_set_parameters() for card & device gets called
+ if (!address.isEmpty()) {
+ char *param = audio_device_address_to_parameter(device, address);
+ mpClientInterface->setParameters(output, String8(param));
+ free(param);
+ }
+
+ // Here is where we step through and resolve any "dynamic" fields
+ String8 reply;
+ char *value;
+ if (profile->mSamplingRates[0] == 0) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
+ ALOGV("checkOutputsForDevice() supported sampling rates %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadSamplingRates(value + 1);
+ }
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
+ ALOGV("checkOutputsForDevice() supported formats %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadFormats(value + 1);
+ }
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
+ ALOGV("checkOutputsForDevice() supported channel masks %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadOutChannels(value + 1);
+ }
+ }
+ if (((profile->mSamplingRates[0] == 0) &&
+ (profile->mSamplingRates.size() < 2)) ||
+ ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
+ (profile->mFormats.size() < 2)) ||
+ ((profile->mChannelMasks[0] == 0) &&
+ (profile->mChannelMasks.size() < 2))) {
+ ALOGW("checkOutputsForDevice() missing param");
+ mpClientInterface->closeOutput(output);
+ output = AUDIO_IO_HANDLE_NONE;
+ } else if (profile->mSamplingRates[0] == 0 || profile->mFormats[0] == 0 ||
+ profile->mChannelMasks[0] == 0) {
+ mpClientInterface->closeOutput(output);
+ config.sample_rate = profile->pickSamplingRate();
+ config.channel_mask = profile->pickChannelMask();
+ config.format = profile->pickFormat();
+ config.offload_info.sample_rate = config.sample_rate;
+ config.offload_info.channel_mask = config.channel_mask;
+ config.offload_info.format = config.format;
+ status = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &output,
+ &config,
+ &desc->mDevice,
+ address,
+ &desc->mLatency,
+ desc->mFlags);
+ if (status == NO_ERROR) {
+ desc->mSamplingRate = config.sample_rate;
+ desc->mChannelMask = config.channel_mask;
+ desc->mFormat = config.format;
+ } else {
+ output = AUDIO_IO_HANDLE_NONE;
+ }
+ }
+
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ addOutput(output, desc);
+ if (device_distinguishes_on_address(device) && address != "0") {
+ sp<AudioPolicyMix> policyMix;
+ if (mPolicyMixes.getAudioPolicyMix(address, policyMix) != NO_ERROR) {
+ ALOGE("checkOutputsForDevice() cannot find policy for address %s",
+ address.string());
+ }
+ policyMix->setOutput(desc);
+ desc->mPolicyMix = &(policyMix->getMix());
+
+ } else if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) {
+ // no duplicated output for direct outputs and
+ // outputs used by dynamic policy mixes
+ audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE;
+
+ // set initial stream volume for device
+ applyStreamVolumes(output, device, 0, true);
+
+ //TODO: configure audio effect output stage here
+
+ // open a duplicating output thread for the new output and the primary output
+ duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
+ mPrimaryOutput);
+ if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) {
+ // add duplicated output descriptor
+ sp<AudioOutputDescriptor> dupOutputDesc =
+ new AudioOutputDescriptor(NULL);
+ dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
+ dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
+ dupOutputDesc->mSamplingRate = desc->mSamplingRate;
+ dupOutputDesc->mFormat = desc->mFormat;
+ dupOutputDesc->mChannelMask = desc->mChannelMask;
+ dupOutputDesc->mLatency = desc->mLatency;
+ addOutput(duplicatedOutput, dupOutputDesc);
+ applyStreamVolumes(duplicatedOutput, device, 0, true);
+ } else {
+ ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
+ mPrimaryOutput, output);
+ mpClientInterface->closeOutput(output);
+ removeOutput(output);
+ nextAudioPortGeneration();
+ output = AUDIO_IO_HANDLE_NONE;
+ }
+ }
+ }
+ } else {
+ output = AUDIO_IO_HANDLE_NONE;
+ }
+ if (output == AUDIO_IO_HANDLE_NONE) {
+ ALOGW("checkOutputsForDevice() could not open output for device %x", device);
+ profiles.removeAt(profile_index);
+ profile_index--;
+ } else {
+ outputs.add(output);
+ devDesc->importAudioPort(profile);
+
+ if (device_distinguishes_on_address(device)) {
+ ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)",
+ device, address.string());
+ setOutputDevice(output, device, true/*force*/, 0/*delay*/,
+ NULL/*patch handle*/, address.string());
+ }
+ ALOGV("checkOutputsForDevice(): adding output %d", output);
+ }
+ }
+
+ if (profiles.isEmpty()) {
+ ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+ return BAD_VALUE;
+ }
+ } else { // Disconnect
+ // check if one opened output is not needed any more after disconnecting one device
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated()) {
+ // exact match on device
+ if (device_distinguishes_on_address(device) &&
+ (desc->mProfile->mSupportedDevices.types() == device)) {
+ findIoHandlesByAddress(desc, device, address, outputs);
+ } else if (!(desc->mProfile->mSupportedDevices.types()
+ & mAvailableOutputDevices.types())) {
+ ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
+ mOutputs.keyAt(i));
+ outputs.add(mOutputs.keyAt(i));
+ }
+ }
+ }
+ // Clear any profiles associated with the disconnected device.
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ if (profile->mSupportedDevices.types() & device) {
+ ALOGV("checkOutputsForDevice(): "
+ "clearing direct output profile %zu on module %zu", j, i);
+ if (profile->mSamplingRates[0] == 0) {
+ profile->mSamplingRates.clear();
+ profile->mSamplingRates.add(0);
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ profile->mFormats.clear();
+ profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ profile->mChannelMasks.clear();
+ profile->mChannelMasks.add(0);
+ }
+ }
+ }
+ }
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& inputs,
+ const String8 address)
+{
+ sp<AudioInputDescriptor> desc;
+ if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+ // first list already open inputs that can be routed to this device
+ for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
+ desc = mInputs.valueAt(input_index);
+ if (desc->mProfile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) {
+ ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index));
+ inputs.add(mInputs.keyAt(input_index));
+ }
+ }
+
+ // then look for input profiles that can be routed to this device
+ SortedVector< sp<IOProfile> > profiles;
+ for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++)
+ {
+ if (mHwModules[module_idx]->mHandle == 0) {
+ continue;
+ }
+ for (size_t profile_index = 0;
+ profile_index < mHwModules[module_idx]->mInputProfiles.size();
+ profile_index++)
+ {
+ sp<IOProfile> profile = mHwModules[module_idx]->mInputProfiles[profile_index];
+
+ if (profile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) {
+ if (!device_distinguishes_on_address(device) ||
+ address == profile->mSupportedDevices[0]->mAddress) {
+ profiles.add(profile);
+ ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
+ profile_index, module_idx);
+ }
+ }
+ }
+ }
+
+ if (profiles.isEmpty() && inputs.isEmpty()) {
+ ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
+ return BAD_VALUE;
+ }
+
+ // open inputs for matching profiles if needed. Direct inputs are also opened to
+ // query for dynamic parameters and will be closed later by setDeviceConnectionState()
+ for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
+
+ sp<IOProfile> profile = profiles[profile_index];
+ // nothing to do if one input is already opened for this profile
+ size_t input_index;
+ for (input_index = 0; input_index < mInputs.size(); input_index++) {
+ desc = mInputs.valueAt(input_index);
+ if (desc->mProfile == profile) {
+ break;
+ }
+ }
+ if (input_index != mInputs.size()) {
+ continue;
+ }
+
+ ALOGV("opening input for device 0x%X with params %s", device, address.string());
+ desc = new AudioInputDescriptor(profile);
+ desc->mDevice = device;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = desc->mSamplingRate;
+ config.channel_mask = desc->mChannelMask;
+ config.format = desc->mFormat;
+ audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
+ status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
+ &input,
+ &config,
+ &desc->mDevice,
+ address,
+ AUDIO_SOURCE_MIC,
+ AUDIO_INPUT_FLAG_NONE /*FIXME*/);
+
+ if (status == NO_ERROR) {
+ desc->mSamplingRate = config.sample_rate;
+ desc->mChannelMask = config.channel_mask;
+ desc->mFormat = config.format;
+
+ if (!address.isEmpty()) {
+ char *param = audio_device_address_to_parameter(device, address);
+ mpClientInterface->setParameters(input, String8(param));
+ free(param);
+ }
+
+ // Here is where we step through and resolve any "dynamic" fields
+ String8 reply;
+ char *value;
+ if (profile->mSamplingRates[0] == 0) {
+ reply = mpClientInterface->getParameters(input,
+ String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
+ ALOGV("checkInputsForDevice() direct input sup sampling rates %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadSamplingRates(value + 1);
+ }
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ reply = mpClientInterface->getParameters(input,
+ String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
+ ALOGV("checkInputsForDevice() direct input sup formats %s", reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadFormats(value + 1);
+ }
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ reply = mpClientInterface->getParameters(input,
+ String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
+ ALOGV("checkInputsForDevice() direct input sup channel masks %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ profile->loadInChannels(value + 1);
+ }
+ }
+ if (((profile->mSamplingRates[0] == 0) && (profile->mSamplingRates.size() < 2)) ||
+ ((profile->mFormats[0] == 0) && (profile->mFormats.size() < 2)) ||
+ ((profile->mChannelMasks[0] == 0) && (profile->mChannelMasks.size() < 2))) {
+ ALOGW("checkInputsForDevice() direct input missing param");
+ mpClientInterface->closeInput(input);
+ input = AUDIO_IO_HANDLE_NONE;
+ }
+
+ if (input != 0) {
+ addInput(input, desc);
+ }
+ } // endif input != 0
+
+ if (input == AUDIO_IO_HANDLE_NONE) {
+ ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
+ profiles.removeAt(profile_index);
+ profile_index--;
+ } else {
+ inputs.add(input);
+ ALOGV("checkInputsForDevice(): adding input %d", input);
+ }
+ } // end scan profiles
+
+ if (profiles.isEmpty()) {
+ ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
+ return BAD_VALUE;
+ }
+ } else {
+ // Disconnect
+ // check if one opened input is not needed any more after disconnecting one device
+ for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
+ desc = mInputs.valueAt(input_index);
+ if (!(desc->mProfile->mSupportedDevices.types() & mAvailableInputDevices.types() &
+ ~AUDIO_DEVICE_BIT_IN)) {
+ ALOGV("checkInputsForDevice(): disconnecting adding input %d",
+ mInputs.keyAt(input_index));
+ inputs.add(mInputs.keyAt(input_index));
+ }
+ }
+ // Clear any profiles associated with the disconnected device.
+ for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) {
+ if (mHwModules[module_index]->mHandle == 0) {
+ continue;
+ }
+ for (size_t profile_index = 0;
+ profile_index < mHwModules[module_index]->mInputProfiles.size();
+ profile_index++) {
+ sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index];
+ if (profile->mSupportedDevices.types() & device & ~AUDIO_DEVICE_BIT_IN) {
+ ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu",
+ profile_index, module_index);
+ if (profile->mSamplingRates[0] == 0) {
+ profile->mSamplingRates.clear();
+ profile->mSamplingRates.add(0);
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ profile->mFormats.clear();
+ profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ profile->mChannelMasks.clear();
+ profile->mChannelMasks.add(0);
+ }
+ }
+ }
+ }
+ } // end disconnect
+
+ return NO_ERROR;
+}
+
+
+void AudioPolicyManager::closeOutput(audio_io_handle_t output)
+{
+ ALOGV("closeOutput(%d)", output);
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ if (outputDesc == NULL) {
+ ALOGW("closeOutput() unknown output %d", output);
+ return;
+ }
+ mPolicyMixes.closeOutput(outputDesc);
+
+ // look for duplicated outputs connected to the output being removed.
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i);
+ if (dupOutputDesc->isDuplicated() &&
+ (dupOutputDesc->mOutput1 == outputDesc ||
+ dupOutputDesc->mOutput2 == outputDesc)) {
+ sp<AudioOutputDescriptor> outputDesc2;
+ if (dupOutputDesc->mOutput1 == outputDesc) {
+ outputDesc2 = dupOutputDesc->mOutput2;
+ } else {
+ outputDesc2 = dupOutputDesc->mOutput1;
+ }
+ // As all active tracks on duplicated output will be deleted,
+ // and as they were also referenced on the other output, the reference
+ // count for their stream type must be adjusted accordingly on
+ // the other output.
+ for (int j = 0; j < AUDIO_STREAM_CNT; j++) {
+ int refCount = dupOutputDesc->mRefCount[j];
+ outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount);
+ }
+ audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
+ ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
+
+ mpClientInterface->closeOutput(duplicatedOutput);
+ removeOutput(duplicatedOutput);
+ }
+ }
+
+ nextAudioPortGeneration();
+
+ ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ mAudioPatches.removeItemsAt(index);
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+
+ AudioParameter param;
+ param.add(String8("closing"), String8("true"));
+ mpClientInterface->setParameters(output, param.toString());
+
+ mpClientInterface->closeOutput(output);
+ removeOutput(output);
+ mPreviousOutputs = mOutputs;
+}
+
+void AudioPolicyManager::closeInput(audio_io_handle_t input)
+{
+ ALOGV("closeInput(%d)", input);
+
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+ if (inputDesc == NULL) {
+ ALOGW("closeInput() unknown input %d", input);
+ return;
+ }
+
+ nextAudioPortGeneration();
+
+ ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ mAudioPatches.removeItemsAt(index);
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+
+ mpClientInterface->closeInput(input);
+ mInputs.removeItem(input);
+}
+
+SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device,
+ AudioOutputCollection openOutputs)
+{
+ SortedVector<audio_io_handle_t> outputs;
+
+ ALOGVV("getOutputsForDevice() device %04x", device);
+ for (size_t i = 0; i < openOutputs.size(); i++) {
+ ALOGVV("output %d isDuplicated=%d device=%04x",
+ i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
+ if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
+ ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
+ outputs.add(openOutputs.keyAt(i));
+ }
+ }
+ return outputs;
+}
+
+bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+ SortedVector<audio_io_handle_t>& outputs2)
+{
+ if (outputs1.size() != outputs2.size()) {
+ return false;
+ }
+ for (size_t i = 0; i < outputs1.size(); i++) {
+ if (outputs1[i] != outputs2[i]) {
+ return false;
+ }
+ }
+ return true;
+}
+
+void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
+{
+ audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
+ audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
+ SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
+
+ // also take into account external policy-related changes: add all outputs which are
+ // associated with policies in the "before" and "after" output vectors
+ ALOGVV("checkOutputForStrategy(): policy related outputs");
+ for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
+ const sp<AudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
+ if (desc != 0 && desc->mPolicyMix != NULL) {
+ srcOutputs.add(desc->mIoHandle);
+ ALOGVV(" previous outputs: adding %d", desc->mIoHandle);
+ }
+ }
+ for (size_t i = 0 ; i < mOutputs.size() ; i++) {
+ const sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ if (desc != 0 && desc->mPolicyMix != NULL) {
+ dstOutputs.add(desc->mIoHandle);
+ ALOGVV(" new outputs: adding %d", desc->mIoHandle);
+ }
+ }
+
+ if (!vectorsEqual(srcOutputs,dstOutputs)) {
+ ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
+ strategy, srcOutputs[0], dstOutputs[0]);
+ // mute strategy while moving tracks from one output to another
+ for (size_t i = 0; i < srcOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]);
+ if (isStrategyActive(desc, strategy)) {
+ setStrategyMute(strategy, true, srcOutputs[i]);
+ setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
+ }
+ }
+
+ // Move effects associated to this strategy from previous output to new output
+ if (strategy == STRATEGY_MEDIA) {
+ audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
+ SortedVector<audio_io_handle_t> moved;
+ for (size_t i = 0; i < mEffects.size(); i++) {
+ sp<EffectDescriptor> effectDesc = mEffects.valueAt(i);
+ if (effectDesc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
+ effectDesc->mIo != fxOutput) {
+ if (moved.indexOf(effectDesc->mIo) < 0) {
+ ALOGV("checkOutputForStrategy() moving effect %d to output %d",
+ mEffects.keyAt(i), fxOutput);
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, effectDesc->mIo,
+ fxOutput);
+ moved.add(effectDesc->mIo);
+ }
+ effectDesc->mIo = fxOutput;
+ }
+ }
+ }
+ // Move tracks associated to this strategy from previous output to new output
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ if (i == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (getStrategy((audio_stream_type_t)i) == strategy) {
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ }
+}
+
+void AudioPolicyManager::checkOutputForAllStrategies()
+{
+ if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
+ checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
+ checkOutputForStrategy(STRATEGY_PHONE);
+ if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
+ checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
+ checkOutputForStrategy(STRATEGY_SONIFICATION);
+ checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+ checkOutputForStrategy(STRATEGY_ACCESSIBILITY);
+ checkOutputForStrategy(STRATEGY_MEDIA);
+ checkOutputForStrategy(STRATEGY_DTMF);
+ checkOutputForStrategy(STRATEGY_REROUTING);
+}
+
+void AudioPolicyManager::checkA2dpSuspend()
+{
+ audio_io_handle_t a2dpOutput = mOutputs.getA2dpOutput();
+ if (a2dpOutput == 0) {
+ mA2dpSuspended = false;
+ return;
+ }
+
+ bool isScoConnected =
+ ((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET &
+ ~AUDIO_DEVICE_BIT_IN) != 0) ||
+ ((mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_ALL_SCO) != 0);
+ // suspend A2DP output if:
+ // (NOT already suspended) &&
+ // ((SCO device is connected &&
+ // (forced usage for communication || for record is SCO))) ||
+ // (phone state is ringing || in call)
+ //
+ // restore A2DP output if:
+ // (Already suspended) &&
+ // ((SCO device is NOT connected ||
+ // (forced usage NOT for communication && NOT for record is SCO))) &&
+ // (phone state is NOT ringing && NOT in call)
+ //
+ if (mA2dpSuspended) {
+ if ((!isScoConnected ||
+ ((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) != AUDIO_POLICY_FORCE_BT_SCO) &&
+ (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) != AUDIO_POLICY_FORCE_BT_SCO))) &&
+ ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) &&
+ (mEngine->getPhoneState() != AUDIO_MODE_RINGTONE))) {
+
+ mpClientInterface->restoreOutput(a2dpOutput);
+ mA2dpSuspended = false;
+ }
+ } else {
+ if ((isScoConnected &&
+ ((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) == AUDIO_POLICY_FORCE_BT_SCO) ||
+ (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO))) ||
+ ((mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) ||
+ (mEngine->getPhoneState() == AUDIO_MODE_RINGTONE))) {
+
+ mpClientInterface->suspendOutput(a2dpOutput);
+ mA2dpSuspended = true;
+ }
+ }
+}
+
+audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache)
+{
+ audio_devices_t device = AUDIO_DEVICE_NONE;
+
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+
+ ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ if (patchDesc->mUid != mUidCached) {
+ ALOGV("getNewOutputDevice() device %08x forced by patch %d",
+ outputDesc->device(), outputDesc->mPatchHandle);
+ return outputDesc->device();
+ }
+ }
+
+ // check the following by order of priority to request a routing change if necessary:
+ // 1: the strategy enforced audible is active and enforced on the output:
+ // use device for strategy enforced audible
+ // 2: we are in call or the strategy phone is active on the output:
+ // use device for strategy phone
+ // 3: the strategy for enforced audible is active but not enforced on the output:
+ // use the device for strategy enforced audible
+ // 4: the strategy sonification is active on the output:
+ // use device for strategy sonification
+ // 5: the strategy "respectful" sonification is active on the output:
+ // use device for strategy "respectful" sonification
+ // 6: the strategy accessibility is active on the output:
+ // use device for strategy accessibility
+ // 7: the strategy media is active on the output:
+ // use device for strategy media
+ // 8: the strategy DTMF is active on the output:
+ // use device for strategy DTMF
+ // 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output:
+ // use device for strategy t-t-s
+ if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE) &&
+ mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+ device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ } else if (isInCall() ||
+ isStrategyActive(outputDesc, STRATEGY_PHONE)) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE)) {
+ device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) {
+ device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_MEDIA)) {
+ device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_DTMF)) {
+ device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) {
+ device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
+ } else if (isStrategyActive(outputDesc, STRATEGY_REROUTING)) {
+ device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache);
+ }
+
+ ALOGV("getNewOutputDevice() selected device %x", device);
+ return device;
+}
+
+audio_devices_t AudioPolicyManager::getNewInputDevice(audio_io_handle_t input)
+{
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+
+ ssize_t index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ if (index >= 0) {
+ sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ if (patchDesc->mUid != mUidCached) {
+ ALOGV("getNewInputDevice() device %08x forced by patch %d",
+ inputDesc->mDevice, inputDesc->mPatchHandle);
+ return inputDesc->mDevice;
+ }
+ }
+
+ audio_devices_t device = getDeviceAndMixForInputSource(inputDesc->mInputSource);
+
+ ALOGV("getNewInputDevice() selected device %x", device);
+ return device;
+}
+
+uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) {
+ return (uint32_t)getStrategy(stream);
+}
+
+audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
+ // By checking the range of stream before calling getStrategy, we avoid
+ // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE
+ // and then return STRATEGY_MEDIA, but we want to return the empty set.
+ if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) {
+ return AUDIO_DEVICE_NONE;
+ }
+ audio_devices_t devices;
+ routing_strategy strategy = getStrategy(stream);
+ devices = getDeviceForStrategy(strategy, true /*fromCache*/);
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(devices, mOutputs);
+ for (size_t i = 0; i < outputs.size(); i++) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]);
+ if (isStrategyActive(outputDesc, strategy)) {
+ devices = outputDesc->device();
+ break;
+ }
+ }
+
+ /*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it
+ and doesn't really need to.*/
+ if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
+ devices |= AUDIO_DEVICE_OUT_SPEAKER;
+ devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ }
+
+ return devices;
+}
+
+routing_strategy AudioPolicyManager::getStrategy(audio_stream_type_t stream) const
+{
+ ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH");
+ return mEngine->getStrategyForStream(stream);
+}
+
+uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) {
+ // flags to strategy mapping
+ if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) {
+ return (uint32_t) STRATEGY_TRANSMITTED_THROUGH_SPEAKER;
+ }
+ if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+ return (uint32_t) STRATEGY_ENFORCED_AUDIBLE;
+ }
+ // usage to strategy mapping
+ return static_cast<uint32_t>(mEngine->getStrategyForUsage(attr->usage));
+}
+
+void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
+ switch(stream) {
+ case AUDIO_STREAM_MUSIC:
+ checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+ updateDevicesAndOutputs();
+ break;
+ default:
+ break;
+ }
+}
+
+uint32_t AudioPolicyManager::handleEventForBeacon(int event) {
+ switch(event) {
+ case STARTING_OUTPUT:
+ mBeaconMuteRefCount++;
+ break;
+ case STOPPING_OUTPUT:
+ if (mBeaconMuteRefCount > 0) {
+ mBeaconMuteRefCount--;
+ }
+ break;
+ case STARTING_BEACON:
+ mBeaconPlayingRefCount++;
+ break;
+ case STOPPING_BEACON:
+ if (mBeaconPlayingRefCount > 0) {
+ mBeaconPlayingRefCount--;
+ }
+ break;
+ }
+
+ if (mBeaconMuteRefCount > 0) {
+ // any playback causes beacon to be muted
+ return setBeaconMute(true);
+ } else {
+ // no other playback: unmute when beacon starts playing, mute when it stops
+ return setBeaconMute(mBeaconPlayingRefCount == 0);
+ }
+}
+
+uint32_t AudioPolicyManager::setBeaconMute(bool mute) {
+ ALOGV("setBeaconMute(%d) mBeaconMuteRefCount=%d mBeaconPlayingRefCount=%d",
+ mute, mBeaconMuteRefCount, mBeaconPlayingRefCount);
+ // keep track of muted state to avoid repeating mute/unmute operations
+ if (mBeaconMuted != mute) {
+ // mute/unmute AUDIO_STREAM_TTS on all outputs
+ ALOGV("\t muting %d", mute);
+ uint32_t maxLatency = 0;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ setStreamMute(AUDIO_STREAM_TTS, mute/*on*/,
+ desc->mIoHandle,
+ 0 /*delay*/, AUDIO_DEVICE_NONE);
+ const uint32_t latency = desc->latency() * 2;
+ if (latency > maxLatency) {
+ maxLatency = latency;
+ }
+ }
+ mBeaconMuted = mute;
+ return maxLatency;
+ }
+ return 0;
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache)
+{
+ if (fromCache) {
+ ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
+ strategy, mDeviceForStrategy[strategy]);
+ return mDeviceForStrategy[strategy];
+ }
+ return mEngine->getDeviceForStrategy(strategy);
+}
+
+void AudioPolicyManager::updateDevicesAndOutputs()
+{
+ for (int i = 0; i < NUM_STRATEGIES; i++) {
+ mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+ }
+ mPreviousOutputs = mOutputs;
+}
+
+uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
+ audio_devices_t prevDevice,
+ uint32_t delayMs)
+{
+ // mute/unmute strategies using an incompatible device combination
+ // if muting, wait for the audio in pcm buffer to be drained before proceeding
+ // if unmuting, unmute only after the specified delay
+ if (outputDesc->isDuplicated()) {
+ return 0;
+ }
+
+ uint32_t muteWaitMs = 0;
+ audio_devices_t device = outputDesc->device();
+ bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2);
+
+ for (size_t i = 0; i < NUM_STRATEGIES; i++) {
+ audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+ curDevice = curDevice & outputDesc->mProfile->mSupportedDevices.types();
+ bool mute = shouldMute && (curDevice & device) && (curDevice != device);
+ bool doMute = false;
+
+ if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
+ doMute = true;
+ outputDesc->mStrategyMutedByDevice[i] = true;
+ } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
+ doMute = true;
+ outputDesc->mStrategyMutedByDevice[i] = false;
+ }
+ if (doMute) {
+ for (size_t j = 0; j < mOutputs.size(); j++) {
+ sp<AudioOutputDescriptor> desc = mOutputs.valueAt(j);
+ // skip output if it does not share any device with current output
+ if ((desc->supportedDevices() & outputDesc->supportedDevices())
+ == AUDIO_DEVICE_NONE) {
+ continue;
+ }
+ audio_io_handle_t curOutput = mOutputs.keyAt(j);
+ ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
+ mute ? "muting" : "unmuting", i, curDevice, curOutput);
+ setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
+ if (isStrategyActive(desc, (routing_strategy)i)) {
+ if (mute) {
+ // FIXME: should not need to double latency if volume could be applied
+ // immediately by the audioflinger mixer. We must account for the delay
+ // between now and the next time the audioflinger thread for this output
+ // will process a buffer (which corresponds to one buffer size,
+ // usually 1/2 or 1/4 of the latency).
+ if (muteWaitMs < desc->latency() * 2) {
+ muteWaitMs = desc->latency() * 2;
+ }
+ }
+ }
+ }
+ }
+ }
+
+ // temporary mute output if device selection changes to avoid volume bursts due to
+ // different per device volumes
+ if (outputDesc->isActive() && (device != prevDevice)) {
+ if (muteWaitMs < outputDesc->latency() * 2) {
+ muteWaitMs = outputDesc->latency() * 2;
+ }
+ for (size_t i = 0; i < NUM_STRATEGIES; i++) {
+ if (isStrategyActive(outputDesc, (routing_strategy)i)) {
+ setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle);
+ // do tempMute unmute after twice the mute wait time
+ setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle,
+ muteWaitMs *2, device);
+ }
+ }
+ }
+
+ // wait for the PCM output buffers to empty before proceeding with the rest of the command
+ if (muteWaitMs > delayMs) {
+ muteWaitMs -= delayMs;
+ usleep(muteWaitMs * 1000);
+ return muteWaitMs;
+ }
+ return 0;
+}
+
+uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output,
+ audio_devices_t device,
+ bool force,
+ int delayMs,
+ audio_patch_handle_t *patchHandle,
+ const char* address)
+{
+ ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ AudioParameter param;
+ uint32_t muteWaitMs;
+
+ if (outputDesc->isDuplicated()) {
+ muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs);
+ muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs);
+ return muteWaitMs;
+ }
+ // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
+ // output profile
+ if ((device != AUDIO_DEVICE_NONE) &&
+ ((device & outputDesc->mProfile->mSupportedDevices.types()) == 0)) {
+ return 0;
+ }
+
+ // filter devices according to output selected
+ device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types());
+
+ audio_devices_t prevDevice = outputDesc->mDevice;
+
+ ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
+
+ if (device != AUDIO_DEVICE_NONE) {
+ outputDesc->mDevice = device;
+ }
+ muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
+
+ // Do not change the routing if:
+ // the requested device is AUDIO_DEVICE_NONE
+ // OR the requested device is the same as current device
+ // AND force is not specified
+ // AND the output is connected by a valid audio patch.
+ // Doing this check here allows the caller to call setOutputDevice() without conditions
+ if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force &&
+ outputDesc->mPatchHandle != 0) {
+ ALOGV("setOutputDevice() setting same device %04x or null device for output %d",
+ device, output);
+ return muteWaitMs;
+ }
+
+ ALOGV("setOutputDevice() changing device");
+
+ // do the routing
+ if (device == AUDIO_DEVICE_NONE) {
+ resetOutputDevice(output, delayMs, NULL);
+ } else {
+ DeviceVector deviceList = (address == NULL) ?
+ mAvailableOutputDevices.getDevicesFromType(device)
+ : mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address));
+ if (!deviceList.isEmpty()) {
+ struct audio_patch patch;
+ outputDesc->toAudioPortConfig(&patch.sources[0]);
+ patch.num_sources = 1;
+ patch.num_sinks = 0;
+ for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) {
+ deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]);
+ patch.num_sinks++;
+ }
+ ssize_t index;
+ if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ }
+ sp< AudioPatch> patchDesc;
+ audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ if (index >= 0) {
+ patchDesc = mAudioPatches.valueAt(index);
+ afPatchHandle = patchDesc->mAfPatchHandle;
+ }
+
+ status_t status = mpClientInterface->createAudioPatch(&patch,
+ &afPatchHandle,
+ delayMs);
+ ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d"
+ "num_sources %d num_sinks %d",
+ status, afPatchHandle, patch.num_sources, patch.num_sinks);
+ if (status == NO_ERROR) {
+ if (index < 0) {
+ patchDesc = new AudioPatch(&patch, mUidCached);
+ addAudioPatch(patchDesc->mHandle, patchDesc);
+ } else {
+ patchDesc->mPatch = patch;
+ }
+ patchDesc->mAfPatchHandle = afPatchHandle;
+ patchDesc->mUid = mUidCached;
+ if (patchHandle) {
+ *patchHandle = patchDesc->mHandle;
+ }
+ outputDesc->mPatchHandle = patchDesc->mHandle;
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+ }
+
+ // inform all input as well
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
+ if (!is_virtual_input_device(inputDescriptor->mDevice)) {
+ AudioParameter inputCmd = AudioParameter();
+ ALOGV("%s: inform input %d of device:%d", __func__,
+ inputDescriptor->mIoHandle, device);
+ inputCmd.addInt(String8(AudioParameter::keyRouting),device);
+ mpClientInterface->setParameters(inputDescriptor->mIoHandle,
+ inputCmd.toString(),
+ delayMs);
+ }
+ }
+ }
+
+ // update stream volumes according to new device
+ applyStreamVolumes(output, device, delayMs);
+
+ return muteWaitMs;
+}
+
+status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output,
+ int delayMs,
+ audio_patch_handle_t *patchHandle)
+{
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ ssize_t index;
+ if (patchHandle) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle);
+ }
+ if (index < 0) {
+ return INVALID_OPERATION;
+ }
+ sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs);
+ ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status);
+ outputDesc->mPatchHandle = 0;
+ removeAudioPatch(patchDesc->mHandle);
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ return status;
+}
+
+status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input,
+ audio_devices_t device,
+ bool force,
+ audio_patch_handle_t *patchHandle)
+{
+ status_t status = NO_ERROR;
+
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+ if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
+ inputDesc->mDevice = device;
+
+ DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device);
+ if (!deviceList.isEmpty()) {
+ struct audio_patch patch;
+ inputDesc->toAudioPortConfig(&patch.sinks[0]);
+ // AUDIO_SOURCE_HOTWORD is for internal use only:
+ // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL
+ if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD &&
+ !inputDesc->mIsSoundTrigger) {
+ patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION;
+ }
+ patch.num_sinks = 1;
+ //only one input device for now
+ deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]);
+ patch.num_sources = 1;
+ ssize_t index;
+ if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ }
+ sp< AudioPatch> patchDesc;
+ audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ if (index >= 0) {
+ patchDesc = mAudioPatches.valueAt(index);
+ afPatchHandle = patchDesc->mAfPatchHandle;
+ }
+
+ status_t status = mpClientInterface->createAudioPatch(&patch,
+ &afPatchHandle,
+ 0);
+ ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d",
+ status, afPatchHandle);
+ if (status == NO_ERROR) {
+ if (index < 0) {
+ patchDesc = new AudioPatch(&patch, mUidCached);
+ addAudioPatch(patchDesc->mHandle, patchDesc);
+ } else {
+ patchDesc->mPatch = patch;
+ }
+ patchDesc->mAfPatchHandle = afPatchHandle;
+ patchDesc->mUid = mUidCached;
+ if (patchHandle) {
+ *patchHandle = patchDesc->mHandle;
+ }
+ inputDesc->mPatchHandle = patchDesc->mHandle;
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ }
+ }
+ }
+ return status;
+}
+
+status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input,
+ audio_patch_handle_t *patchHandle)
+{
+ sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
+ ssize_t index;
+ if (patchHandle) {
+ index = mAudioPatches.indexOfKey(*patchHandle);
+ } else {
+ index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
+ }
+ if (index < 0) {
+ return INVALID_OPERATION;
+ }
+ sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ ALOGV("resetInputDevice() releaseAudioPatch returned %d", status);
+ inputDesc->mPatchHandle = 0;
+ removeAudioPatch(patchDesc->mHandle);
+ nextAudioPortGeneration();
+ mpClientInterface->onAudioPatchListUpdate();
+ return status;
+}
+
+sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
+ String8 address,
+ uint32_t& samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags)
+{
+ // Choose an input profile based on the requested capture parameters: select the first available
+ // profile supporting all requested parameters.
+
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
+ {
+ sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j];
+ // profile->log();
+ if (profile->isCompatibleProfile(device, address, samplingRate,
+ &samplingRate /*updatedSamplingRate*/,
+ format, channelMask, (audio_output_flags_t) flags)) {
+
+ return profile;
+ }
+ }
+ }
+ return NULL;
+}
+
+
+audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource,
+ AudioMix **policyMix)
+{
+ audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
+ audio_devices_t selectedDeviceFromMix =
+ mPolicyMixes.getDeviceAndMixForInputSource(inputSource, availableDeviceTypes, policyMix);
+
+ if (selectedDeviceFromMix != AUDIO_DEVICE_NONE) {
+ return selectedDeviceFromMix;
+ }
+ return getDeviceForInputSource(inputSource);
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
+{
+ return mEngine->getDeviceForInputSource(inputSource);
+}
+
+float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
+ int index,
+ audio_io_handle_t output,
+ audio_devices_t device)
+{
+ float volume = 1.0;
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->device();
+ }
+ volume = mEngine->volIndexToAmpl(Volume::getDeviceCategory(device), stream, index);
+
+ // if a headset is connected, apply the following rules to ring tones and notifications
+ // to avoid sound level bursts in user's ears:
+ // - always attenuate ring tones and notifications volume by 6dB
+ // - if music is playing, always limit the volume to current music volume,
+ // with a minimum threshold at -36dB so that notification is always perceived.
+ const routing_strategy stream_strategy = getStrategy(stream);
+ if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+ AUDIO_DEVICE_OUT_WIRED_HEADSET |
+ AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
+ ((stream_strategy == STRATEGY_SONIFICATION)
+ || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
+ || (stream == AUDIO_STREAM_SYSTEM)
+ || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
+ (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) &&
+ mStreams.canBeMuted(stream)) {
+ volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
+ // when the phone is ringing we must consider that music could have been paused just before
+ // by the music application and behave as if music was active if the last music track was
+ // just stopped
+ if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
+ mLimitRingtoneVolume) {
+ audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
+ float musicVol = computeVolume(AUDIO_STREAM_MUSIC,
+ mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice),
+ output,
+ musicDevice);
+ float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
+ musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
+ if (volume > minVol) {
+ volume = minVol;
+ ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
+ }
+ }
+ }
+
+ return volume;
+}
+
+status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream,
+ int index,
+ audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs,
+ bool force)
+{
+
+ // do not change actual stream volume if the stream is muted
+ if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
+ ALOGVV("checkAndSetVolume() stream %d muted count %d",
+ stream, mOutputs.valueFor(output)->mMuteCount[stream]);
+ return NO_ERROR;
+ }
+ audio_policy_forced_cfg_t forceUseForComm =
+ mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION);
+ // do not change in call volume if bluetooth is connected and vice versa
+ if ((stream == AUDIO_STREAM_VOICE_CALL && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) ||
+ (stream == AUDIO_STREAM_BLUETOOTH_SCO && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO)) {
+ ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
+ stream, forceUseForComm);
+ return INVALID_OPERATION;
+ }
+
+ float volume = computeVolume(stream, index, output, device);
+ // unit gain if rerouting to external policy
+ if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index >= 0) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+ if (outputDesc->mPolicyMix != NULL) {
+ ALOGV("max gain when rerouting for output=%d", output);
+ volume = 1.0f;
+ }
+ }
+
+ }
+ // We actually change the volume if:
+ // - the float value returned by computeVolume() changed
+ // - the force flag is set
+ if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
+ force) {
+ mOutputs.valueFor(output)->mCurVolume[stream] = volume;
+ ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
+ // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
+ // enabled
+ if (stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+ mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs);
+ }
+ mpClientInterface->setStreamVolume(stream, volume, output, delayMs);
+ }
+
+ if (stream == AUDIO_STREAM_VOICE_CALL ||
+ stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+ float voiceVolume;
+ // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+ if (stream == AUDIO_STREAM_VOICE_CALL) {
+ voiceVolume = (float)index/(float)mStreams[stream].getVolumeIndexMax();
+ } else {
+ voiceVolume = 1.0;
+ }
+
+ if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
+ mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+ mLastVoiceVolume = voiceVolume;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs,
+ bool force)
+{
+ ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
+
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ checkAndSetVolume((audio_stream_type_t)stream,
+ mStreams[stream].getVolumeIndex(device),
+ output,
+ device,
+ delayMs,
+ force);
+ }
+}
+
+void AudioPolicyManager::setStrategyMute(routing_strategy strategy,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs,
+ audio_devices_t device)
+{
+ ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (stream == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (getStrategy((audio_stream_type_t)stream) == strategy) {
+ setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device);
+ }
+ }
+}
+
+void AudioPolicyManager::setStreamMute(audio_stream_type_t stream,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs,
+ audio_devices_t device)
+{
+ const StreamDescriptor &streamDesc = mStreams[stream];
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->device();
+ }
+
+ ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
+ stream, on, output, outputDesc->mMuteCount[stream], device);
+
+ if (on) {
+ if (outputDesc->mMuteCount[stream] == 0) {
+ if (streamDesc.canBeMuted() &&
+ ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) ||
+ (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) {
+ checkAndSetVolume(stream, 0, output, device, delayMs);
+ }
+ }
+ // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
+ outputDesc->mMuteCount[stream]++;
+ } else {
+ if (outputDesc->mMuteCount[stream] == 0) {
+ ALOGV("setStreamMute() unmuting non muted stream!");
+ return;
+ }
+ if (--outputDesc->mMuteCount[stream] == 0) {
+ checkAndSetVolume(stream,
+ streamDesc.getVolumeIndex(device),
+ output,
+ device,
+ delayMs);
+ }
+ }
+}
+
+void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream,
+ bool starting, bool stateChange)
+{
+ // if the stream pertains to sonification strategy and we are in call we must
+ // mute the stream if it is low visibility. If it is high visibility, we must play a tone
+ // in the device used for phone strategy and play the tone if the selected device does not
+ // interfere with the device used for phone strategy
+ // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
+ // many times as there are active tracks on the output
+ const routing_strategy stream_strategy = getStrategy(stream);
+ if ((stream_strategy == STRATEGY_SONIFICATION) ||
+ ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
+ stream, starting, outputDesc->mDevice, stateChange);
+ if (outputDesc->mRefCount[stream]) {
+ int muteCount = 1;
+ if (stateChange) {
+ muteCount = outputDesc->mRefCount[stream];
+ }
+ if (audio_is_low_visibility(stream)) {
+ ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, mPrimaryOutput);
+ }
+ } else {
+ ALOGV("handleIncallSonification() high visibility");
+ if (outputDesc->device() &
+ getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
+ ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, mPrimaryOutput);
+ }
+ }
+ if (starting) {
+ mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
+ AUDIO_STREAM_VOICE_CALL);
+ } else {
+ mpClientInterface->stopTone();
+ }
+ }
+ }
+ }
+}
+
+void AudioPolicyManager::defaultAudioPolicyConfig(void)
+{
+ sp<HwModule> module;
+ sp<IOProfile> profile;
+ sp<DeviceDescriptor> defaultInputDevice =
+ new DeviceDescriptor(String8("builtin-mic"), AUDIO_DEVICE_IN_BUILTIN_MIC);
+ mAvailableOutputDevices.add(mDefaultOutputDevice);
+ mAvailableInputDevices.add(defaultInputDevice);
+
+ module = new HwModule("primary");
+
+ profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE, module);
+ profile->mSamplingRates.add(44100);
+ profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+ profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
+ profile->mSupportedDevices.add(mDefaultOutputDevice);
+ profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
+ module->mOutputProfiles.add(profile);
+
+ profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK, module);
+ profile->mSamplingRates.add(8000);
+ profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+ profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
+ profile->mSupportedDevices.add(defaultInputDevice);
+ module->mInputProfiles.add(profile);
+
+ mHwModules.add(module);
+}
+
+audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr)
+{
+ // flags to stream type mapping
+ if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+ return AUDIO_STREAM_ENFORCED_AUDIBLE;
+ }
+ if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
+ return AUDIO_STREAM_BLUETOOTH_SCO;
+ }
+ if ((attr->flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) {
+ return AUDIO_STREAM_TTS;
+ }
+
+ // usage to stream type mapping
+ switch (attr->usage) {
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ return AUDIO_STREAM_MUSIC;
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ if (isStreamActive(AUDIO_STREAM_ALARM)) {
+ return AUDIO_STREAM_ALARM;
+ }
+ if (isStreamActive(AUDIO_STREAM_RING)) {
+ return AUDIO_STREAM_RING;
+ }
+ if (isInCall()) {
+ return AUDIO_STREAM_VOICE_CALL;
+ }
+ return AUDIO_STREAM_ACCESSIBILITY;
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ return AUDIO_STREAM_SYSTEM;
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ return AUDIO_STREAM_VOICE_CALL;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ return AUDIO_STREAM_DTMF;
+
+ case AUDIO_USAGE_ALARM:
+ return AUDIO_STREAM_ALARM;
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ return AUDIO_STREAM_RING;
+
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ return AUDIO_STREAM_NOTIFICATION;
+
+ case AUDIO_USAGE_UNKNOWN:
+ default:
+ return AUDIO_STREAM_MUSIC;
+ }
+}
+
+bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa)
+{
+ // has flags that map to a strategy?
+ if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) {
+ return true;
+ }
+
+ // has known usage?
+ switch (paa->usage) {
+ case AUDIO_USAGE_UNKNOWN:
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ case AUDIO_USAGE_ALARM:
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_VIRTUAL_SOURCE:
+ break;
+ default:
+ return false;
+ }
+ return true;
+}
+
+bool AudioPolicyManager::isStrategyActive(const sp<AudioOutputDescriptor> outputDesc,
+ routing_strategy strategy, uint32_t inPastMs,
+ nsecs_t sysTime) const
+{
+ if ((sysTime == 0) && (inPastMs != 0)) {
+ sysTime = systemTime();
+ }
+ for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+ if (i == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (((getStrategy((audio_stream_type_t)i) == strategy) ||
+ (NUM_STRATEGIES == strategy)) &&
+ outputDesc->isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage)
+{
+ return mEngine->getForceUse(usage);
+}
+
+bool AudioPolicyManager::isInCall()
+{
+ return isStateInCall(mEngine->getPhoneState());
+}
+
+bool AudioPolicyManager::isStateInCall(int state)
+{
+ return is_state_in_call(state);
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
new file mode 100644
index 0000000..02b678a
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -0,0 +1,578 @@
+/*
+ * 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.
+ */
+
+#pragma once
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <cutils/config_utils.h>
+#include <cutils/misc.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/SortedVector.h>
+#include <media/AudioPolicy.h>
+#include "AudioPolicyInterface.h"
+
+#include <AudioPolicyManagerInterface.h>
+#include <AudioPolicyManagerObserver.h>
+#include <AudioGain.h>
+#include <AudioPort.h>
+#include <AudioPatch.h>
+#include <ConfigParsingUtils.h>
+#include <DeviceDescriptor.h>
+#include <IOProfile.h>
+#include <HwModule.h>
+#include <AudioInputDescriptor.h>
+#include <AudioOutputDescriptor.h>
+#include <AudioPolicyMix.h>
+#include <EffectDescriptor.h>
+#include <SoundTriggerSession.h>
+#include <StreamDescriptor.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
+#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
+// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
+#define SONIFICATION_HEADSET_VOLUME_MIN 0.016
+// Time in milliseconds during which we consider that music is still active after a music
+// track was stopped - see computeVolume()
+#define SONIFICATION_HEADSET_MUSIC_DELAY 5000
+
+// Time in milliseconds during witch some streams are muted while the audio path
+// is switched
+#define MUTE_TIME_MS 2000
+
+#define NUM_TEST_OUTPUTS 5
+
+#define NUM_VOL_CURVE_KNEES 2
+
+// Default minimum length allowed for offloading a compressed track
+// Can be overridden by the audio.offload.min.duration.secs property
+#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager implements audio policy manager behavior common to all platforms.
+// ----------------------------------------------------------------------------
+
+class AudioPolicyManager : public AudioPolicyInterface, public AudioPolicyManagerObserver
+
+#ifdef AUDIO_POLICY_TEST
+ , public Thread
+#endif //AUDIO_POLICY_TEST
+{
+
+public:
+ AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+ virtual ~AudioPolicyManager();
+
+ // AudioPolicyInterface
+ virtual status_t setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name);
+ virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
+ const char *device_address);
+ virtual void setPhoneState(audio_mode_t state);
+ virtual void setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config);
+ virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+
+ virtual void setSystemProperty(const char* property, const char* value);
+ virtual status_t initCheck();
+ 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 status_t getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ 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 status_t startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ virtual status_t stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ virtual void releaseOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ virtual status_t getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ input_type_t *inputType);
+
+ // indicates to the audio policy manager that the input starts being used.
+ virtual status_t startInput(audio_io_handle_t input,
+ audio_session_t session);
+
+ // indicates to the audio policy manager that the input stops being used.
+ virtual status_t stopInput(audio_io_handle_t input,
+ audio_session_t session);
+ virtual void releaseInput(audio_io_handle_t input,
+ audio_session_t session);
+ virtual void closeAllInputs();
+ virtual void initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax);
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device);
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device);
+
+ // return the strategy corresponding to a given stream type
+ virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
+ // return the strategy corresponding to the given audio attributes
+ virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr);
+
+ // return the enabled output devices for the given stream type
+ virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
+
+ virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
+ virtual status_t registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id);
+ virtual status_t unregisterEffect(int id)
+ {
+ return mEffects.unregisterEffect(id);
+ }
+ virtual status_t setEffectEnabled(int id, bool enabled)
+ {
+ return mEffects.setEffectEnabled(id, enabled);
+ }
+
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const
+ {
+ return mOutputs.isStreamActive(stream, inPastMs);
+ }
+ // return whether a stream is playing remotely, override to change the definition of
+ // local/remote playback, used for instance by notification manager to not make
+ // media players lose audio focus when not playing locally
+ // For the base implementation, "remotely" means playing during screen mirroring which
+ // uses an output for playback with a non-empty, non "0" address.
+ virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const
+ {
+ return mOutputs.isStreamActiveRemotely(stream, inPastMs);
+ }
+ virtual bool isSourceActive(audio_source_t source) const;
+
+ virtual status_t dump(int fd);
+
+ virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+ virtual status_t listAudioPorts(audio_port_role_t role,
+ audio_port_type_t type,
+ unsigned int *num_ports,
+ struct audio_port *ports,
+ unsigned int *generation);
+ virtual status_t getAudioPort(struct audio_port *port);
+ virtual status_t createAudioPatch(const struct audio_patch *patch,
+ audio_patch_handle_t *handle,
+ uid_t uid);
+ virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
+ uid_t uid);
+ virtual status_t listAudioPatches(unsigned int *num_patches,
+ struct audio_patch *patches,
+ unsigned int *generation);
+ virtual status_t setAudioPortConfig(const struct audio_port_config *config);
+ virtual void clearAudioPatches(uid_t uid);
+
+ virtual status_t acquireSoundTriggerSession(audio_session_t *session,
+ audio_io_handle_t *ioHandle,
+ audio_devices_t *device);
+
+ virtual status_t releaseSoundTriggerSession(audio_session_t session)
+ {
+ return mSoundTriggerSessions.releaseSession(session);
+ }
+
+ virtual status_t registerPolicyMixes(Vector<AudioMix> mixes);
+ virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
+
+ // Audio policy configuration file parsing (audio_policy.conf)
+ // TODO candidates to be moved to ConfigParsingUtils
+ void defaultAudioPolicyConfig(void);
+
+ // return the strategy corresponding to a given stream type
+ routing_strategy getStrategy(audio_stream_type_t stream) const;
+
+ // From AudioPolicyManagerObserver
+ virtual const AudioPatchCollection &getAudioPatches() const
+ {
+ return mAudioPatches;
+ }
+ virtual const SoundTriggerSessionCollection &getSoundTriggerSessionCollection() const
+ {
+ return mSoundTriggerSessions;
+ }
+ virtual const AudioPolicyMixCollection &getAudioPolicyMixCollection() const
+ {
+ return mPolicyMixes;
+ }
+ virtual const AudioOutputCollection &getOutputs() const
+ {
+ return mOutputs;
+ }
+ virtual const AudioInputCollection &getInputs() const
+ {
+ return mInputs;
+ }
+ virtual const DeviceVector &getAvailableOutputDevices() const
+ {
+ return mAvailableOutputDevices;
+ }
+ virtual const DeviceVector &getAvailableInputDevices() const
+ {
+ return mAvailableInputDevices;
+ }
+ virtual StreamDescriptorCollection &getStreamDescriptors()
+ {
+ return mStreams;
+ }
+ virtual const sp<DeviceDescriptor> &getDefaultOutputDevice() const
+ {
+ return mDefaultOutputDevice;
+ }
+protected:
+ void addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc);
+ void removeOutput(audio_io_handle_t output);
+ void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc);
+
+ // return appropriate device for streams handled by the specified strategy according to current
+ // phone state, connected devices...
+ // if fromCache is true, the device is returned from mDeviceForStrategy[],
+ // otherwise it is determine by current state
+ // (device connected,phone state, force use, a2dp output...)
+ // This allows to:
+ // 1 speed up process when the state is stable (when starting or stopping an output)
+ // 2 access to either current device selection (fromCache == true) or
+ // "future" device selection (fromCache == false) when called from a context
+ // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
+ // before updateDevicesAndOutputs() is called.
+ virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache);
+
+ bool isStrategyActive(const sp<AudioOutputDescriptor> outputDesc, routing_strategy strategy,
+ uint32_t inPastMs = 0, nsecs_t sysTime = 0) const;
+
+ // change the route of the specified output. Returns the number of ms we have slept to
+ // allow new routing to take effect in certain cases.
+ virtual uint32_t setOutputDevice(audio_io_handle_t output,
+ audio_devices_t device,
+ bool force = false,
+ int delayMs = 0,
+ audio_patch_handle_t *patchHandle = NULL,
+ const char* address = NULL);
+ status_t resetOutputDevice(audio_io_handle_t output,
+ int delayMs = 0,
+ audio_patch_handle_t *patchHandle = NULL);
+ status_t setInputDevice(audio_io_handle_t input,
+ audio_devices_t device,
+ bool force = false,
+ audio_patch_handle_t *patchHandle = NULL);
+ status_t resetInputDevice(audio_io_handle_t input,
+ audio_patch_handle_t *patchHandle = NULL);
+
+ // select input device corresponding to requested audio source
+ virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
+
+ // compute the actual volume for a given stream according to the requested index and a particular
+ // device
+ virtual float computeVolume(audio_stream_type_t stream, int index,
+ audio_io_handle_t output, audio_devices_t device);
+
+ // check that volume change is permitted, compute and send new volume to audio hardware
+ virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index,
+ audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs = 0, bool force = false);
+
+ // apply all stream volumes to the specified output and device
+ void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+
+ // Mute or unmute all streams handled by the specified strategy on the specified output
+ void setStrategyMute(routing_strategy strategy,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs = 0,
+ audio_devices_t device = (audio_devices_t)0);
+
+ // Mute or unmute the stream on the specified output
+ void setStreamMute(audio_stream_type_t stream,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs = 0,
+ audio_devices_t device = (audio_devices_t)0);
+
+ // handle special cases for sonification strategy while in call: mute streams or replace by
+ // a special tone in the device used for communication
+ void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange);
+
+ audio_mode_t getPhoneState();
+
+ // true if device is in a telephony or VoIP call
+ virtual bool isInCall();
+ // true if given state represents a device in a telephony or VoIP call
+ virtual bool isStateInCall(int state);
+
+ // when a device is connected, checks if an open output can be routed
+ // to this device. If none is open, tries to open one of the available outputs.
+ // Returns an output suitable to this device or 0.
+ // when a device is disconnected, checks if an output is not used any more and
+ // returns its handle if any.
+ // transfers the audio tracks and effects from one output thread to another accordingly.
+ status_t checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& outputs,
+ const String8 address);
+
+ status_t checkInputsForDevice(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& inputs,
+ const String8 address);
+
+ // close an output and its companion duplicating output.
+ void closeOutput(audio_io_handle_t output);
+
+ // close an input.
+ void closeInput(audio_io_handle_t input);
+
+ // checks and if necessary changes outputs used for all strategies.
+ // must be called every time a condition that affects the output choice for a given strategy
+ // changes: connected device, phone state, force use...
+ // Must be called before updateDevicesAndOutputs()
+ void checkOutputForStrategy(routing_strategy strategy);
+
+ // Same as checkOutputForStrategy() but for a all strategies in order of priority
+ void checkOutputForAllStrategies();
+
+ // manages A2DP output suspend/restore according to phone state and BT SCO usage
+ void checkA2dpSuspend();
+
+ // selects the most appropriate device on output for current state
+ // must be called every time a condition that affects the device choice for a given output is
+ // changed: connected device, phone state, force use, output start, output stop..
+ // see getDeviceForStrategy() for the use of fromCache parameter
+ audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache);
+
+ // updates cache of device used by all strategies (mDeviceForStrategy[])
+ // must be called every time a condition that affects the device choice for a given strategy is
+ // changed: connected device, phone state, force use...
+ // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
+ // Must be called after checkOutputForAllStrategies()
+ void updateDevicesAndOutputs();
+
+ // selects the most appropriate device on input for current state
+ audio_devices_t getNewInputDevice(audio_io_handle_t input);
+
+ virtual uint32_t getMaxEffectsCpuLoad()
+ {
+ return mEffects.getMaxEffectsCpuLoad();
+ }
+
+ virtual uint32_t getMaxEffectsMemory()
+ {
+ return mEffects.getMaxEffectsMemory();
+ }
+#ifdef AUDIO_POLICY_TEST
+ virtual bool threadLoop();
+ void exit();
+ int testOutputIndex(audio_io_handle_t output);
+#endif //AUDIO_POLICY_TEST
+
+ SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
+ AudioOutputCollection openOutputs);
+ bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+ SortedVector<audio_io_handle_t>& outputs2);
+
+ // mute/unmute strategies using an incompatible device combination
+ // if muting, wait for the audio in pcm buffer to be drained before proceeding
+ // if unmuting, unmute only after the specified delay
+ // Returns the number of ms waited
+ virtual uint32_t checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
+ audio_devices_t prevDevice,
+ uint32_t delayMs);
+
+ audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+ audio_output_flags_t flags,
+ audio_format_t format);
+ // samplingRate parameter is an in/out and so may be modified
+ sp<IOProfile> getInputProfile(audio_devices_t device,
+ String8 address,
+ uint32_t& samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags);
+ sp<IOProfile> getProfileForDirectOutput(audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags);
+
+ audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
+
+ virtual status_t addAudioPatch(audio_patch_handle_t handle, const sp<AudioPatch>& patch)
+ {
+ return mAudioPatches.addAudioPatch(handle, patch);
+ }
+ virtual status_t removeAudioPatch(audio_patch_handle_t handle)
+ {
+ return mAudioPatches.removeAudioPatch(handle);
+ }
+
+ audio_devices_t availablePrimaryOutputDevices() const
+ {
+ return mOutputs.getSupportedDevices(mPrimaryOutput) & mAvailableOutputDevices.types();
+ }
+ audio_devices_t availablePrimaryInputDevices() const
+ {
+ return mAvailableInputDevices.getDevicesFromHwModule(
+ mOutputs.valueFor(mPrimaryOutput)->getModuleHandle());
+ }
+
+ void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0);
+
+ uid_t mUidCached;
+ AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
+ audio_io_handle_t mPrimaryOutput; // primary output handle
+ // list of descriptors for outputs currently opened
+ AudioOutputCollection mOutputs;
+ // copy of mOutputs before setDeviceConnectionState() opens new outputs
+ // reset to mOutputs when updateDevicesAndOutputs() is called.
+ AudioOutputCollection mPreviousOutputs;
+ AudioInputCollection mInputs; // list of input descriptors
+ DeviceVector mAvailableOutputDevices; // all available output devices
+ DeviceVector mAvailableInputDevices; // all available input devices
+
+ StreamDescriptorCollection mStreams; // stream descriptors for volume control
+ bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected
+ audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
+ float mLastVoiceVolume; // last voice volume value sent to audio HAL
+
+ EffectDescriptorCollection mEffects; // list of registered audio effects
+ bool mA2dpSuspended; // true if A2DP output is suspended
+ sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
+ bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
+ // to boost soft sounds, used to adjust volume curves accordingly
+
+ HwModuleCollection mHwModules;
+
+ volatile int32_t mAudioPortGeneration;
+
+ AudioPatchCollection mAudioPatches;
+
+ SoundTriggerSessionCollection mSoundTriggerSessions;
+
+ sp<AudioPatch> mCallTxPatch;
+ sp<AudioPatch> mCallRxPatch;
+
+ // for supporting "beacon" streams, i.e. streams that only play on speaker, and never
+ // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing
+ enum {
+ STARTING_OUTPUT,
+ STARTING_BEACON,
+ STOPPING_OUTPUT,
+ STOPPING_BEACON
+ };
+ uint32_t mBeaconMuteRefCount; // ref count for stream that would mute beacon
+ uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams
+ bool mBeaconMuted; // has STREAM_TTS been muted
+
+ AudioPolicyMixCollection mPolicyMixes; // list of registered mixes
+
+#ifdef AUDIO_POLICY_TEST
+ Mutex mLock;
+ Condition mWaitWorkCV;
+
+ int mCurOutput;
+ bool mDirectOutput;
+ audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
+ int mTestInput;
+ uint32_t mTestDevice;
+ uint32_t mTestSamplingRate;
+ uint32_t mTestFormat;
+ uint32_t mTestChannels;
+ uint32_t mTestLatencyMs;
+#endif //AUDIO_POLICY_TEST
+
+ uint32_t nextAudioPortGeneration();
+
+ // Audio Policy Engine Interface.
+ AudioPolicyManagerInterface *mEngine;
+private:
+ // updates device caching and output for streams that can influence the
+ // routing of notifications
+ void handleNotificationRoutingForStream(audio_stream_type_t stream);
+ // find the outputs on a given output descriptor that have the given address.
+ // to be called on an AudioOutputDescriptor whose supported devices (as defined
+ // in mProfile->mSupportedDevices) matches the device whose address is to be matched.
+ // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
+ // where addresses are used to distinguish between one connected device and another.
+ void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
+ const audio_devices_t device /*in*/,
+ const String8 address /*in*/,
+ SortedVector<audio_io_handle_t>& outputs /*out*/);
+ uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
+ // internal method to return the output handle for the given device and format
+ audio_io_handle_t getOutputForDevice(
+ audio_devices_t device,
+ audio_session_t session,
+ 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);
+ // internal function to derive a stream type value from audio attributes
+ audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
+ // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
+ // returns 0 if no mute/unmute event happened, the largest latency of the device where
+ // the mute/unmute happened
+ uint32_t handleEventForBeacon(int event);
+ uint32_t setBeaconMute(bool mute);
+ bool isValidAttributes(const audio_attributes_t *paa);
+
+ // select input device corresponding to requested audio source and return associated policy
+ // mix if any. Calls getDeviceForInputSource().
+ audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
+ AudioMix **policyMix = NULL);
+
+ // Called by setDeviceConnectionState().
+ status_t setDeviceConnectionStateInt(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name);
+};
+
+};
diff --git a/services/audiopolicy/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyClientImpl.cpp
rename to services/audiopolicy/service/AudioPolicyClientImpl.cpp
diff --git a/services/audiopolicy/AudioPolicyClientImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyClientImplLegacy.cpp
rename to services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp
diff --git a/services/audiopolicy/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyEffects.cpp
rename to services/audiopolicy/service/AudioPolicyEffects.cpp
diff --git a/services/audiopolicy/AudioPolicyEffects.h b/services/audiopolicy/service/AudioPolicyEffects.h
similarity index 100%
rename from services/audiopolicy/AudioPolicyEffects.h
rename to services/audiopolicy/service/AudioPolicyEffects.h
diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
similarity index 98%
rename from services/audiopolicy/AudioPolicyInterfaceImpl.cpp
rename to services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index a45dbb3..e9ff8389 100644
--- a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -28,7 +28,8 @@
status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address)
+ const char *device_address,
+ const char *device_name)
{
if (mAudioPolicyManager == NULL) {
return NO_INIT;
@@ -46,8 +47,8 @@
ALOGV("setDeviceConnectionState()");
Mutex::Autolock _l(mLock);
- return mAudioPolicyManager->setDeviceConnectionState(device,
- state, device_address);
+ return mAudioPolicyManager->setDeviceConnectionState(device, state,
+ device_address, device_name);
}
audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState(
diff --git a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
similarity index 99%
rename from services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp
rename to services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
index b8846c6..5a91192 100644
--- a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
@@ -33,7 +33,8 @@
status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address)
+ const char *device_address,
+ const char *device_name __unused)
{
if (mpAudioPolicy == NULL) {
return NO_INIT;
diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
similarity index 99%
rename from services/audiopolicy/AudioPolicyService.cpp
rename to services/audiopolicy/service/AudioPolicyService.cpp
index eb9116d..00f188f 100644
--- a/services/audiopolicy/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -532,7 +532,7 @@
mLock.unlock();
svc.clear();
mLock.lock();
- if (!exitPending() && mAudioCommands.isEmpty()) {
+ if (!exitPending() && (mAudioCommands.isEmpty() || waitTime != INT64_MAX)) {
// release delayed commands wake lock
release_wake_lock(mName.string());
ALOGV("AudioCommandThread() going to sleep");
diff --git a/services/audiopolicy/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
similarity index 99%
rename from services/audiopolicy/AudioPolicyService.h
rename to services/audiopolicy/service/AudioPolicyService.h
index 80284a4..0378384 100644
--- a/services/audiopolicy/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -35,7 +35,7 @@
#include <hardware_legacy/AudioPolicyInterface.h>
#endif
#include "AudioPolicyEffects.h"
-#include "AudioPolicyManager.h"
+#include "managerdefault/AudioPolicyManager.h"
namespace android {
@@ -61,7 +61,8 @@
virtual status_t setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address);
+ const char *device_address,
+ const char *device_name);
virtual audio_policy_dev_state_t getDeviceConnectionState(
audio_devices_t device,
const char *device_address);
diff --git a/services/camera/libcameraservice/Android.mk b/services/camera/libcameraservice/Android.mk
index e184d97..9c60911 100644
--- a/services/camera/libcameraservice/Android.mk
+++ b/services/camera/libcameraservice/Android.mk
@@ -23,8 +23,10 @@
LOCAL_SRC_FILES:= \
CameraService.cpp \
CameraDeviceFactory.cpp \
+ CameraFlashlight.cpp \
common/Camera2ClientBase.cpp \
common/CameraDeviceBase.cpp \
+ common/CameraModule.cpp \
common/FrameProcessorBase.cpp \
api1/CameraClient.cpp \
api1/Camera2Client.cpp \
@@ -40,7 +42,6 @@
api1/client2/CaptureSequencer.cpp \
api1/client2/ZslProcessor3.cpp \
api2/CameraDeviceClient.cpp \
- api_pro/ProCamera2Client.cpp \
device2/Camera2Device.cpp \
device3/Camera3Device.cpp \
device3/Camera3Stream.cpp \
@@ -52,6 +53,7 @@
device3/StatusTracker.cpp \
gui/RingBufferConsumer.cpp \
utils/CameraTraces.cpp \
+ utils/AutoConditionLock.cpp \
LOCAL_SHARED_LIBRARIES:= \
libui \
diff --git a/services/camera/libcameraservice/CameraDeviceFactory.cpp b/services/camera/libcameraservice/CameraDeviceFactory.cpp
index bfef50e..6589e27 100644
--- a/services/camera/libcameraservice/CameraDeviceFactory.cpp
+++ b/services/camera/libcameraservice/CameraDeviceFactory.cpp
@@ -48,6 +48,7 @@
case CAMERA_DEVICE_API_VERSION_3_0:
case CAMERA_DEVICE_API_VERSION_3_1:
case CAMERA_DEVICE_API_VERSION_3_2:
+ case CAMERA_DEVICE_API_VERSION_3_3:
device = new Camera3Device(cameraId);
break;
default:
diff --git a/services/camera/libcameraservice/CameraFlashlight.cpp b/services/camera/libcameraservice/CameraFlashlight.cpp
new file mode 100644
index 0000000..8613ac6
--- /dev/null
+++ b/services/camera/libcameraservice/CameraFlashlight.cpp
@@ -0,0 +1,886 @@
+/*
+ * Copyright (C) 2015 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 "CameraFlashlight"
+#define ATRACE_TAG ATRACE_TAG_CAMERA
+// #define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+#include <utils/Trace.h>
+#include <cutils/properties.h>
+
+#include "camera/CameraMetadata.h"
+#include "CameraFlashlight.h"
+#include "gui/IGraphicBufferConsumer.h"
+#include "gui/BufferQueue.h"
+#include "camera/camera2/CaptureRequest.h"
+#include "CameraDeviceFactory.h"
+
+
+namespace android {
+
+/////////////////////////////////////////////////////////////////////
+// CameraFlashlight implementation begins
+// used by camera service to control flashflight.
+/////////////////////////////////////////////////////////////////////
+CameraFlashlight::CameraFlashlight(CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks) :
+ mCameraModule(&cameraModule),
+ mCallbacks(&callbacks),
+ mFlashlightMapInitialized(false) {
+}
+
+CameraFlashlight::~CameraFlashlight() {
+}
+
+status_t CameraFlashlight::createFlashlightControl(const String8& cameraId) {
+ ALOGV("%s: creating a flash light control for camera %s", __FUNCTION__,
+ cameraId.string());
+ if (mFlashControl != NULL) {
+ return INVALID_OPERATION;
+ }
+
+ status_t res = OK;
+
+ if (mCameraModule->getModuleApiVersion() >= CAMERA_MODULE_API_VERSION_2_4) {
+ mFlashControl = new ModuleFlashControl(*mCameraModule, *mCallbacks);
+ if (mFlashControl == NULL) {
+ ALOGV("%s: cannot create flash control for module api v2.4+",
+ __FUNCTION__);
+ return NO_MEMORY;
+ }
+ } else {
+ uint32_t deviceVersion = CAMERA_DEVICE_API_VERSION_1_0;
+
+ if (mCameraModule->getModuleApiVersion() >=
+ CAMERA_MODULE_API_VERSION_2_0) {
+ camera_info info;
+ res = mCameraModule->getCameraInfo(
+ atoi(String8(cameraId).string()), &info);
+ if (res) {
+ ALOGE("%s: failed to get camera info for camera %s",
+ __FUNCTION__, cameraId.string());
+ return res;
+ }
+ deviceVersion = info.device_version;
+ }
+
+ if (deviceVersion >= CAMERA_DEVICE_API_VERSION_2_0) {
+ CameraDeviceClientFlashControl *flashControl =
+ new CameraDeviceClientFlashControl(*mCameraModule,
+ *mCallbacks);
+ if (!flashControl) {
+ return NO_MEMORY;
+ }
+
+ mFlashControl = flashControl;
+ } else {
+ mFlashControl =
+ new CameraHardwareInterfaceFlashControl(*mCameraModule,
+ *mCallbacks);
+ }
+ }
+
+ return OK;
+}
+
+status_t CameraFlashlight::setTorchMode(const String8& cameraId, bool enabled) {
+ if (!mFlashlightMapInitialized) {
+ ALOGE("%s: findFlashUnits() must be called before this method.");
+ return NO_INIT;
+ }
+
+ ALOGV("%s: set torch mode of camera %s to %d", __FUNCTION__,
+ cameraId.string(), enabled);
+
+ status_t res = OK;
+ Mutex::Autolock l(mLock);
+
+ if (mOpenedCameraIds.indexOf(cameraId) != NAME_NOT_FOUND) {
+ // This case is needed to avoid state corruption during the following call sequence:
+ // CameraService::setTorchMode for camera ID 0 begins, does torch status checks
+ // CameraService::connect for camera ID 0 begins, calls prepareDeviceOpen, ends
+ // CameraService::setTorchMode for camera ID 0 continues, calls
+ // CameraFlashlight::setTorchMode
+
+ // TODO: Move torch status checks and state updates behind this CameraFlashlight lock
+ // to avoid other similar race conditions.
+ ALOGE("%s: Camera device %s is in use, cannot set torch mode.",
+ __FUNCTION__, cameraId.string());
+ return -EBUSY;
+ }
+
+ if (mFlashControl == NULL) {
+ if (enabled == false) {
+ return OK;
+ }
+
+ res = createFlashlightControl(cameraId);
+ if (res) {
+ return res;
+ }
+ res = mFlashControl->setTorchMode(cameraId, enabled);
+ return res;
+ }
+
+ // if flash control already exists, turning on torch mode may fail if it's
+ // tied to another camera device for module v2.3 and below.
+ res = mFlashControl->setTorchMode(cameraId, enabled);
+ if (res == BAD_INDEX) {
+ // flash control is tied to another camera device, need to close it and
+ // try again.
+ mFlashControl.clear();
+ res = createFlashlightControl(cameraId);
+ if (res) {
+ return res;
+ }
+ res = mFlashControl->setTorchMode(cameraId, enabled);
+ }
+
+ return res;
+}
+
+status_t CameraFlashlight::findFlashUnits() {
+ Mutex::Autolock l(mLock);
+ status_t res;
+ int32_t numCameras = mCameraModule->getNumberOfCameras();
+
+ mHasFlashlightMap.clear();
+ mFlashlightMapInitialized = false;
+
+ for (int32_t i = 0; i < numCameras; i++) {
+ bool hasFlash = false;
+ String8 id = String8::format("%d", i);
+
+ res = createFlashlightControl(id);
+ if (res) {
+ ALOGE("%s: failed to create flash control for %s", __FUNCTION__,
+ id.string());
+ } else {
+ res = mFlashControl->hasFlashUnit(id, &hasFlash);
+ if (res == -EUSERS || res == -EBUSY) {
+ ALOGE("%s: failed to check if camera %s has a flash unit. Some "
+ "camera devices may be opened", __FUNCTION__,
+ id.string());
+ return res;
+ } else if (res) {
+ ALOGE("%s: failed to check if camera %s has a flash unit. %s"
+ " (%d)", __FUNCTION__, id.string(), strerror(-res),
+ res);
+ }
+
+ mFlashControl.clear();
+ }
+ mHasFlashlightMap.add(id, hasFlash);
+ }
+
+ mFlashlightMapInitialized = true;
+ return OK;
+}
+
+bool CameraFlashlight::hasFlashUnit(const String8& cameraId) {
+ status_t res;
+
+ Mutex::Autolock l(mLock);
+ return hasFlashUnitLocked(cameraId);
+}
+
+bool CameraFlashlight::hasFlashUnitLocked(const String8& cameraId) {
+ if (!mFlashlightMapInitialized) {
+ ALOGE("%s: findFlashUnits() must be called before this method.");
+ return false;
+ }
+
+ ssize_t index = mHasFlashlightMap.indexOfKey(cameraId);
+ if (index == NAME_NOT_FOUND) {
+ ALOGE("%s: camera %s not present when findFlashUnits() was called",
+ __FUNCTION__, cameraId.string());
+ return false;
+ }
+
+ return mHasFlashlightMap.valueAt(index);
+}
+
+status_t CameraFlashlight::prepareDeviceOpen(const String8& cameraId) {
+ ALOGV("%s: prepare for device open", __FUNCTION__);
+
+ Mutex::Autolock l(mLock);
+ if (!mFlashlightMapInitialized) {
+ ALOGE("%s: findFlashUnits() must be called before this method.");
+ return NO_INIT;
+ }
+
+ if (mCameraModule->getModuleApiVersion() < CAMERA_MODULE_API_VERSION_2_4) {
+ // framework is going to open a camera device, all flash light control
+ // should be closed for backward compatible support.
+ mFlashControl.clear();
+
+ if (mOpenedCameraIds.size() == 0) {
+ // notify torch unavailable for all cameras with a flash
+ int numCameras = mCameraModule->getNumberOfCameras();
+ for (int i = 0; i < numCameras; i++) {
+ if (hasFlashUnitLocked(String8::format("%d", i))) {
+ mCallbacks->torch_mode_status_change(mCallbacks,
+ String8::format("%d", i).string(),
+ TORCH_MODE_STATUS_NOT_AVAILABLE);
+ }
+ }
+ }
+
+ // close flash control that may be opened by calling hasFlashUnitLocked.
+ mFlashControl.clear();
+ }
+
+ if (mOpenedCameraIds.indexOf(cameraId) == NAME_NOT_FOUND) {
+ mOpenedCameraIds.add(cameraId);
+ }
+
+ return OK;
+}
+
+status_t CameraFlashlight::deviceClosed(const String8& cameraId) {
+ ALOGV("%s: device %s is closed", __FUNCTION__, cameraId.string());
+
+ Mutex::Autolock l(mLock);
+ if (!mFlashlightMapInitialized) {
+ ALOGE("%s: findFlashUnits() must be called before this method.");
+ return NO_INIT;
+ }
+
+ ssize_t index = mOpenedCameraIds.indexOf(cameraId);
+ if (index == NAME_NOT_FOUND) {
+ ALOGE("%s: couldn't find camera %s in the opened list", __FUNCTION__,
+ cameraId.string());
+ } else {
+ mOpenedCameraIds.removeAt(index);
+ }
+
+ // Cannot do anything until all cameras are closed.
+ if (mOpenedCameraIds.size() != 0)
+ return OK;
+
+ if (mCameraModule->getModuleApiVersion() < CAMERA_MODULE_API_VERSION_2_4) {
+ // notify torch available for all cameras with a flash
+ int numCameras = mCameraModule->getNumberOfCameras();
+ for (int i = 0; i < numCameras; i++) {
+ if (hasFlashUnitLocked(String8::format("%d", i))) {
+ mCallbacks->torch_mode_status_change(mCallbacks,
+ String8::format("%d", i).string(),
+ TORCH_MODE_STATUS_AVAILABLE_OFF);
+ }
+ }
+ }
+
+ return OK;
+}
+// CameraFlashlight implementation ends
+
+
+FlashControlBase::~FlashControlBase() {
+}
+
+/////////////////////////////////////////////////////////////////////
+// ModuleFlashControl implementation begins
+// Flash control for camera module v2.4 and above.
+/////////////////////////////////////////////////////////////////////
+ModuleFlashControl::ModuleFlashControl(CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks) :
+ mCameraModule(&cameraModule) {
+}
+
+ModuleFlashControl::~ModuleFlashControl() {
+}
+
+status_t ModuleFlashControl::hasFlashUnit(const String8& cameraId, bool *hasFlash) {
+ if (!hasFlash) {
+ return BAD_VALUE;
+ }
+
+ *hasFlash = false;
+ Mutex::Autolock l(mLock);
+
+ camera_info info;
+ status_t res = mCameraModule->getCameraInfo(atoi(cameraId.string()),
+ &info);
+ if (res != 0) {
+ return res;
+ }
+
+ CameraMetadata metadata;
+ metadata = info.static_camera_characteristics;
+ camera_metadata_entry flashAvailable =
+ metadata.find(ANDROID_FLASH_INFO_AVAILABLE);
+ if (flashAvailable.count == 1 && flashAvailable.data.u8[0] == 1) {
+ *hasFlash = true;
+ }
+
+ return OK;
+}
+
+status_t ModuleFlashControl::setTorchMode(const String8& cameraId, bool enabled) {
+ ALOGV("%s: set camera %s torch mode to %d", __FUNCTION__,
+ cameraId.string(), enabled);
+
+ Mutex::Autolock l(mLock);
+ return mCameraModule->setTorchMode(cameraId.string(), enabled);
+}
+// ModuleFlashControl implementation ends
+
+/////////////////////////////////////////////////////////////////////
+// CameraDeviceClientFlashControl implementation begins
+// Flash control for camera module <= v2.3 and camera HAL v2-v3
+/////////////////////////////////////////////////////////////////////
+CameraDeviceClientFlashControl::CameraDeviceClientFlashControl(
+ CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks) :
+ mCameraModule(&cameraModule),
+ mCallbacks(&callbacks),
+ mTorchEnabled(false),
+ mMetadata(NULL),
+ mStreaming(false) {
+}
+
+CameraDeviceClientFlashControl::~CameraDeviceClientFlashControl() {
+ disconnectCameraDevice();
+ if (mMetadata) {
+ delete mMetadata;
+ }
+
+ mAnw.clear();
+ mSurfaceTexture.clear();
+ mProducer.clear();
+ mConsumer.clear();
+
+ if (mTorchEnabled) {
+ if (mCallbacks) {
+ ALOGV("%s: notify the framework that torch was turned off",
+ __FUNCTION__);
+ mCallbacks->torch_mode_status_change(mCallbacks,
+ mCameraId.string(), TORCH_MODE_STATUS_AVAILABLE_OFF);
+ }
+ }
+}
+
+status_t CameraDeviceClientFlashControl::initializeSurface(
+ sp<CameraDeviceBase> &device, int32_t width, int32_t height) {
+ status_t res;
+ BufferQueue::createBufferQueue(&mProducer, &mConsumer);
+
+ mSurfaceTexture = new GLConsumer(mConsumer, 0, GLConsumer::TEXTURE_EXTERNAL,
+ true, true);
+ if (mSurfaceTexture == NULL) {
+ return NO_MEMORY;
+ }
+
+ int32_t format = HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;
+ res = mSurfaceTexture->setDefaultBufferSize(width, height);
+ if (res) {
+ return res;
+ }
+ res = mSurfaceTexture->setDefaultBufferFormat(format);
+ if (res) {
+ return res;
+ }
+
+ mAnw = new Surface(mProducer, /*useAsync*/ true);
+ if (mAnw == NULL) {
+ return NO_MEMORY;
+ }
+ res = device->createStream(mAnw, width, height, format,
+ HAL_DATASPACE_UNKNOWN, CAMERA3_STREAM_ROTATION_0, &mStreamId);
+ if (res) {
+ return res;
+ }
+
+ res = device->configureStreams();
+ if (res) {
+ return res;
+ }
+
+ return res;
+}
+
+status_t CameraDeviceClientFlashControl::getSmallestSurfaceSize(
+ const camera_info& info, int32_t *width, int32_t *height) {
+ if (!width || !height) {
+ return BAD_VALUE;
+ }
+
+ int32_t w = INT32_MAX;
+ int32_t h = 1;
+
+ CameraMetadata metadata;
+ metadata = info.static_camera_characteristics;
+ camera_metadata_entry streamConfigs =
+ metadata.find(ANDROID_SCALER_AVAILABLE_STREAM_CONFIGURATIONS);
+ for (size_t i = 0; i < streamConfigs.count; i += 4) {
+ int32_t fmt = streamConfigs.data.i32[i];
+ if (fmt == ANDROID_SCALER_AVAILABLE_FORMATS_IMPLEMENTATION_DEFINED) {
+ int32_t ww = streamConfigs.data.i32[i + 1];
+ int32_t hh = streamConfigs.data.i32[i + 2];
+
+ if (w * h > ww * hh) {
+ w = ww;
+ h = hh;
+ }
+ }
+ }
+
+ // if stream configuration is not found, try available processed sizes.
+ if (streamConfigs.count == 0) {
+ camera_metadata_entry availableProcessedSizes =
+ metadata.find(ANDROID_SCALER_AVAILABLE_PROCESSED_SIZES);
+ for (size_t i = 0; i < availableProcessedSizes.count; i += 2) {
+ int32_t ww = availableProcessedSizes.data.i32[i];
+ int32_t hh = availableProcessedSizes.data.i32[i + 1];
+ if (w * h > ww * hh) {
+ w = ww;
+ h = hh;
+ }
+ }
+ }
+
+ if (w == INT32_MAX) {
+ return NAME_NOT_FOUND;
+ }
+
+ *width = w;
+ *height = h;
+
+ return OK;
+}
+
+status_t CameraDeviceClientFlashControl::connectCameraDevice(
+ const String8& cameraId) {
+ camera_info info;
+ status_t res = mCameraModule->getCameraInfo(atoi(cameraId.string()), &info);
+ if (res != 0) {
+ ALOGE("%s: failed to get camera info for camera %s", __FUNCTION__,
+ cameraId.string());
+ return res;
+ }
+
+ sp<CameraDeviceBase> device =
+ CameraDeviceFactory::createDevice(atoi(cameraId.string()));
+ if (device == NULL) {
+ return NO_MEMORY;
+ }
+
+ res = device->initialize(mCameraModule);
+ if (res) {
+ return res;
+ }
+
+ int32_t width, height;
+ res = getSmallestSurfaceSize(info, &width, &height);
+ if (res) {
+ return res;
+ }
+ res = initializeSurface(device, width, height);
+ if (res) {
+ return res;
+ }
+
+ mCameraId = cameraId;
+ mStreaming = (info.device_version <= CAMERA_DEVICE_API_VERSION_3_1);
+ mDevice = device;
+
+ return OK;
+}
+
+status_t CameraDeviceClientFlashControl::disconnectCameraDevice() {
+ if (mDevice != NULL) {
+ mDevice->disconnect();
+ mDevice.clear();
+ }
+
+ return OK;
+}
+
+
+
+status_t CameraDeviceClientFlashControl::hasFlashUnit(const String8& cameraId,
+ bool *hasFlash) {
+ ALOGV("%s: checking if camera %s has a flash unit", __FUNCTION__,
+ cameraId.string());
+
+ Mutex::Autolock l(mLock);
+ return hasFlashUnitLocked(cameraId, hasFlash);
+
+}
+
+status_t CameraDeviceClientFlashControl::hasFlashUnitLocked(
+ const String8& cameraId, bool *hasFlash) {
+ if (!hasFlash) {
+ return BAD_VALUE;
+ }
+
+ camera_info info;
+ status_t res = mCameraModule->getCameraInfo(
+ atoi(cameraId.string()), &info);
+ if (res != 0) {
+ ALOGE("%s: failed to get camera info for camera %s", __FUNCTION__,
+ cameraId.string());
+ return res;
+ }
+
+ CameraMetadata metadata;
+ metadata = info.static_camera_characteristics;
+ camera_metadata_entry flashAvailable =
+ metadata.find(ANDROID_FLASH_INFO_AVAILABLE);
+ if (flashAvailable.count == 1 && flashAvailable.data.u8[0] == 1) {
+ *hasFlash = true;
+ }
+
+ return OK;
+}
+
+status_t CameraDeviceClientFlashControl::submitTorchEnabledRequest() {
+ status_t res;
+
+ if (mMetadata == NULL) {
+ mMetadata = new CameraMetadata();
+ if (mMetadata == NULL) {
+ return NO_MEMORY;
+ }
+ res = mDevice->createDefaultRequest(
+ CAMERA3_TEMPLATE_PREVIEW, mMetadata);
+ if (res) {
+ return res;
+ }
+ }
+
+ uint8_t torchOn = ANDROID_FLASH_MODE_TORCH;
+ mMetadata->update(ANDROID_FLASH_MODE, &torchOn, 1);
+ mMetadata->update(ANDROID_REQUEST_OUTPUT_STREAMS, &mStreamId, 1);
+
+ uint8_t aeMode = ANDROID_CONTROL_AE_MODE_ON;
+ mMetadata->update(ANDROID_CONTROL_AE_MODE, &aeMode, 1);
+
+ int32_t requestId = 0;
+ mMetadata->update(ANDROID_REQUEST_ID, &requestId, 1);
+
+ if (mStreaming) {
+ res = mDevice->setStreamingRequest(*mMetadata);
+ } else {
+ res = mDevice->capture(*mMetadata);
+ }
+ return res;
+}
+
+
+
+
+status_t CameraDeviceClientFlashControl::setTorchMode(
+ const String8& cameraId, bool enabled) {
+ bool hasFlash = false;
+
+ Mutex::Autolock l(mLock);
+ status_t res = hasFlashUnitLocked(cameraId, &hasFlash);
+
+ // pre-check
+ if (enabled) {
+ // invalid camera?
+ if (res) {
+ return -EINVAL;
+ }
+ // no flash unit?
+ if (!hasFlash) {
+ return -ENOSYS;
+ }
+ // already opened for a different device?
+ if (mDevice != NULL && cameraId != mCameraId) {
+ return BAD_INDEX;
+ }
+ } else if (mDevice == NULL || cameraId != mCameraId) {
+ // disabling the torch mode of an un-opened or different device.
+ return OK;
+ } else {
+ // disabling the torch mode of currently opened device
+ disconnectCameraDevice();
+ mTorchEnabled = false;
+ mCallbacks->torch_mode_status_change(mCallbacks,
+ cameraId.string(), TORCH_MODE_STATUS_AVAILABLE_OFF);
+ return OK;
+ }
+
+ if (mDevice == NULL) {
+ res = connectCameraDevice(cameraId);
+ if (res) {
+ return res;
+ }
+ }
+
+ res = submitTorchEnabledRequest();
+ if (res) {
+ return res;
+ }
+
+ mTorchEnabled = true;
+ mCallbacks->torch_mode_status_change(mCallbacks,
+ cameraId.string(), TORCH_MODE_STATUS_AVAILABLE_ON);
+ return OK;
+}
+// CameraDeviceClientFlashControl implementation ends
+
+
+/////////////////////////////////////////////////////////////////////
+// CameraHardwareInterfaceFlashControl implementation begins
+// Flash control for camera module <= v2.3 and camera HAL v1
+/////////////////////////////////////////////////////////////////////
+CameraHardwareInterfaceFlashControl::CameraHardwareInterfaceFlashControl(
+ CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks) :
+ mCameraModule(&cameraModule),
+ mCallbacks(&callbacks),
+ mTorchEnabled(false) {
+
+}
+
+CameraHardwareInterfaceFlashControl::~CameraHardwareInterfaceFlashControl() {
+ disconnectCameraDevice();
+
+ mAnw.clear();
+ mSurfaceTexture.clear();
+ mProducer.clear();
+ mConsumer.clear();
+
+ if (mTorchEnabled) {
+ if (mCallbacks) {
+ ALOGV("%s: notify the framework that torch was turned off",
+ __FUNCTION__);
+ mCallbacks->torch_mode_status_change(mCallbacks,
+ mCameraId.string(), TORCH_MODE_STATUS_AVAILABLE_OFF);
+ }
+ }
+}
+
+status_t CameraHardwareInterfaceFlashControl::setTorchMode(
+ const String8& cameraId, bool enabled) {
+ Mutex::Autolock l(mLock);
+
+ // pre-check
+ status_t res;
+ if (enabled) {
+ bool hasFlash = false;
+ res = hasFlashUnitLocked(cameraId, &hasFlash);
+ // invalid camera?
+ if (res) {
+ // hasFlashUnitLocked() returns BAD_INDEX if mDevice is connected to
+ // another camera device.
+ return res == BAD_INDEX ? BAD_INDEX : -EINVAL;
+ }
+ // no flash unit?
+ if (!hasFlash) {
+ return -ENOSYS;
+ }
+ } else if (mDevice == NULL || cameraId != mCameraId) {
+ // disabling the torch mode of an un-opened or different device.
+ return OK;
+ } else {
+ // disabling the torch mode of currently opened device
+ disconnectCameraDevice();
+ mTorchEnabled = false;
+ mCallbacks->torch_mode_status_change(mCallbacks,
+ cameraId.string(), TORCH_MODE_STATUS_AVAILABLE_OFF);
+ return OK;
+ }
+
+ res = startPreviewAndTorch();
+ if (res) {
+ return res;
+ }
+
+ mTorchEnabled = true;
+ mCallbacks->torch_mode_status_change(mCallbacks,
+ cameraId.string(), TORCH_MODE_STATUS_AVAILABLE_ON);
+ return OK;
+}
+
+status_t CameraHardwareInterfaceFlashControl::hasFlashUnit(
+ const String8& cameraId, bool *hasFlash) {
+ Mutex::Autolock l(mLock);
+ return hasFlashUnitLocked(cameraId, hasFlash);
+}
+
+status_t CameraHardwareInterfaceFlashControl::hasFlashUnitLocked(
+ const String8& cameraId, bool *hasFlash) {
+ if (!hasFlash) {
+ return BAD_VALUE;
+ }
+
+ status_t res;
+ if (mDevice == NULL) {
+ res = connectCameraDevice(cameraId);
+ if (res) {
+ return res;
+ }
+ }
+
+ if (cameraId != mCameraId) {
+ return BAD_INDEX;
+ }
+
+ const char *flashMode =
+ mParameters.get(CameraParameters::KEY_SUPPORTED_FLASH_MODES);
+ if (flashMode && strstr(flashMode, CameraParameters::FLASH_MODE_TORCH)) {
+ *hasFlash = true;
+ } else {
+ *hasFlash = false;
+ }
+
+ return OK;
+}
+
+status_t CameraHardwareInterfaceFlashControl::startPreviewAndTorch() {
+ status_t res = OK;
+ res = mDevice->startPreview();
+ if (res) {
+ ALOGE("%s: start preview failed. %s (%d)", __FUNCTION__,
+ strerror(-res), res);
+ return res;
+ }
+
+ mParameters.set(CameraParameters::KEY_FLASH_MODE,
+ CameraParameters::FLASH_MODE_TORCH);
+
+ return mDevice->setParameters(mParameters);
+}
+
+status_t CameraHardwareInterfaceFlashControl::getSmallestSurfaceSize(
+ int32_t *width, int32_t *height) {
+ if (!width || !height) {
+ return BAD_VALUE;
+ }
+
+ int32_t w = INT32_MAX;
+ int32_t h = 1;
+ Vector<Size> sizes;
+
+ mParameters.getSupportedPreviewSizes(sizes);
+ for (size_t i = 0; i < sizes.size(); i++) {
+ Size s = sizes[i];
+ if (w * h > s.width * s.height) {
+ w = s.width;
+ h = s.height;
+ }
+ }
+
+ if (w == INT32_MAX) {
+ return NAME_NOT_FOUND;
+ }
+
+ *width = w;
+ *height = h;
+
+ return OK;
+}
+
+status_t CameraHardwareInterfaceFlashControl::initializePreviewWindow(
+ sp<CameraHardwareInterface> device, int32_t width, int32_t height) {
+ status_t res;
+ BufferQueue::createBufferQueue(&mProducer, &mConsumer);
+
+ mSurfaceTexture = new GLConsumer(mConsumer, 0, GLConsumer::TEXTURE_EXTERNAL,
+ true, true);
+ if (mSurfaceTexture == NULL) {
+ return NO_MEMORY;
+ }
+
+ int32_t format = HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;
+ res = mSurfaceTexture->setDefaultBufferSize(width, height);
+ if (res) {
+ return res;
+ }
+ res = mSurfaceTexture->setDefaultBufferFormat(format);
+ if (res) {
+ return res;
+ }
+
+ mAnw = new Surface(mProducer, /*useAsync*/ true);
+ if (mAnw == NULL) {
+ return NO_MEMORY;
+ }
+
+ res = native_window_api_connect(mAnw.get(), NATIVE_WINDOW_API_CAMERA);
+ if (res) {
+ ALOGE("%s: Unable to connect to native window", __FUNCTION__);
+ return res;
+ }
+
+ return device->setPreviewWindow(mAnw);
+}
+
+status_t CameraHardwareInterfaceFlashControl::connectCameraDevice(
+ const String8& cameraId) {
+ sp<CameraHardwareInterface> device =
+ new CameraHardwareInterface(cameraId.string());
+
+ status_t res = device->initialize(mCameraModule);
+ if (res) {
+ ALOGE("%s: initializing camera %s failed", __FUNCTION__,
+ cameraId.string());
+ return res;
+ }
+
+ // need to set __get_memory in set_callbacks().
+ device->setCallbacks(NULL, NULL, NULL, NULL);
+
+ mParameters = device->getParameters();
+
+ int32_t width, height;
+ res = getSmallestSurfaceSize(&width, &height);
+ if (res) {
+ ALOGE("%s: failed to get smallest surface size for camera %s",
+ __FUNCTION__, cameraId.string());
+ return res;
+ }
+
+ res = initializePreviewWindow(device, width, height);
+ if (res) {
+ ALOGE("%s: failed to initialize preview window for camera %s",
+ __FUNCTION__, cameraId.string());
+ return res;
+ }
+
+ mCameraId = cameraId;
+ mDevice = device;
+ return OK;
+}
+
+status_t CameraHardwareInterfaceFlashControl::disconnectCameraDevice() {
+ if (mDevice == NULL) {
+ return OK;
+ }
+
+ mParameters.set(CameraParameters::KEY_FLASH_MODE,
+ CameraParameters::FLASH_MODE_OFF);
+ mDevice->setParameters(mParameters);
+ mDevice->stopPreview();
+ status_t res = native_window_api_disconnect(mAnw.get(),
+ NATIVE_WINDOW_API_CAMERA);
+ if (res) {
+ ALOGW("%s: native_window_api_disconnect failed: %s (%d)",
+ __FUNCTION__, strerror(-res), res);
+ }
+ mDevice->setPreviewWindow(NULL);
+ mDevice->release();
+
+ return OK;
+}
+// CameraHardwareInterfaceFlashControl implementation ends
+
+}
diff --git a/services/camera/libcameraservice/CameraFlashlight.h b/services/camera/libcameraservice/CameraFlashlight.h
new file mode 100644
index 0000000..30f01f0
--- /dev/null
+++ b/services/camera/libcameraservice/CameraFlashlight.h
@@ -0,0 +1,225 @@
+/*
+ * Copyright (C) 2015 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_SERVERS_CAMERA_CAMERAFLASHLIGHT_H
+#define ANDROID_SERVERS_CAMERA_CAMERAFLASHLIGHT_H
+
+#include "hardware/camera_common.h"
+#include "utils/KeyedVector.h"
+#include "utils/SortedVector.h"
+#include "gui/GLConsumer.h"
+#include "gui/Surface.h"
+#include "common/CameraDeviceBase.h"
+#include "device1/CameraHardwareInterface.h"
+
+namespace android {
+
+/**
+ * FlashControlBase is a base class for flash control. It defines the functions
+ * that a flash control for each camera module/device version should implement.
+ */
+class FlashControlBase : public virtual VirtualLightRefBase {
+ public:
+ virtual ~FlashControlBase();
+
+ // Whether a camera device has a flash unit. Calling this function may
+ // cause the torch mode to be turned off in HAL v1 devices. If
+ // previously-on torch mode is turned off,
+ // callbacks.torch_mode_status_change() should be invoked.
+ virtual status_t hasFlashUnit(const String8& cameraId,
+ bool *hasFlash) = 0;
+
+ // set the torch mode to on or off.
+ virtual status_t setTorchMode(const String8& cameraId,
+ bool enabled) = 0;
+};
+
+/**
+ * CameraFlashlight can be used by camera service to control flashflight.
+ */
+class CameraFlashlight : public virtual VirtualLightRefBase {
+ public:
+ CameraFlashlight(CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks);
+ virtual ~CameraFlashlight();
+
+ // Find all flash units. This must be called before other methods. All
+ // camera devices must be closed when it's called because HAL v1 devices
+ // need to be opened to query available flash modes.
+ status_t findFlashUnits();
+
+ // Whether a camera device has a flash unit. Before findFlashUnits() is
+ // called, this function always returns false.
+ bool hasFlashUnit(const String8& cameraId);
+
+ // set the torch mode to on or off.
+ status_t setTorchMode(const String8& cameraId, bool enabled);
+
+ // Notify CameraFlashlight that camera service is going to open a camera
+ // device. CameraFlashlight will free the resources that may cause the
+ // camera open to fail. Camera service must call this function before
+ // opening a camera device.
+ status_t prepareDeviceOpen(const String8& cameraId);
+
+ // Notify CameraFlashlight that camera service has closed a camera
+ // device. CameraFlashlight may invoke callbacks for torch mode
+ // available depending on the implementation.
+ status_t deviceClosed(const String8& cameraId);
+
+ private:
+ // create flashlight control based on camera module API and camera
+ // device API versions.
+ status_t createFlashlightControl(const String8& cameraId);
+
+ // mLock should be locked.
+ bool hasFlashUnitLocked(const String8& cameraId);
+
+ sp<FlashControlBase> mFlashControl;
+ CameraModule *mCameraModule;
+ const camera_module_callbacks_t *mCallbacks;
+ SortedVector<String8> mOpenedCameraIds;
+
+ // camera id -> if it has a flash unit
+ KeyedVector<String8, bool> mHasFlashlightMap;
+ bool mFlashlightMapInitialized;
+
+ Mutex mLock; // protect CameraFlashlight API
+};
+
+/**
+ * Flash control for camera module v2.4 and above.
+ */
+class ModuleFlashControl : public FlashControlBase {
+ public:
+ ModuleFlashControl(CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks);
+ virtual ~ModuleFlashControl();
+
+ // FlashControlBase
+ status_t hasFlashUnit(const String8& cameraId, bool *hasFlash);
+ status_t setTorchMode(const String8& cameraId, bool enabled);
+
+ private:
+ CameraModule *mCameraModule;
+
+ Mutex mLock;
+};
+
+/**
+ * Flash control for camera module <= v2.3 and camera HAL v2-v3
+ */
+class CameraDeviceClientFlashControl : public FlashControlBase {
+ public:
+ CameraDeviceClientFlashControl(CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks);
+ virtual ~CameraDeviceClientFlashControl();
+
+ // FlashControlBase
+ status_t setTorchMode(const String8& cameraId, bool enabled);
+ status_t hasFlashUnit(const String8& cameraId, bool *hasFlash);
+
+ private:
+ // connect to a camera device
+ status_t connectCameraDevice(const String8& cameraId);
+ // disconnect and free mDevice
+ status_t disconnectCameraDevice();
+
+ // initialize a surface
+ status_t initializeSurface(sp<CameraDeviceBase>& device, int32_t width,
+ int32_t height);
+
+ // submit a request to enable the torch mode
+ status_t submitTorchEnabledRequest();
+
+ // get the smallest surface size of IMPLEMENTATION_DEFINED
+ status_t getSmallestSurfaceSize(const camera_info& info, int32_t *width,
+ int32_t *height);
+
+ // protected by mLock
+ status_t hasFlashUnitLocked(const String8& cameraId, bool *hasFlash);
+
+ CameraModule *mCameraModule;
+ const camera_module_callbacks_t *mCallbacks;
+ String8 mCameraId;
+ bool mTorchEnabled;
+ CameraMetadata *mMetadata;
+ // WORKAROUND: will be set to true for HAL v2 devices where
+ // setStreamingRequest() needs to be call for torch mode settings to
+ // take effect.
+ bool mStreaming;
+
+ sp<CameraDeviceBase> mDevice;
+
+ sp<IGraphicBufferProducer> mProducer;
+ sp<IGraphicBufferConsumer> mConsumer;
+ sp<GLConsumer> mSurfaceTexture;
+ sp<ANativeWindow> mAnw;
+ int32_t mStreamId;
+
+ Mutex mLock;
+};
+
+/**
+ * Flash control for camera module <= v2.3 and camera HAL v1
+ */
+class CameraHardwareInterfaceFlashControl : public FlashControlBase {
+ public:
+ CameraHardwareInterfaceFlashControl(CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks);
+ virtual ~CameraHardwareInterfaceFlashControl();
+
+ // FlashControlBase
+ status_t setTorchMode(const String8& cameraId, bool enabled);
+ status_t hasFlashUnit(const String8& cameraId, bool *hasFlash);
+
+ private:
+ // connect to a camera device
+ status_t connectCameraDevice(const String8& cameraId);
+
+ // disconnect and free mDevice
+ status_t disconnectCameraDevice();
+
+ // initialize the preview window
+ status_t initializePreviewWindow(sp<CameraHardwareInterface> device,
+ int32_t width, int32_t height);
+
+ // start preview and enable torch
+ status_t startPreviewAndTorch();
+
+ // get the smallest surface
+ status_t getSmallestSurfaceSize(int32_t *width, int32_t *height);
+
+ // protected by mLock
+ status_t hasFlashUnitLocked(const String8& cameraId, bool *hasFlash);
+
+ CameraModule *mCameraModule;
+ const camera_module_callbacks_t *mCallbacks;
+ sp<CameraHardwareInterface> mDevice;
+ String8 mCameraId;
+ CameraParameters mParameters;
+ bool mTorchEnabled;
+
+ sp<IGraphicBufferProducer> mProducer;
+ sp<IGraphicBufferConsumer> mConsumer;
+ sp<GLConsumer> mSurfaceTexture;
+ sp<ANativeWindow> mAnw;
+
+ Mutex mLock;
+};
+
+} // namespace android
+
+#endif
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 1232c32..e9c96c6 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -17,9 +17,14 @@
#define LOG_TAG "CameraService"
//#define LOG_NDEBUG 0
+#include <algorithm>
+#include <climits>
#include <stdio.h>
-#include <string.h>
+#include <cstring>
+#include <ctime>
+#include <string>
#include <sys/types.h>
+#include <inttypes.h>
#include <pthread.h>
#include <binder/AppOpsManager.h>
@@ -27,6 +32,7 @@
#include <binder/IServiceManager.h>
#include <binder/MemoryBase.h>
#include <binder/MemoryHeapBase.h>
+#include <binder/ProcessInfoService.h>
#include <cutils/atomic.h>
#include <cutils/properties.h>
#include <gui/Surface.h>
@@ -45,7 +51,6 @@
#include "CameraService.h"
#include "api1/CameraClient.h"
#include "api1/Camera2Client.h"
-#include "api_pro/ProCamera2Client.h"
#include "api2/CameraDeviceClient.h"
#include "utils/CameraTraces.h"
#include "CameraDeviceFactory.h"
@@ -66,25 +71,48 @@
// ----------------------------------------------------------------------------
-static int getCallingPid() {
- return IPCThreadState::self()->getCallingPid();
-}
-
-static int getCallingUid() {
- return IPCThreadState::self()->getCallingUid();
-}
-
extern "C" {
static void camera_device_status_change(
const struct camera_module_callbacks* callbacks,
int camera_id,
int new_status) {
sp<CameraService> cs = const_cast<CameraService*>(
+ static_cast<const CameraService*>(callbacks));
+
+ cs->onDeviceStatusChanged(static_cast<camera_device_status_t>(camera_id),
+ static_cast<camera_device_status_t>(new_status));
+}
+
+static void torch_mode_status_change(
+ const struct camera_module_callbacks* callbacks,
+ const char* camera_id,
+ int new_status) {
+ if (!callbacks || !camera_id) {
+ ALOGE("%s invalid parameters. callbacks %p, camera_id %p", __FUNCTION__,
+ callbacks, camera_id);
+ }
+ sp<CameraService> cs = const_cast<CameraService*>(
static_cast<const CameraService*>(callbacks));
- cs->onDeviceStatusChanged(
- camera_id,
- new_status);
+ ICameraServiceListener::TorchStatus status;
+ switch (new_status) {
+ case TORCH_MODE_STATUS_NOT_AVAILABLE:
+ status = ICameraServiceListener::TORCH_STATUS_NOT_AVAILABLE;
+ break;
+ case TORCH_MODE_STATUS_AVAILABLE_OFF:
+ status = ICameraServiceListener::TORCH_STATUS_AVAILABLE_OFF;
+ break;
+ case TORCH_MODE_STATUS_AVAILABLE_ON:
+ status = ICameraServiceListener::TORCH_STATUS_AVAILABLE_ON;
+ break;
+ default:
+ ALOGE("Unknown torch status %d", new_status);
+ return;
+ }
+
+ cs->onTorchStatusChanged(
+ String8(camera_id),
+ status);
}
} // extern "C"
@@ -94,50 +122,89 @@
// should be ok for now.
static CameraService *gCameraService;
-CameraService::CameraService()
- :mSoundRef(0), mModule(0)
-{
+CameraService::CameraService() : mEventLog(DEFAULT_EVICTION_LOG_LENGTH),
+ mLastUserId(DEFAULT_LAST_USER_ID), mSoundRef(0), mModule(0), mFlashlight(0) {
ALOGI("CameraService started (pid=%d)", getpid());
gCameraService = this;
- for (size_t i = 0; i < MAX_CAMERAS; ++i) {
- mStatusList[i] = ICameraServiceListener::STATUS_PRESENT;
- }
-
this->camera_device_status_change = android::camera_device_status_change;
+ this->torch_mode_status_change = android::torch_mode_status_change;
+
+ mServiceLockWrapper = std::make_shared<WaitableMutexWrapper>(&mServiceLock);
}
void CameraService::onFirstRef()
{
- LOG1("CameraService::onFirstRef");
+ ALOGI("CameraService process starting");
BnCameraService::onFirstRef();
+ camera_module_t *rawModule;
if (hw_get_module(CAMERA_HARDWARE_MODULE_ID,
- (const hw_module_t **)&mModule) < 0) {
+ (const hw_module_t **)&rawModule) < 0) {
ALOGE("Could not load camera HAL module");
mNumberOfCameras = 0;
}
else {
- ALOGI("Loaded \"%s\" camera module", mModule->common.name);
- mNumberOfCameras = mModule->get_number_of_cameras();
- if (mNumberOfCameras > MAX_CAMERAS) {
- ALOGE("Number of cameras(%d) > MAX_CAMERAS(%d).",
- mNumberOfCameras, MAX_CAMERAS);
- mNumberOfCameras = MAX_CAMERAS;
- }
- for (int i = 0; i < mNumberOfCameras; i++) {
- setCameraFree(i);
+ mModule = new CameraModule(rawModule);
+ ALOGI("Loaded \"%s\" camera module", mModule->getModuleName());
+ mNumberOfCameras = mModule->getNumberOfCameras();
+
+ mFlashlight = new CameraFlashlight(*mModule, *this);
+ status_t res = mFlashlight->findFlashUnits();
+ if (res) {
+ // impossible because we haven't open any camera devices.
+ ALOGE("Failed to find flash units.");
}
- if (mModule->common.module_api_version >=
- CAMERA_MODULE_API_VERSION_2_1) {
- mModule->set_callbacks(this);
+ for (int i = 0; i < mNumberOfCameras; i++) {
+ String8 cameraId = String8::format("%d", i);
+
+ // Defaults to use for cost and conflicting devices
+ int cost = 100;
+ char** conflicting_devices = nullptr;
+ size_t conflicting_devices_length = 0;
+
+ // If using post-2.4 module version, query the cost + conflicting devices from the HAL
+ if (mModule->getModuleApiVersion() >= CAMERA_MODULE_API_VERSION_2_4) {
+ struct camera_info info;
+ status_t rc = mModule->getCameraInfo(i, &info);
+ if (rc == NO_ERROR) {
+ cost = info.resource_cost;
+ conflicting_devices = info.conflicting_devices;
+ conflicting_devices_length = info.conflicting_devices_length;
+ } else {
+ ALOGE("%s: Received error loading camera info for device %d, cost and"
+ " conflicting devices fields set to defaults for this device.",
+ __FUNCTION__, i);
+ }
+ }
+
+ std::set<String8> conflicting;
+ for (size_t i = 0; i < conflicting_devices_length; i++) {
+ conflicting.emplace(String8(conflicting_devices[i]));
+ }
+
+ // Initialize state for each camera device
+ {
+ Mutex::Autolock lock(mCameraStatesLock);
+ mCameraStates.emplace(cameraId, std::make_shared<CameraState>(cameraId, cost,
+ conflicting));
+ }
+
+ if (mFlashlight->hasFlashUnit(cameraId)) {
+ mTorchStatusMap.add(cameraId,
+ ICameraServiceListener::TORCH_STATUS_AVAILABLE_OFF);
+ }
+ }
+
+ if (mModule->getModuleApiVersion() >= CAMERA_MODULE_API_VERSION_2_1) {
+ mModule->setCallbacks(this);
}
VendorTagDescriptor::clearGlobalVendorTagDescriptor();
- if (mModule->common.module_api_version >= CAMERA_MODULE_API_VERSION_2_2) {
+ if (mModule->getModuleApiVersion() >= CAMERA_MODULE_API_VERSION_2_2) {
setUpVendorTags();
}
@@ -146,80 +213,111 @@
}
CameraService::~CameraService() {
- for (int i = 0; i < mNumberOfCameras; i++) {
- if (mBusy[i]) {
- ALOGE("camera %d is still in use in destructor!", i);
- }
+ if (mModule) {
+ delete mModule;
+ mModule = nullptr;
}
-
VendorTagDescriptor::clearGlobalVendorTagDescriptor();
- gCameraService = NULL;
+ gCameraService = nullptr;
}
-void CameraService::onDeviceStatusChanged(int cameraId,
- int newStatus)
-{
+void CameraService::onDeviceStatusChanged(camera_device_status_t cameraId,
+ camera_device_status_t newStatus) {
ALOGI("%s: Status changed for cameraId=%d, newStatus=%d", __FUNCTION__,
cameraId, newStatus);
- if (cameraId < 0 || cameraId >= MAX_CAMERAS) {
+ String8 id = String8::format("%d", cameraId);
+ std::shared_ptr<CameraState> state = getCameraState(id);
+
+ if (state == nullptr) {
ALOGE("%s: Bad camera ID %d", __FUNCTION__, cameraId);
return;
}
- if ((int)getStatus(cameraId) == newStatus) {
- ALOGE("%s: State transition to the same status 0x%x not allowed",
+ ICameraServiceListener::Status oldStatus = state->getStatus();
+
+ if (oldStatus == static_cast<ICameraServiceListener::Status>(newStatus)) {
+ ALOGE("%s: State transition to the same status %#x not allowed", __FUNCTION__, newStatus);
+ return;
+ }
+
+ if (newStatus == CAMERA_DEVICE_STATUS_NOT_PRESENT) {
+ sp<BasicClient> clientToDisconnect;
+ {
+ // Don't do this in updateStatus to avoid deadlock over mServiceLock
+ Mutex::Autolock lock(mServiceLock);
+
+ // Set the device status to NOT_PRESENT, clients will no longer be able to connect
+ // to this device until the status changes
+ updateStatus(ICameraServiceListener::STATUS_NOT_PRESENT, id);
+
+ // Remove cached shim parameters
+ state->setShimParams(CameraParameters());
+
+ // Remove the client from the list of active clients
+ clientToDisconnect = removeClientLocked(id);
+
+ // Notify the client of disconnection
+ clientToDisconnect->notifyError(ICameraDeviceCallbacks::ERROR_CAMERA_DISCONNECTED,
+ CaptureResultExtras{});
+ }
+
+ ALOGI("%s: Client for camera ID %s evicted due to device status change from HAL",
+ __FUNCTION__, id.string());
+
+ // Disconnect client
+ if (clientToDisconnect.get() != nullptr) {
+ // Ensure not in binder RPC so client disconnect PID checks work correctly
+ LOG_ALWAYS_FATAL_IF(getCallingPid() != getpid(),
+ "onDeviceStatusChanged must be called from the camera service process!");
+ clientToDisconnect->disconnect();
+ }
+
+ } else {
+ updateStatus(static_cast<ICameraServiceListener::Status>(newStatus), id);
+ }
+
+}
+
+void CameraService::onTorchStatusChanged(const String8& cameraId,
+ ICameraServiceListener::TorchStatus newStatus) {
+ Mutex::Autolock al(mTorchStatusMutex);
+ onTorchStatusChangedLocked(cameraId, newStatus);
+}
+
+void CameraService::onTorchStatusChangedLocked(const String8& cameraId,
+ ICameraServiceListener::TorchStatus newStatus) {
+ ALOGI("%s: Torch status changed for cameraId=%s, newStatus=%d",
+ __FUNCTION__, cameraId.string(), newStatus);
+
+ ICameraServiceListener::TorchStatus status;
+ status_t res = getTorchStatusLocked(cameraId, &status);
+ if (res) {
+ ALOGE("%s: cannot get torch status of camera %s", cameraId.string());
+ return;
+ }
+ if (status == newStatus) {
+ ALOGE("%s: Torch state transition to the same status 0x%x not allowed",
__FUNCTION__, (uint32_t)newStatus);
return;
}
- /* don't do this in updateStatus
- since it is also called from connect and we could get into a deadlock */
- if (newStatus == CAMERA_DEVICE_STATUS_NOT_PRESENT) {
- Vector<sp<BasicClient> > clientsToDisconnect;
- {
- Mutex::Autolock al(mServiceLock);
-
- /* Remove cached parameters from shim cache */
- mShimParams.removeItem(cameraId);
-
- /* Find all clients that we need to disconnect */
- sp<BasicClient> client = mClient[cameraId].promote();
- if (client.get() != NULL) {
- clientsToDisconnect.push_back(client);
- }
-
- int i = cameraId;
- for (size_t j = 0; j < mProClientList[i].size(); ++j) {
- sp<ProClient> cl = mProClientList[i][j].promote();
- if (cl != NULL) {
- clientsToDisconnect.push_back(cl);
- }
- }
- }
-
- /* now disconnect them. don't hold the lock
- or we can get into a deadlock */
-
- for (size_t i = 0; i < clientsToDisconnect.size(); ++i) {
- sp<BasicClient> client = clientsToDisconnect[i];
-
- client->disconnect();
- /**
- * The remote app will no longer be able to call methods on the
- * client since the client PID will be reset to 0
- */
- }
-
- ALOGV("%s: After unplug, disconnected %zu clients",
- __FUNCTION__, clientsToDisconnect.size());
+ res = setTorchStatusLocked(cameraId, newStatus);
+ if (res) {
+ ALOGE("%s: Failed to set the torch status", __FUNCTION__,
+ (uint32_t)newStatus);
+ return;
}
- updateStatus(
- static_cast<ICameraServiceListener::Status>(newStatus), cameraId);
-
+ {
+ Mutex::Autolock lock(mStatusListenerLock);
+ for (auto& i : mListenerList) {
+ i->onTorchStatusChanged(newStatus, String16{cameraId});
+ }
+ }
}
+
int32_t CameraService::getNumberOfCameras() {
return mNumberOfCameras;
}
@@ -236,12 +334,21 @@
struct camera_info info;
status_t rc = filterGetInfoErrorCode(
- mModule->get_camera_info(cameraId, &info));
+ mModule->getCameraInfo(cameraId, &info));
cameraInfo->facing = info.facing;
cameraInfo->orientation = info.orientation;
return rc;
}
+int CameraService::cameraIdToInt(const String8& cameraId) {
+ errno = 0;
+ size_t pos = 0;
+ int ret = stoi(std::string{cameraId.string()}, &pos);
+ if (errno != 0 || pos != cameraId.size()) {
+ return -1;
+ }
+ return ret;
+}
status_t CameraService::generateShimMetadata(int cameraId, /*out*/CameraMetadata* cameraInfo) {
status_t ret = OK;
@@ -347,7 +454,7 @@
int facing;
status_t ret = OK;
- if (mModule->common.module_api_version < CAMERA_MODULE_API_VERSION_2_0 ||
+ if (mModule->getModuleApiVersion() < CAMERA_MODULE_API_VERSION_2_0 ||
getDeviceVersion(cameraId, &facing) <= CAMERA_DEVICE_API_VERSION_2_1 ) {
/**
* Backwards compatibility mode for old HALs:
@@ -368,13 +475,61 @@
* Normal HAL 2.1+ codepath.
*/
struct camera_info info;
- ret = filterGetInfoErrorCode(mModule->get_camera_info(cameraId, &info));
+ ret = filterGetInfoErrorCode(mModule->getCameraInfo(cameraId, &info));
*cameraInfo = info.static_camera_characteristics;
}
return ret;
}
+int CameraService::getCallingPid() {
+ return IPCThreadState::self()->getCallingPid();
+}
+
+int CameraService::getCallingUid() {
+ return IPCThreadState::self()->getCallingUid();
+}
+
+String8 CameraService::getFormattedCurrentTime() {
+ time_t now = time(nullptr);
+ char formattedTime[64];
+ strftime(formattedTime, sizeof(formattedTime), "%m-%d %H:%M:%S", localtime(&now));
+ return String8(formattedTime);
+}
+
+int CameraService::getCameraPriorityFromProcState(int procState) {
+ // Find the priority for the camera usage based on the process state. Higher priority clients
+ // win for evictions.
+ // Note: Unlike the ordering for ActivityManager, persistent system processes will always lose
+ // the camera to the top/foreground applications.
+ switch(procState) {
+ case PROCESS_STATE_TOP: // User visible
+ return 100;
+ case PROCESS_STATE_IMPORTANT_FOREGROUND: // Foreground
+ return 90;
+ case PROCESS_STATE_PERSISTENT: // Persistent system services
+ case PROCESS_STATE_PERSISTENT_UI:
+ return 80;
+ case PROCESS_STATE_IMPORTANT_BACKGROUND: // "Important" background processes
+ return 70;
+ case PROCESS_STATE_BACKUP: // Everything else
+ case PROCESS_STATE_HEAVY_WEIGHT:
+ case PROCESS_STATE_SERVICE:
+ case PROCESS_STATE_RECEIVER:
+ case PROCESS_STATE_HOME:
+ case PROCESS_STATE_LAST_ACTIVITY:
+ case PROCESS_STATE_CACHED_ACTIVITY:
+ case PROCESS_STATE_CACHED_ACTIVITY_CLIENT:
+ case PROCESS_STATE_CACHED_EMPTY:
+ return 1;
+ case PROCESS_STATE_NONEXISTENT:
+ return -1;
+ default:
+ ALOGE("%s: Received unknown process state from ActivityManagerService!", __FUNCTION__);
+ return -1;
+ }
+}
+
status_t CameraService::getCameraVendorTagDescriptor(/*out*/sp<VendorTagDescriptor>& desc) {
if (!mModule) {
ALOGE("%s: camera hardware module doesn't exist", __FUNCTION__);
@@ -387,12 +542,12 @@
int CameraService::getDeviceVersion(int cameraId, int* facing) {
struct camera_info info;
- if (mModule->get_camera_info(cameraId, &info) != OK) {
+ if (mModule->getCameraInfo(cameraId, &info) != OK) {
return -1;
}
int deviceVersion;
- if (mModule->common.module_api_version >= CAMERA_MODULE_API_VERSION_2_0) {
+ if (mModule->getModuleApiVersion() >= CAMERA_MODULE_API_VERSION_2_0) {
deviceVersion = info.device_version;
} else {
deviceVersion = CAMERA_DEVICE_API_VERSION_1_0;
@@ -405,19 +560,6 @@
return deviceVersion;
}
-status_t CameraService::filterOpenErrorCode(status_t err) {
- switch(err) {
- case NO_ERROR:
- case -EBUSY:
- case -EINVAL:
- case -EUSERS:
- return err;
- default:
- break;
- }
- return -ENODEV;
-}
-
status_t CameraService::filterGetInfoErrorCode(status_t err) {
switch(err) {
case NO_ERROR:
@@ -433,13 +575,13 @@
vendor_tag_ops_t vOps = vendor_tag_ops_t();
// Check if vendor operations have been implemented
- if (mModule->get_vendor_tag_ops == NULL) {
+ if (!mModule->isVendorTagDefined()) {
ALOGI("%s: No vendor tags defined for this device.", __FUNCTION__);
return false;
}
ATRACE_BEGIN("camera3->get_metadata_vendor_tag_ops");
- mModule->get_vendor_tag_ops(&vOps);
+ mModule->getVendorTagOps(&vOps);
ATRACE_END();
// Ensure all vendor operations are present
@@ -467,54 +609,90 @@
return true;
}
-status_t CameraService::initializeShimMetadata(int cameraId) {
- int pid = getCallingPid();
- int uid = getCallingUid();
- status_t ret = validateConnect(cameraId, uid);
- if (ret != OK) {
- // Error already logged by callee
- return ret;
+status_t CameraService::makeClient(const sp<CameraService>& cameraService,
+ const sp<IInterface>& cameraCb, const String16& packageName, const String8& cameraId,
+ int facing, int clientPid, uid_t clientUid, int servicePid, bool legacyMode,
+ int halVersion, int deviceVersion, apiLevel effectiveApiLevel,
+ /*out*/sp<BasicClient>* client) {
+
+ // TODO: Update CameraClients + HAL interface to use strings for Camera IDs
+ int id = cameraIdToInt(cameraId);
+ if (id == -1) {
+ ALOGE("%s: Invalid camera ID %s, cannot convert to integer.", __FUNCTION__,
+ cameraId.string());
+ return BAD_VALUE;
}
- bool needsNewClient = false;
- sp<Client> client;
+ if (halVersion < 0 || halVersion == deviceVersion) {
+ // Default path: HAL version is unspecified by caller, create CameraClient
+ // based on device version reported by the HAL.
+ switch(deviceVersion) {
+ case CAMERA_DEVICE_API_VERSION_1_0:
+ if (effectiveApiLevel == API_1) { // Camera1 API route
+ sp<ICameraClient> tmp = static_cast<ICameraClient*>(cameraCb.get());
+ *client = new CameraClient(cameraService, tmp, packageName, id, facing,
+ clientPid, clientUid, getpid(), legacyMode);
+ } else { // Camera2 API route
+ ALOGW("Camera using old HAL version: %d", deviceVersion);
+ return -EOPNOTSUPP;
+ }
+ break;
+ case CAMERA_DEVICE_API_VERSION_2_0:
+ case CAMERA_DEVICE_API_VERSION_2_1:
+ case CAMERA_DEVICE_API_VERSION_3_0:
+ case CAMERA_DEVICE_API_VERSION_3_1:
+ case CAMERA_DEVICE_API_VERSION_3_2:
+ case CAMERA_DEVICE_API_VERSION_3_3:
+ if (effectiveApiLevel == API_1) { // Camera1 API route
+ sp<ICameraClient> tmp = static_cast<ICameraClient*>(cameraCb.get());
+ *client = new Camera2Client(cameraService, tmp, packageName, id, facing,
+ clientPid, clientUid, servicePid, legacyMode);
+ } else { // Camera2 API route
+ sp<ICameraDeviceCallbacks> tmp =
+ static_cast<ICameraDeviceCallbacks*>(cameraCb.get());
+ *client = new CameraDeviceClient(cameraService, tmp, packageName, id,
+ facing, clientPid, clientUid, servicePid);
+ }
+ break;
+ default:
+ // Should not be reachable
+ ALOGE("Unknown camera device HAL version: %d", deviceVersion);
+ return INVALID_OPERATION;
+ }
+ } else {
+ // A particular HAL version is requested by caller. Create CameraClient
+ // based on the requested HAL version.
+ if (deviceVersion > CAMERA_DEVICE_API_VERSION_1_0 &&
+ halVersion == CAMERA_DEVICE_API_VERSION_1_0) {
+ // Only support higher HAL version device opened as HAL1.0 device.
+ sp<ICameraClient> tmp = static_cast<ICameraClient*>(cameraCb.get());
+ *client = new CameraClient(cameraService, tmp, packageName, id, facing,
+ clientPid, clientUid, servicePid, legacyMode);
+ } else {
+ // Other combinations (e.g. HAL3.x open as HAL2.x) are not supported yet.
+ ALOGE("Invalid camera HAL version %x: HAL %x device can only be"
+ " opened as HAL %x device", halVersion, deviceVersion,
+ CAMERA_DEVICE_API_VERSION_1_0);
+ return INVALID_OPERATION;
+ }
+ }
+ return NO_ERROR;
+}
+
+status_t CameraService::initializeShimMetadata(int cameraId) {
+ int uid = getCallingUid();
String16 internalPackageName("media");
- { // Scope for service lock
- Mutex::Autolock lock(mServiceLock);
- if (mClient[cameraId] != NULL) {
- client = static_cast<Client*>(mClient[cameraId].promote().get());
- }
- if (client == NULL) {
- needsNewClient = true;
- ret = connectHelperLocked(/*out*/client,
- /*cameraClient*/NULL, // Empty binder callbacks
- cameraId,
- internalPackageName,
- uid,
- pid);
-
- if (ret != OK) {
- // Error already logged by callee
- return ret;
- }
- }
-
- if (client == NULL) {
- ALOGE("%s: Could not connect to client camera device.", __FUNCTION__);
- return BAD_VALUE;
- }
-
- String8 rawParams = client->getParameters();
- CameraParameters params(rawParams);
- mShimParams.add(cameraId, params);
+ String8 id = String8::format("%d", cameraId);
+ status_t ret = NO_ERROR;
+ sp<Client> tmp = nullptr;
+ if ((ret = connectHelper<ICameraClient,Client>(sp<ICameraClient>{nullptr}, id,
+ static_cast<int>(CAMERA_HAL_API_VERSION_UNSPECIFIED), internalPackageName, uid, API_1,
+ false, true, tmp)) != NO_ERROR) {
+ ALOGE("%s: Error %d (%s) initializing shim metadata.", __FUNCTION__, ret, strerror(ret));
+ return ret;
}
-
- // Close client if one was opened solely for this call
- if (needsNewClient) {
- client->disconnect();
- }
- return OK;
+ return NO_ERROR;
}
status_t CameraService::getLegacyParametersLazy(int cameraId,
@@ -530,42 +708,55 @@
return BAD_VALUE;
}
- ssize_t index = -1;
- { // Scope for service lock
+ String8 id = String8::format("%d", cameraId);
+
+ // Check if we already have parameters
+ {
+ // Scope for service lock
Mutex::Autolock lock(mServiceLock);
- index = mShimParams.indexOfKey(cameraId);
- // Release service lock so initializeShimMetadata can be called correctly.
-
- if (index >= 0) {
- *parameters = mShimParams[index];
+ auto cameraState = getCameraState(id);
+ if (cameraState == nullptr) {
+ ALOGE("%s: Invalid camera ID: %s", __FUNCTION__, id.string());
+ return BAD_VALUE;
+ }
+ CameraParameters p = cameraState->getShimParams();
+ if (!p.isEmpty()) {
+ *parameters = p;
+ return NO_ERROR;
}
}
- if (index < 0) {
- int64_t token = IPCThreadState::self()->clearCallingIdentity();
- ret = initializeShimMetadata(cameraId);
- IPCThreadState::self()->restoreCallingIdentity(token);
- if (ret != OK) {
- // Error already logged by callee
- return ret;
+ int64_t token = IPCThreadState::self()->clearCallingIdentity();
+ ret = initializeShimMetadata(cameraId);
+ IPCThreadState::self()->restoreCallingIdentity(token);
+ if (ret != NO_ERROR) {
+ // Error already logged by callee
+ return ret;
+ }
+
+ // Check for parameters again
+ {
+ // Scope for service lock
+ Mutex::Autolock lock(mServiceLock);
+ auto cameraState = getCameraState(id);
+ if (cameraState == nullptr) {
+ ALOGE("%s: Invalid camera ID: %s", __FUNCTION__, id.string());
+ return BAD_VALUE;
}
-
- { // Scope for service lock
- Mutex::Autolock lock(mServiceLock);
- index = mShimParams.indexOfKey(cameraId);
-
- LOG_ALWAYS_FATAL_IF(index < 0, "index should have been initialized");
-
- *parameters = mShimParams[index];
+ CameraParameters p = cameraState->getShimParams();
+ if (!p.isEmpty()) {
+ *parameters = p;
+ return NO_ERROR;
}
}
- return OK;
+ ALOGE("%s: Parameters were not initialized, or were empty. Device may not be present.",
+ __FUNCTION__);
+ return INVALID_OPERATION;
}
-status_t CameraService::validateConnect(int cameraId,
- /*inout*/
- int& clientUid) const {
+status_t CameraService::validateConnectLocked(const String8& cameraId, /*inout*/int& clientUid)
+ const {
int callingPid = getCallingPid();
@@ -574,160 +765,251 @@
} else {
// We only trust our own process to forward client UIDs
if (callingPid != getpid()) {
- ALOGE("CameraService::connect X (pid %d) rejected (don't trust clientUid)",
+ ALOGE("CameraService::connect X (PID %d) rejected (don't trust clientUid)",
callingPid);
return PERMISSION_DENIED;
}
}
if (!mModule) {
- ALOGE("Camera HAL module not loaded");
+ ALOGE("CameraService::connect X (PID %d) rejected (camera HAL module not loaded)",
+ callingPid);
return -ENODEV;
}
- if (cameraId < 0 || cameraId >= mNumberOfCameras) {
- ALOGE("CameraService::connect X (pid %d) rejected (invalid cameraId %d).",
- callingPid, cameraId);
+ if (getCameraState(cameraId) == nullptr) {
+ ALOGE("CameraService::connect X (PID %d) rejected (invalid camera ID %s)", callingPid,
+ cameraId.string());
return -ENODEV;
}
+ // Check device policy for this camera
char value[PROPERTY_VALUE_MAX];
- property_get("sys.secpolicy.camera.disabled", value, "0");
+ char key[PROPERTY_KEY_MAX];
+ int clientUserId = multiuser_get_user_id(clientUid);
+ snprintf(key, PROPERTY_KEY_MAX, "sys.secpolicy.camera.off_%d", clientUserId);
+ property_get(key, value, "0");
if (strcmp(value, "1") == 0) {
// Camera is disabled by DevicePolicyManager.
- ALOGI("Camera is disabled. connect X (pid %d) rejected", callingPid);
+ ALOGE("CameraService::connect X (PID %d) rejected (camera %s is disabled by device "
+ "policy)", callingPid, cameraId.string());
return -EACCES;
}
- ICameraServiceListener::Status currentStatus = getStatus(cameraId);
+ // Only allow clients who are being used by the current foreground device user.
+ if (mLastUserId != clientUserId && mLastUserId != DEFAULT_LAST_USER_ID) {
+ ALOGE("CameraService::connect X (PID %d) rejected (cannot connect from non-foreground "
+ "device user)", callingPid);
+ return PERMISSION_DENIED;
+ }
+
+ return checkIfDeviceIsUsable(cameraId);
+}
+
+status_t CameraService::checkIfDeviceIsUsable(const String8& cameraId) const {
+ auto cameraState = getCameraState(cameraId);
+ int callingPid = getCallingPid();
+ if (cameraState == nullptr) {
+ ALOGE("CameraService::connect X (PID %d) rejected (invalid camera ID %s)", callingPid,
+ cameraId.string());
+ return -ENODEV;
+ }
+
+ ICameraServiceListener::Status currentStatus = cameraState->getStatus();
if (currentStatus == ICameraServiceListener::STATUS_NOT_PRESENT) {
- ALOGI("Camera is not plugged in,"
- " connect X (pid %d) rejected", callingPid);
+ ALOGE("CameraService::connect X (PID %d) rejected (camera %s is not connected)",
+ callingPid, cameraId.string());
return -ENODEV;
} else if (currentStatus == ICameraServiceListener::STATUS_ENUMERATING) {
- ALOGI("Camera is enumerating,"
- " connect X (pid %d) rejected", callingPid);
+ ALOGE("CameraService::connect X (PID %d) rejected, (camera %s is initializing)",
+ callingPid, cameraId.string());
return -EBUSY;
}
- // Else don't check for STATUS_NOT_AVAILABLE.
- // -- It's done implicitly in canConnectUnsafe /w the mBusy array
- return OK;
+ return NO_ERROR;
}
-bool CameraService::canConnectUnsafe(int cameraId,
- const String16& clientPackageName,
- const sp<IBinder>& remoteCallback,
- sp<BasicClient> &client) {
- String8 clientName8(clientPackageName);
- int callingPid = getCallingPid();
+void CameraService::finishConnectLocked(const sp<BasicClient>& client,
+ const CameraService::DescriptorPtr& desc) {
- if (mClient[cameraId] != 0) {
- client = mClient[cameraId].promote();
- if (client != 0) {
- if (remoteCallback == client->getRemote()) {
- LOG1("CameraService::connect X (pid %d) (the same client)",
- callingPid);
- return true;
- } else {
- // TODOSC: need to support 1 regular client,
- // multiple shared clients here
- ALOGW("CameraService::connect X (pid %d) rejected"
- " (existing client).", callingPid);
- return false;
+ // Make a descriptor for the incoming client
+ auto clientDescriptor = CameraService::CameraClientManager::makeClientDescriptor(client, desc);
+ auto evicted = mActiveClientManager.addAndEvict(clientDescriptor);
+
+ logConnected(desc->getKey(), static_cast<int>(desc->getOwnerId()),
+ String8(client->getPackageName()));
+
+ if (evicted.size() > 0) {
+ // This should never happen - clients should already have been removed in disconnect
+ for (auto& i : evicted) {
+ ALOGE("%s: Invalid state: Client for camera %s was not removed in disconnect",
+ __FUNCTION__, i->getKey().string());
+ }
+
+ LOG_ALWAYS_FATAL("%s: Invalid state for CameraService, clients not evicted properly",
+ __FUNCTION__);
+ }
+}
+
+status_t CameraService::handleEvictionsLocked(const String8& cameraId, int clientPid,
+ apiLevel effectiveApiLevel, const sp<IBinder>& remoteCallback, const String8& packageName,
+ /*out*/
+ sp<BasicClient>* client,
+ std::shared_ptr<resource_policy::ClientDescriptor<String8, sp<BasicClient>>>* partial) {
+
+ status_t ret = NO_ERROR;
+ std::vector<sp<BasicClient>> evictedClients;
+ DescriptorPtr clientDescriptor;
+ {
+ if (effectiveApiLevel == API_1) {
+ // If we are using API1, any existing client for this camera ID with the same remote
+ // should be returned rather than evicted to allow MediaRecorder to work properly.
+
+ auto current = mActiveClientManager.get(cameraId);
+ if (current != nullptr) {
+ auto clientSp = current->getValue();
+ if (clientSp.get() != nullptr) { // should never be needed
+ if (clientSp->getRemote() == remoteCallback) {
+ ALOGI("CameraService::connect X (PID %d) (second call from same"
+ "app binder, returning the same client)", clientPid);
+ *client = clientSp;
+ return NO_ERROR;
+ }
+ }
}
}
- mClient[cameraId].clear();
- }
- /*
- mBusy is set to false as the last step of the Client destructor,
- after which it is guaranteed that the Client destructor has finished (
- including any inherited destructors)
+ // Return error if the device was unplugged or removed by the HAL for some reason
+ if ((ret = checkIfDeviceIsUsable(cameraId)) != NO_ERROR) {
+ return ret;
+ }
- We only need this for a Client subclasses since we don't allow
- multiple Clents to be opened concurrently, but multiple BasicClient
- would be fine
- */
- if (mBusy[cameraId]) {
- ALOGW("CameraService::connect X (pid %d, \"%s\") rejected"
- " (camera %d is still busy).", callingPid,
- clientName8.string(), cameraId);
- return false;
- }
+ // Get current active client PIDs
+ std::vector<int> ownerPids(mActiveClientManager.getAllOwners());
+ ownerPids.push_back(clientPid);
- return true;
-}
+ // Use the value +PROCESS_STATE_NONEXISTENT, to avoid taking
+ // address of PROCESS_STATE_NONEXISTENT as a reference argument
+ // for the vector constructor. PROCESS_STATE_NONEXISTENT does
+ // not have an out-of-class definition.
+ std::vector<int> priorities(ownerPids.size(), +PROCESS_STATE_NONEXISTENT);
-status_t CameraService::connectHelperLocked(
- /*out*/
- sp<Client>& client,
- /*in*/
- const sp<ICameraClient>& cameraClient,
- int cameraId,
- const String16& clientPackageName,
- int clientUid,
- int callingPid,
- int halVersion,
- bool legacyMode) {
+ // Get priorites of all active PIDs
+ ProcessInfoService::getProcessStatesFromPids(ownerPids.size(), &ownerPids[0],
+ /*out*/&priorities[0]);
- int facing = -1;
- int deviceVersion = getDeviceVersion(cameraId, &facing);
+ // Update all active clients' priorities
+ std::map<int,int> pidToPriorityMap;
+ for (size_t i = 0; i < ownerPids.size() - 1; i++) {
+ pidToPriorityMap.emplace(ownerPids[i], getCameraPriorityFromProcState(priorities[i]));
+ }
+ mActiveClientManager.updatePriorities(pidToPriorityMap);
- if (halVersion < 0 || halVersion == deviceVersion) {
- // Default path: HAL version is unspecified by caller, create CameraClient
- // based on device version reported by the HAL.
- switch(deviceVersion) {
- case CAMERA_DEVICE_API_VERSION_1_0:
- client = new CameraClient(this, cameraClient,
- clientPackageName, cameraId,
- facing, callingPid, clientUid, getpid(), legacyMode);
- break;
- case CAMERA_DEVICE_API_VERSION_2_0:
- case CAMERA_DEVICE_API_VERSION_2_1:
- case CAMERA_DEVICE_API_VERSION_3_0:
- case CAMERA_DEVICE_API_VERSION_3_1:
- case CAMERA_DEVICE_API_VERSION_3_2:
- client = new Camera2Client(this, cameraClient,
- clientPackageName, cameraId,
- facing, callingPid, clientUid, getpid(), legacyMode);
- break;
- case -1:
- ALOGE("Invalid camera id %d", cameraId);
+ // Get state for the given cameraId
+ auto state = getCameraState(cameraId);
+ if (state == nullptr) {
+ ALOGE("CameraService::connect X (PID %d) rejected (no camera device with ID %s)",
+ clientPid, cameraId.string());
return BAD_VALUE;
- default:
- ALOGE("Unknown camera device HAL version: %d", deviceVersion);
- return INVALID_OPERATION;
}
- } else {
- // A particular HAL version is requested by caller. Create CameraClient
- // based on the requested HAL version.
- if (deviceVersion > CAMERA_DEVICE_API_VERSION_1_0 &&
- halVersion == CAMERA_DEVICE_API_VERSION_1_0) {
- // Only support higher HAL version device opened as HAL1.0 device.
- client = new CameraClient(this, cameraClient,
- clientPackageName, cameraId,
- facing, callingPid, clientUid, getpid(), legacyMode);
- } else {
- // Other combinations (e.g. HAL3.x open as HAL2.x) are not supported yet.
- ALOGE("Invalid camera HAL version %x: HAL %x device can only be"
- " opened as HAL %x device", halVersion, deviceVersion,
- CAMERA_DEVICE_API_VERSION_1_0);
- return INVALID_OPERATION;
+
+ // Make descriptor for incoming client
+ clientDescriptor = CameraClientManager::makeClientDescriptor(cameraId,
+ sp<BasicClient>{nullptr}, static_cast<int32_t>(state->getCost()),
+ state->getConflicting(),
+ getCameraPriorityFromProcState(priorities[priorities.size() - 1]), clientPid);
+
+ // Find clients that would be evicted
+ auto evicted = mActiveClientManager.wouldEvict(clientDescriptor);
+
+ // If the incoming client was 'evicted,' higher priority clients have the camera in the
+ // background, so we cannot do evictions
+ if (std::find(evicted.begin(), evicted.end(), clientDescriptor) != evicted.end()) {
+ ALOGE("CameraService::connect X (PID %d) rejected (existing client(s) with higher"
+ " priority).", clientPid);
+
+ sp<BasicClient> clientSp = clientDescriptor->getValue();
+ String8 curTime = getFormattedCurrentTime();
+ auto incompatibleClients =
+ mActiveClientManager.getIncompatibleClients(clientDescriptor);
+
+ String8 msg = String8::format("%s : DENIED connect device %s client for package %s "
+ "(PID %d, priority %d)", curTime.string(),
+ cameraId.string(), packageName.string(), clientPid,
+ getCameraPriorityFromProcState(priorities[priorities.size() - 1]));
+
+ for (auto& i : incompatibleClients) {
+ msg.appendFormat("\n - Blocked by existing device %s client for package %s"
+ "(PID %" PRId32 ", priority %" PRId32 ")", i->getKey().string(),
+ String8{i->getValue()->getPackageName()}.string(), i->getOwnerId(),
+ i->getPriority());
+ }
+
+ // Log the client's attempt
+ mEventLog.add(msg);
+
+ return -EBUSY;
+ }
+
+ for (auto& i : evicted) {
+ sp<BasicClient> clientSp = i->getValue();
+ if (clientSp.get() == nullptr) {
+ ALOGE("%s: Invalid state: Null client in active client list.", __FUNCTION__);
+
+ // TODO: Remove this
+ LOG_ALWAYS_FATAL("%s: Invalid state for CameraService, null client in active list",
+ __FUNCTION__);
+ mActiveClientManager.remove(i);
+ continue;
+ }
+
+ ALOGE("CameraService::connect evicting conflicting client for camera ID %s",
+ i->getKey().string());
+ evictedClients.push_back(clientSp);
+
+ String8 curTime = getFormattedCurrentTime();
+
+ // Log the clients evicted
+ mEventLog.add(String8::format("%s : EVICT device %s client for package %s (PID %"
+ PRId32 ", priority %" PRId32 ")\n - Evicted by device %s client for "
+ "package %s (PID %d, priority %" PRId32 ")", curTime.string(),
+ i->getKey().string(), String8{clientSp->getPackageName()}.string(),
+ i->getOwnerId(), i->getPriority(), cameraId.string(),
+ packageName.string(), clientPid,
+ getCameraPriorityFromProcState(priorities[priorities.size() - 1])));
+
+ // Notify the client of disconnection
+ clientSp->notifyError(ICameraDeviceCallbacks::ERROR_CAMERA_DISCONNECTED,
+ CaptureResultExtras());
}
}
- status_t status = connectFinishUnsafe(client, client->getRemote());
- if (status != OK) {
- // this is probably not recoverable.. maybe the client can try again
- return status;
+ // Do not hold mServiceLock while disconnecting clients, but retain the condition blocking
+ // other clients from connecting in mServiceLockWrapper if held
+ mServiceLock.unlock();
+
+ // Clear caller identity temporarily so client disconnect PID checks work correctly
+ int64_t token = IPCThreadState::self()->clearCallingIdentity();
+
+ // Destroy evicted clients
+ for (auto& i : evictedClients) {
+ // Disconnect is blocking, and should only have returned when HAL has cleaned up
+ i->disconnect(); // Clients will remove themselves from the active client list here
+ }
+ evictedClients.clear();
+
+ IPCThreadState::self()->restoreCallingIdentity(token);
+
+ // Once clients have been disconnected, relock
+ mServiceLock.lock();
+
+ // Check again if the device was unplugged or something while we weren't holding mServiceLock
+ if ((ret = checkIfDeviceIsUsable(cameraId)) != NO_ERROR) {
+ return ret;
}
- mClient[cameraId] = client;
- LOG1("CameraService::connect X (id %d, this pid is %d)", cameraId,
- getpid());
-
- return OK;
+ *partial = clientDescriptor;
+ return NO_ERROR;
}
status_t CameraService::connect(
@@ -738,47 +1020,18 @@
/*out*/
sp<ICamera>& device) {
- String8 clientName8(clientPackageName);
- int callingPid = getCallingPid();
+ status_t ret = NO_ERROR;
+ String8 id = String8::format("%d", cameraId);
+ sp<Client> client = nullptr;
+ ret = connectHelper<ICameraClient,Client>(cameraClient, id, CAMERA_HAL_API_VERSION_UNSPECIFIED,
+ clientPackageName, clientUid, API_1, false, false, /*out*/client);
- LOG1("CameraService::connect E (pid %d \"%s\", id %d)", callingPid,
- clientName8.string(), cameraId);
-
- status_t status = validateConnect(cameraId, /*inout*/clientUid);
- if (status != OK) {
- return status;
+ if(ret != NO_ERROR) {
+ return ret;
}
-
- sp<Client> client;
- {
- Mutex::Autolock lock(mServiceLock);
- sp<BasicClient> clientTmp;
- if (!canConnectUnsafe(cameraId, clientPackageName,
- IInterface::asBinder(cameraClient),
- /*out*/clientTmp)) {
- return -EBUSY;
- } else if (client.get() != NULL) {
- device = static_cast<Client*>(clientTmp.get());
- return OK;
- }
-
- status = connectHelperLocked(/*out*/client,
- cameraClient,
- cameraId,
- clientPackageName,
- clientUid,
- callingPid);
- if (status != OK) {
- return status;
- }
-
- }
- // important: release the mutex here so the client can call back
- // into the service from its destructor (can be at the end of the call)
-
device = client;
- return OK;
+ return NO_ERROR;
}
status_t CameraService::connectLegacy(
@@ -789,8 +1042,9 @@
/*out*/
sp<ICamera>& device) {
+ int apiVersion = mModule->getModuleApiVersion();
if (halVersion != CAMERA_HAL_API_VERSION_UNSPECIFIED &&
- mModule->common.module_api_version < CAMERA_MODULE_API_VERSION_2_3) {
+ apiVersion < CAMERA_MODULE_API_VERSION_2_3) {
/*
* Either the HAL version is unspecified in which case this just creates
* a camera client selected by the latest device version, or
@@ -798,142 +1052,22 @@
* the open_legacy call
*/
ALOGE("%s: camera HAL module version %x doesn't support connecting to legacy HAL devices!",
- __FUNCTION__, mModule->common.module_api_version);
+ __FUNCTION__, apiVersion);
return INVALID_OPERATION;
}
- String8 clientName8(clientPackageName);
- int callingPid = getCallingPid();
+ status_t ret = NO_ERROR;
+ String8 id = String8::format("%d", cameraId);
+ sp<Client> client = nullptr;
+ ret = connectHelper<ICameraClient,Client>(cameraClient, id, halVersion, clientPackageName,
+ clientUid, API_1, true, false, /*out*/client);
- LOG1("CameraService::connect legacy E (pid %d \"%s\", id %d)", callingPid,
- clientName8.string(), cameraId);
-
- status_t status = validateConnect(cameraId, /*inout*/clientUid);
- if (status != OK) {
- return status;
+ if(ret != NO_ERROR) {
+ return ret;
}
- sp<Client> client;
- {
- Mutex::Autolock lock(mServiceLock);
- sp<BasicClient> clientTmp;
- if (!canConnectUnsafe(cameraId, clientPackageName,
- IInterface::asBinder(cameraClient),
- /*out*/clientTmp)) {
- return -EBUSY;
- } else if (client.get() != NULL) {
- device = static_cast<Client*>(clientTmp.get());
- return OK;
- }
-
- status = connectHelperLocked(/*out*/client,
- cameraClient,
- cameraId,
- clientPackageName,
- clientUid,
- callingPid,
- halVersion,
- /*legacyMode*/true);
- if (status != OK) {
- return status;
- }
-
- }
- // important: release the mutex here so the client can call back
- // into the service from its destructor (can be at the end of the call)
-
device = client;
- return OK;
-}
-
-status_t CameraService::connectFinishUnsafe(const sp<BasicClient>& client,
- const sp<IBinder>& remoteCallback) {
- status_t status = client->initialize(mModule);
- if (status != OK) {
- ALOGE("%s: Could not initialize client from HAL module.", __FUNCTION__);
- return status;
- }
- if (remoteCallback != NULL) {
- remoteCallback->linkToDeath(this);
- }
-
- return OK;
-}
-
-status_t CameraService::connectPro(
- const sp<IProCameraCallbacks>& cameraCb,
- int cameraId,
- const String16& clientPackageName,
- int clientUid,
- /*out*/
- sp<IProCameraUser>& device)
-{
- if (cameraCb == 0) {
- ALOGE("%s: Callback must not be null", __FUNCTION__);
- return BAD_VALUE;
- }
-
- String8 clientName8(clientPackageName);
- int callingPid = getCallingPid();
-
- LOG1("CameraService::connectPro E (pid %d \"%s\", id %d)", callingPid,
- clientName8.string(), cameraId);
- status_t status = validateConnect(cameraId, /*inout*/clientUid);
- if (status != OK) {
- return status;
- }
-
- sp<ProClient> client;
- {
- Mutex::Autolock lock(mServiceLock);
- {
- sp<BasicClient> client;
- if (!canConnectUnsafe(cameraId, clientPackageName,
- IInterface::asBinder(cameraCb),
- /*out*/client)) {
- return -EBUSY;
- }
- }
-
- int facing = -1;
- int deviceVersion = getDeviceVersion(cameraId, &facing);
-
- switch(deviceVersion) {
- case CAMERA_DEVICE_API_VERSION_1_0:
- ALOGE("Camera id %d uses HALv1, doesn't support ProCamera",
- cameraId);
- return -EOPNOTSUPP;
- break;
- case CAMERA_DEVICE_API_VERSION_2_0:
- case CAMERA_DEVICE_API_VERSION_2_1:
- case CAMERA_DEVICE_API_VERSION_3_0:
- case CAMERA_DEVICE_API_VERSION_3_1:
- case CAMERA_DEVICE_API_VERSION_3_2:
- client = new ProCamera2Client(this, cameraCb, clientPackageName,
- cameraId, facing, callingPid, clientUid, getpid());
- break;
- case -1:
- ALOGE("Invalid camera id %d", cameraId);
- return BAD_VALUE;
- default:
- ALOGE("Unknown camera device HAL version: %d", deviceVersion);
- return INVALID_OPERATION;
- }
-
- status_t status = connectFinishUnsafe(client, client->getRemote());
- if (status != OK) {
- return status;
- }
-
- mProClientList[cameraId].push(client);
-
- LOG1("CameraService::connectPro X (id %d, this pid is %d)", cameraId,
- getpid());
- }
- // important: release the mutex here so the client can call back
- // into the service from its destructor (can be at the end of the call)
- device = client;
- return OK;
+ return NO_ERROR;
}
status_t CameraService::connectDevice(
@@ -942,74 +1076,113 @@
const String16& clientPackageName,
int clientUid,
/*out*/
- sp<ICameraDeviceUser>& device)
-{
+ sp<ICameraDeviceUser>& device) {
- String8 clientName8(clientPackageName);
- int callingPid = getCallingPid();
+ status_t ret = NO_ERROR;
+ String8 id = String8::format("%d", cameraId);
+ sp<CameraDeviceClient> client = nullptr;
+ ret = connectHelper<ICameraDeviceCallbacks,CameraDeviceClient>(cameraCb, id,
+ CAMERA_HAL_API_VERSION_UNSPECIFIED, clientPackageName, clientUid, API_2, false, false,
+ /*out*/client);
- LOG1("CameraService::connectDevice E (pid %d \"%s\", id %d)", callingPid,
- clientName8.string(), cameraId);
-
- status_t status = validateConnect(cameraId, /*inout*/clientUid);
- if (status != OK) {
- return status;
+ if(ret != NO_ERROR) {
+ return ret;
}
- sp<CameraDeviceClient> client;
- {
- Mutex::Autolock lock(mServiceLock);
- {
- sp<BasicClient> client;
- if (!canConnectUnsafe(cameraId, clientPackageName,
- IInterface::asBinder(cameraCb),
- /*out*/client)) {
- return -EBUSY;
- }
- }
-
- int facing = -1;
- int deviceVersion = getDeviceVersion(cameraId, &facing);
-
- switch(deviceVersion) {
- case CAMERA_DEVICE_API_VERSION_1_0:
- ALOGW("Camera using old HAL version: %d", deviceVersion);
- return -EOPNOTSUPP;
- // TODO: don't allow 2.0 Only allow 2.1 and higher
- case CAMERA_DEVICE_API_VERSION_2_0:
- case CAMERA_DEVICE_API_VERSION_2_1:
- case CAMERA_DEVICE_API_VERSION_3_0:
- case CAMERA_DEVICE_API_VERSION_3_1:
- case CAMERA_DEVICE_API_VERSION_3_2:
- client = new CameraDeviceClient(this, cameraCb, clientPackageName,
- cameraId, facing, callingPid, clientUid, getpid());
- break;
- case -1:
- ALOGE("Invalid camera id %d", cameraId);
- return BAD_VALUE;
- default:
- ALOGE("Unknown camera device HAL version: %d", deviceVersion);
- return INVALID_OPERATION;
- }
-
- status_t status = connectFinishUnsafe(client, client->getRemote());
- if (status != OK) {
- // this is probably not recoverable.. maybe the client can try again
- return status;
- }
-
- LOG1("CameraService::connectDevice X (id %d, this pid is %d)", cameraId,
- getpid());
-
- mClient[cameraId] = client;
- }
- // important: release the mutex here so the client can call back
- // into the service from its destructor (can be at the end of the call)
-
device = client;
+ return NO_ERROR;
+}
+
+status_t CameraService::setTorchMode(const String16& cameraId, bool enabled,
+ const sp<IBinder>& clientBinder) {
+ if (enabled && clientBinder == NULL) {
+ ALOGE("%s: torch client binder is NULL", __FUNCTION__);
+ return -EINVAL;
+ }
+
+ String8 id = String8(cameraId.string());
+
+ // verify id is valid.
+ auto state = getCameraState(id);
+ if (state == nullptr) {
+ ALOGE("%s: camera id is invalid %s", id.string());
+ return -EINVAL;
+ }
+
+ ICameraServiceListener::Status cameraStatus = state->getStatus();
+ if (cameraStatus != ICameraServiceListener::STATUS_PRESENT &&
+ cameraStatus != ICameraServiceListener::STATUS_NOT_AVAILABLE) {
+ ALOGE("%s: camera id is invalid %s", id.string());
+ return -EINVAL;
+ }
+
+ {
+ Mutex::Autolock al(mTorchStatusMutex);
+ ICameraServiceListener::TorchStatus status;
+ status_t res = getTorchStatusLocked(id, &status);
+ if (res) {
+ ALOGE("%s: getting current torch status failed for camera %s",
+ __FUNCTION__, id.string());
+ return -EINVAL;
+ }
+
+ if (status == ICameraServiceListener::TORCH_STATUS_NOT_AVAILABLE) {
+ if (cameraStatus == ICameraServiceListener::STATUS_NOT_AVAILABLE) {
+ ALOGE("%s: torch mode of camera %s is not available because "
+ "camera is in use", __FUNCTION__, id.string());
+ return -EBUSY;
+ } else {
+ ALOGE("%s: torch mode of camera %s is not available due to "
+ "insufficient resources", __FUNCTION__, id.string());
+ return -EUSERS;
+ }
+ }
+ }
+
+ status_t res = mFlashlight->setTorchMode(id, enabled);
+ if (res) {
+ ALOGE("%s: setting torch mode of camera %s to %d failed. %s (%d)",
+ __FUNCTION__, id.string(), enabled, strerror(-res), res);
+ return res;
+ }
+
+ {
+ // update the link to client's death
+ Mutex::Autolock al(mTorchClientMapMutex);
+ ssize_t index = mTorchClientMap.indexOfKey(id);
+ if (enabled) {
+ if (index == NAME_NOT_FOUND) {
+ mTorchClientMap.add(id, clientBinder);
+ } else {
+ const sp<IBinder> oldBinder = mTorchClientMap.valueAt(index);
+ oldBinder->unlinkToDeath(this);
+
+ mTorchClientMap.replaceValueAt(index, clientBinder);
+ }
+ clientBinder->linkToDeath(this);
+ } else if (index != NAME_NOT_FOUND) {
+ sp<IBinder> oldBinder = mTorchClientMap.valueAt(index);
+ oldBinder->unlinkToDeath(this);
+ }
+ }
+
return OK;
}
+void CameraService::notifySystemEvent(int eventId, int arg0) {
+ switch(eventId) {
+ case ICameraService::USER_SWITCHED: {
+ doUserSwitch(/*newUserId*/arg0);
+ break;
+ }
+ case ICameraService::NO_EVENT:
+ default: {
+ ALOGW("%s: Received invalid system event from system_server: %d", __FUNCTION__,
+ eventId);
+ break;
+ }
+ }
+}
status_t CameraService::addListener(
const sp<ICameraServiceListener>& listener) {
@@ -1022,30 +1195,45 @@
Mutex::Autolock lock(mServiceLock);
- Vector<sp<ICameraServiceListener> >::iterator it, end;
- for (it = mListenerList.begin(); it != mListenerList.end(); ++it) {
- if (IInterface::asBinder(*it) == IInterface::asBinder(listener)) {
- ALOGW("%s: Tried to add listener %p which was already subscribed",
- __FUNCTION__, listener.get());
- return ALREADY_EXISTS;
+ {
+ Mutex::Autolock lock(mStatusListenerLock);
+ for (auto& it : mListenerList) {
+ if (IInterface::asBinder(it) == IInterface::asBinder(listener)) {
+ ALOGW("%s: Tried to add listener %p which was already subscribed",
+ __FUNCTION__, listener.get());
+ return ALREADY_EXISTS;
+ }
}
+
+ mListenerList.push_back(listener);
}
- mListenerList.push_back(listener);
/* Immediately signal current status to this listener only */
{
- Mutex::Autolock m(mStatusMutex) ;
- int numCams = getNumberOfCameras();
- for (int i = 0; i < numCams; ++i) {
- listener->onStatusChanged(mStatusList[i], i);
+ Mutex::Autolock lock(mCameraStatesLock);
+ for (auto& i : mCameraStates) {
+ // TODO: Update binder to use String16 for camera IDs and remove;
+ int id = cameraIdToInt(i.first);
+ if (id == -1) continue;
+
+ listener->onStatusChanged(i.second->getStatus(), id);
+ }
+ }
+
+ /* Immediately signal current torch status to this listener only */
+ {
+ Mutex::Autolock al(mTorchStatusMutex);
+ for (size_t i = 0; i < mTorchStatusMap.size(); i++ ) {
+ String16 id = String16(mTorchStatusMap.keyAt(i).string());
+ listener->onTorchStatusChanged(mTorchStatusMap.valueAt(i), id);
}
}
return OK;
}
-status_t CameraService::removeListener(
- const sp<ICameraServiceListener>& listener) {
+
+status_t CameraService::removeListener(const sp<ICameraServiceListener>& listener) {
ALOGV("%s: Remove listener %p", __FUNCTION__, listener.get());
if (listener == 0) {
@@ -1055,11 +1243,13 @@
Mutex::Autolock lock(mServiceLock);
- Vector<sp<ICameraServiceListener> >::iterator it;
- for (it = mListenerList.begin(); it != mListenerList.end(); ++it) {
- if (IInterface::asBinder(*it) == IInterface::asBinder(listener)) {
- mListenerList.erase(it);
- return OK;
+ {
+ Mutex::Autolock lock(mStatusListenerLock);
+ for (auto it = mListenerList.begin(); it != mListenerList.end(); it++) {
+ if (IInterface::asBinder(*it) == IInterface::asBinder(listener)) {
+ mListenerList.erase(it);
+ return OK;
+ }
}
}
@@ -1069,10 +1259,7 @@
return BAD_VALUE;
}
-status_t CameraService::getLegacyParameters(
- int cameraId,
- /*out*/
- String16* parameters) {
+status_t CameraService::getLegacyParameters(int cameraId, /*out*/String16* parameters) {
ALOGV("%s: for camera ID = %d", __FUNCTION__, cameraId);
if (parameters == NULL) {
@@ -1127,6 +1314,7 @@
return OK;
}
case CAMERA_DEVICE_API_VERSION_3_2:
+ case CAMERA_DEVICE_API_VERSION_3_3:
ALOGV("%s: Camera id %d uses HAL3.2 or newer, supports api1/api2 directly",
__FUNCTION__, cameraId);
return OK;
@@ -1141,140 +1329,177 @@
return OK;
}
-void CameraService::removeClientByRemote(const wp<IBinder>& remoteBinder) {
- int callingPid = getCallingPid();
- LOG1("CameraService::removeClientByRemote E (pid %d)", callingPid);
-
- // Declare this before the lock to make absolutely sure the
- // destructor won't be called with the lock held.
+void CameraService::removeByClient(const BasicClient* client) {
Mutex::Autolock lock(mServiceLock);
-
- int outIndex;
- sp<BasicClient> client = findClientUnsafe(remoteBinder, outIndex);
-
- if (client != 0) {
- // Found our camera, clear and leave.
- LOG1("removeClient: clear camera %d", outIndex);
-
- sp<IBinder> remote = client->getRemote();
- if (remote != NULL) {
- remote->unlinkToDeath(this);
- }
-
- mClient[outIndex].clear();
- } else {
-
- sp<ProClient> clientPro = findProClientUnsafe(remoteBinder);
-
- if (clientPro != NULL) {
- // Found our camera, clear and leave.
- LOG1("removeClient: clear pro %p", clientPro.get());
-
- IInterface::asBinder(clientPro->getRemoteCallback())->unlinkToDeath(this);
+ for (auto& i : mActiveClientManager.getAll()) {
+ auto clientSp = i->getValue();
+ if (clientSp.get() == client) {
+ mActiveClientManager.remove(i);
}
}
-
- LOG1("CameraService::removeClientByRemote X (pid %d)", callingPid);
}
-sp<CameraService::ProClient> CameraService::findProClientUnsafe(
- const wp<IBinder>& cameraCallbacksRemote)
-{
- sp<ProClient> clientPro;
+bool CameraService::evictClientIdByRemote(const wp<IBinder>& remote) {
+ const int callingPid = getCallingPid();
+ const int servicePid = getpid();
+ bool ret = false;
+ {
+ // Acquire mServiceLock and prevent other clients from connecting
+ std::unique_ptr<AutoConditionLock> lock =
+ AutoConditionLock::waitAndAcquire(mServiceLockWrapper);
- for (int i = 0; i < mNumberOfCameras; ++i) {
- Vector<size_t> removeIdx;
- for (size_t j = 0; j < mProClientList[i].size(); ++j) {
- wp<ProClient> cl = mProClientList[i][j];
+ std::vector<sp<BasicClient>> evicted;
+ for (auto& i : mActiveClientManager.getAll()) {
+ auto clientSp = i->getValue();
+ if (clientSp.get() == nullptr) {
+ ALOGE("%s: Dead client still in mActiveClientManager.", __FUNCTION__);
+ mActiveClientManager.remove(i);
+ continue;
+ }
+ if (remote == clientSp->getRemote() && (callingPid == servicePid ||
+ callingPid == clientSp->getClientPid())) {
+ mActiveClientManager.remove(i);
+ evicted.push_back(clientSp);
- sp<ProClient> clStrong = cl.promote();
- if (clStrong != NULL && clStrong->getRemote() == cameraCallbacksRemote) {
- clientPro = clStrong;
- break;
- } else if (clStrong == NULL) {
- // mark to clean up dead ptr
- removeIdx.push(j);
+ // Notify the client of disconnection
+ clientSp->notifyError(ICameraDeviceCallbacks::ERROR_CAMERA_DISCONNECTED,
+ CaptureResultExtras());
}
}
- // remove stale ptrs (in reverse so the indices dont change)
- for (ssize_t j = (ssize_t)removeIdx.size() - 1; j >= 0; --j) {
- mProClientList[i].removeAt(removeIdx[j]);
+ // Do not hold mServiceLock while disconnecting clients, but retain the condition blocking
+ // other clients from connecting in mServiceLockWrapper if held
+ mServiceLock.unlock();
+
+ // Do not clear caller identity, remote caller should be client proccess
+
+ for (auto& i : evicted) {
+ if (i.get() != nullptr) {
+ i->disconnect();
+ ret = true;
+ }
}
- }
+ // Reacquire mServiceLock
+ mServiceLock.lock();
- return clientPro;
+ } // lock is destroyed, allow further connect calls
+
+ return ret;
}
-sp<CameraService::BasicClient> CameraService::findClientUnsafe(
- const wp<IBinder>& cameraClient, int& outIndex) {
- sp<BasicClient> client;
- for (int i = 0; i < mNumberOfCameras; i++) {
+std::shared_ptr<CameraService::CameraState> CameraService::getCameraState(
+ const String8& cameraId) const {
+ std::shared_ptr<CameraState> state;
+ {
+ Mutex::Autolock lock(mCameraStatesLock);
+ auto iter = mCameraStates.find(cameraId);
+ if (iter != mCameraStates.end()) {
+ state = iter->second;
+ }
+ }
+ return state;
+}
- // This happens when we have already disconnected (or this is
- // just another unused camera).
- if (mClient[i] == 0) continue;
+sp<CameraService::BasicClient> CameraService::removeClientLocked(const String8& cameraId) {
+ // Remove from active clients list
+ auto clientDescriptorPtr = mActiveClientManager.remove(cameraId);
+ if (clientDescriptorPtr == nullptr) {
+ ALOGW("%s: Could not evict client, no client for camera ID %s", __FUNCTION__,
+ cameraId.string());
+ return sp<BasicClient>{nullptr};
+ }
- // Promote mClient. It can fail if we are called from this path:
- // Client::~Client() -> disconnect() -> removeClientByRemote().
- client = mClient[i].promote();
+ return clientDescriptorPtr->getValue();
+}
- // Clean up stale client entry
- if (client == NULL) {
- mClient[i].clear();
+void CameraService::doUserSwitch(int newUserId) {
+ // Acquire mServiceLock and prevent other clients from connecting
+ std::unique_ptr<AutoConditionLock> lock =
+ AutoConditionLock::waitAndAcquire(mServiceLockWrapper);
+
+ if (newUserId <= 0) {
+ ALOGW("%s: Bad user ID %d given during user switch, resetting to default.", __FUNCTION__,
+ newUserId);
+ newUserId = DEFAULT_LAST_USER_ID;
+ }
+
+ mLastUserId = newUserId;
+
+ // Current user has switched, evict all current clients.
+ std::vector<sp<BasicClient>> evicted;
+ for (auto& i : mActiveClientManager.getAll()) {
+ auto clientSp = i->getValue();
+
+ if (clientSp.get() == nullptr) {
+ ALOGE("%s: Dead client still in mActiveClientManager.", __FUNCTION__);
continue;
}
- if (cameraClient == client->getRemote()) {
- // Found our camera
- outIndex = i;
- return client;
- }
+ evicted.push_back(clientSp);
+
+ String8 curTime = getFormattedCurrentTime();
+
+ ALOGE("Evicting conflicting client for camera ID %s due to user change",
+ i->getKey().string());
+ // Log the clients evicted
+ mEventLog.add(String8::format("%s : EVICT device %s client for package %s (PID %"
+ PRId32 ", priority %" PRId32 ")\n - Evicted due to user switch.",
+ curTime.string(), i->getKey().string(),
+ String8{clientSp->getPackageName()}.string(), i->getOwnerId(),
+ i->getPriority()));
+
}
- outIndex = -1;
- return NULL;
+ // Do not hold mServiceLock while disconnecting clients, but retain the condition
+ // blocking other clients from connecting in mServiceLockWrapper if held.
+ mServiceLock.unlock();
+
+ // Clear caller identity temporarily so client disconnect PID checks work correctly
+ int64_t token = IPCThreadState::self()->clearCallingIdentity();
+
+ for (auto& i : evicted) {
+ i->disconnect();
+ }
+
+ IPCThreadState::self()->restoreCallingIdentity(token);
+
+ // Reacquire mServiceLock
+ mServiceLock.lock();
}
-CameraService::BasicClient* CameraService::getClientByIdUnsafe(int cameraId) {
- if (cameraId < 0 || cameraId >= mNumberOfCameras) return NULL;
- return mClient[cameraId].unsafe_get();
+void CameraService::logDisconnected(const String8& cameraId, int clientPid,
+ const String8& clientPackage) {
+
+ String8 curTime = getFormattedCurrentTime();
+ // Log the clients evicted
+ mEventLog.add(String8::format("%s : DISCONNECT device %s client for package %s (PID %d)",
+ curTime.string(), cameraId.string(), clientPackage.string(), clientPid));
}
-Mutex* CameraService::getClientLockById(int cameraId) {
- if (cameraId < 0 || cameraId >= mNumberOfCameras) return NULL;
- return &mClientLock[cameraId];
+void CameraService::logConnected(const String8& cameraId, int clientPid,
+ const String8& clientPackage) {
+
+ String8 curTime = getFormattedCurrentTime();
+ // Log the clients evicted
+ mEventLog.add(String8::format("%s : CONNECT device %s client for package %s (PID %d)",
+ curTime.string(), cameraId.string(), clientPackage.string(), clientPid));
}
-sp<CameraService::BasicClient> CameraService::getClientByRemote(
- const wp<IBinder>& cameraClient) {
+status_t CameraService::onTransact(uint32_t code, const Parcel& data, Parcel* reply,
+ uint32_t flags) {
- // Declare this before the lock to make absolutely sure the
- // destructor won't be called with the lock held.
- sp<BasicClient> client;
+ const int pid = getCallingPid();
+ const int selfPid = getpid();
- Mutex::Autolock lock(mServiceLock);
-
- int outIndex;
- client = findClientUnsafe(cameraClient, outIndex);
-
- return client;
-}
-
-status_t CameraService::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) {
// Permission checks
switch (code) {
case BnCameraService::CONNECT:
- case BnCameraService::CONNECT_PRO:
case BnCameraService::CONNECT_DEVICE:
- case BnCameraService::CONNECT_LEGACY:
- const int pid = getCallingPid();
- const int self_pid = getpid();
- if (pid != self_pid) {
+ case BnCameraService::CONNECT_LEGACY: {
+ if (pid != selfPid) {
// we're called from a different process, do the real check
if (!checkCallingPermission(
String16("android.permission.CAMERA"))) {
@@ -1285,29 +1510,26 @@
}
}
break;
+ }
+ case BnCameraService::NOTIFY_SYSTEM_EVENT: {
+ if (pid != selfPid) {
+ // Ensure we're being called by system_server, or similar process with
+ // permissions to notify the camera service about system events
+ if (!checkCallingPermission(
+ String16("android.permission.CAMERA_SEND_SYSTEM_EVENTS"))) {
+ const int uid = getCallingUid();
+ ALOGE("Permission Denial: cannot send updates to camera service about system"
+ " events from pid=%d, uid=%d", pid, uid);
+ return PERMISSION_DENIED;
+ }
+ }
+ break;
+ }
}
return BnCameraService::onTransact(code, data, reply, flags);
}
-// The reason we need this busy bit is a new CameraService::connect() request
-// may come in while the previous Client's destructor has not been run or is
-// still running. If the last strong reference of the previous Client is gone
-// but the destructor has not been finished, we should not allow the new Client
-// to be created because we need to wait for the previous Client to tear down
-// the hardware first.
-void CameraService::setCameraBusy(int cameraId) {
- android_atomic_write(1, &mBusy[cameraId]);
-
- ALOGV("setCameraBusy cameraId=%d", cameraId);
-}
-
-void CameraService::setCameraFree(int cameraId) {
- android_atomic_write(0, &mBusy[cameraId]);
-
- ALOGV("setCameraFree cameraId=%d", cameraId);
-}
-
// We share the media players for shutter and recording sound for all clients.
// A reference count is kept to determine when we will actually release the
// media players.
@@ -1376,7 +1598,6 @@
mRemoteCallback = cameraClient;
- cameraService->setCameraBusy(cameraId);
cameraService->loadSound();
LOG1("Client::Client X (pid %d, id %d)", callingPid, cameraId);
@@ -1398,7 +1619,7 @@
int cameraId, int cameraFacing,
int clientPid, uid_t clientUid,
int servicePid):
- mClientPackageName(clientPackageName)
+ mClientPackageName(clientPackageName), mDisconnected(false)
{
mCameraService = cameraService;
mRemoteBinder = remoteCallback;
@@ -1417,14 +1638,38 @@
}
void CameraService::BasicClient::disconnect() {
- ALOGV("BasicClient::disconnect");
- mCameraService->removeClientByRemote(mRemoteBinder);
+ if (mDisconnected) {
+ ALOGE("%s: Disconnect called on already disconnected client for device %d", __FUNCTION__,
+ mCameraId);
+ return;
+ }
+ mDisconnected = true;;
+
+ mCameraService->removeByClient(this);
+ mCameraService->logDisconnected(String8::format("%d", mCameraId), mClientPid,
+ String8(mClientPackageName));
+
+ sp<IBinder> remote = getRemote();
+ if (remote != nullptr) {
+ remote->unlinkToDeath(mCameraService);
+ }
finishCameraOps();
+ ALOGI("%s: Disconnected client for camera %d for PID %d", __FUNCTION__, mCameraId, mClientPid);
+
// client shouldn't be able to call into us anymore
mClientPid = 0;
}
+String16 CameraService::BasicClient::getPackageName() const {
+ return mClientPackageName;
+}
+
+
+int CameraService::BasicClient::getClientPid() const {
+ return mClientPid;
+}
+
status_t CameraService::BasicClient::startCameraOps() {
int32_t res;
// Notify app ops that the camera is not available
@@ -1450,7 +1695,7 @@
// Transition device availability listeners from PRESENT -> NOT_AVAILABLE
mCameraService->updateStatus(ICameraServiceListener::STATUS_NOT_AVAILABLE,
- mCameraId);
+ String8::format("%d", mCameraId));
return OK;
}
@@ -1463,19 +1708,16 @@
mClientPackageName);
mOpsActive = false;
- // Notify device availability listeners that this camera is available
- // again
+ auto rejected = {ICameraServiceListener::STATUS_NOT_PRESENT,
+ ICameraServiceListener::STATUS_ENUMERATING};
- StatusVector rejectSourceStates;
- rejectSourceStates.push_back(ICameraServiceListener::STATUS_NOT_PRESENT);
- rejectSourceStates.push_back(ICameraServiceListener::STATUS_ENUMERATING);
-
- // Transition to PRESENT if the camera is not in either of above 2
- // states
+ // Transition to PRESENT if the camera is not in either of the rejected states
mCameraService->updateStatus(ICameraServiceListener::STATUS_PRESENT,
- mCameraId,
- &rejectSourceStates);
+ String8::format("%d", mCameraId), rejected);
+ // Notify flashlight that a camera device is closed.
+ mCameraService->mFlashlight->deviceClosed(
+ String8::format("%d", mCameraId));
}
// Always stop watching, even if no camera op is active
if (mOpsCallback != NULL) {
@@ -1518,26 +1760,15 @@
// ----------------------------------------------------------------------------
-Mutex* CameraService::Client::getClientLockFromCookie(void* user) {
- return gCameraService->getClientLockById((int)(intptr_t) user);
-}
-
-// Provide client pointer for callbacks. Client lock returned from getClientLockFromCookie should
-// be acquired for this to be safe
-CameraService::Client* CameraService::Client::getClientFromCookie(void* user) {
- BasicClient *basicClient = gCameraService->getClientByIdUnsafe((int)(intptr_t) user);
- // OK: only CameraClient calls this, and they already cast anyway.
- Client* client = static_cast<Client*>(basicClient);
-
- // This could happen if the Client is in the process of shutting down (the
- // last strong reference is gone, but the destructor hasn't finished
- // stopping the hardware).
- if (client == NULL) return NULL;
-
- // destruction already started, so should not be accessed
- if (client->mDestructionStarted) return NULL;
-
- return client;
+// Provide client strong pointer for callbacks.
+sp<CameraService::Client> CameraService::Client::getClientFromCookie(void* user) {
+ String8 cameraId = String8::format("%d", (int)(intptr_t) user);
+ auto clientDescriptor = gCameraService->mActiveClientManager.get(cameraId);
+ if (clientDescriptor != nullptr) {
+ return sp<Client>{
+ static_cast<Client*>(clientDescriptor->getValue().get())};
+ }
+ return sp<Client>{nullptr};
}
void CameraService::Client::notifyError(ICameraDeviceCallbacks::CameraErrorCode errorCode,
@@ -1549,7 +1780,6 @@
void CameraService::Client::disconnect() {
ALOGV("Client::disconnect");
BasicClient::disconnect();
- mCameraService->setCameraFree(mCameraId);
}
CameraService::Client::OpsCallback::OpsCallback(wp<BasicClient> client):
@@ -1565,30 +1795,101 @@
}
// ----------------------------------------------------------------------------
-// IProCamera
+// CameraState
// ----------------------------------------------------------------------------
-CameraService::ProClient::ProClient(const sp<CameraService>& cameraService,
- const sp<IProCameraCallbacks>& remoteCallback,
- const String16& clientPackageName,
- int cameraId,
- int cameraFacing,
- int clientPid,
- uid_t clientUid,
- int servicePid)
- : CameraService::BasicClient(cameraService, IInterface::asBinder(remoteCallback),
- clientPackageName, cameraId, cameraFacing,
- clientPid, clientUid, servicePid)
-{
- mRemoteCallback = remoteCallback;
+CameraService::CameraState::CameraState(const String8& id, int cost,
+ const std::set<String8>& conflicting) : mId(id),
+ mStatus(ICameraServiceListener::STATUS_PRESENT), mCost(cost), mConflicting(conflicting) {}
+
+CameraService::CameraState::~CameraState() {}
+
+ICameraServiceListener::Status CameraService::CameraState::getStatus() const {
+ Mutex::Autolock lock(mStatusLock);
+ return mStatus;
}
-CameraService::ProClient::~ProClient() {
+CameraParameters CameraService::CameraState::getShimParams() const {
+ return mShimParams;
}
-void CameraService::ProClient::notifyError(ICameraDeviceCallbacks::CameraErrorCode errorCode,
- const CaptureResultExtras& resultExtras) {
- mRemoteCallback->notifyCallback(CAMERA_MSG_ERROR, CAMERA_ERROR_RELEASED, 0);
+void CameraService::CameraState::setShimParams(const CameraParameters& params) {
+ mShimParams = params;
+}
+
+int CameraService::CameraState::getCost() const {
+ return mCost;
+}
+
+std::set<String8> CameraService::CameraState::getConflicting() const {
+ return mConflicting;
+}
+
+String8 CameraService::CameraState::getId() const {
+ return mId;
+}
+
+// ----------------------------------------------------------------------------
+// CameraClientManager
+// ----------------------------------------------------------------------------
+
+CameraService::CameraClientManager::~CameraClientManager() {}
+
+sp<CameraService::BasicClient> CameraService::CameraClientManager::getCameraClient(
+ const String8& id) const {
+ auto descriptor = get(id);
+ if (descriptor == nullptr) {
+ return sp<BasicClient>{nullptr};
+ }
+ return descriptor->getValue();
+}
+
+String8 CameraService::CameraClientManager::toString() const {
+ auto all = getAll();
+ String8 ret("[");
+ bool hasAny = false;
+ for (auto& i : all) {
+ hasAny = true;
+ String8 key = i->getKey();
+ int32_t cost = i->getCost();
+ int32_t pid = i->getOwnerId();
+ int32_t priority = i->getPriority();
+ auto conflicting = i->getConflicting();
+ auto clientSp = i->getValue();
+ String8 packageName;
+ if (clientSp.get() != nullptr) {
+ packageName = String8{clientSp->getPackageName()};
+ }
+ ret.appendFormat("\n(Camera ID: %s, Cost: %" PRId32 ", PID: %" PRId32 ", Priority: %"
+ PRId32 ", ", key.string(), cost, pid, priority);
+
+ if (packageName.size() != 0) {
+ ret.appendFormat("Client Package Name: %s", packageName.string());
+ }
+
+ ret.append(", Conflicting Client Devices: {");
+ for (auto& j : conflicting) {
+ ret.appendFormat("%s, ", j.string());
+ }
+ ret.append("})");
+ }
+ if (hasAny) ret.append("\n");
+ ret.append("]\n");
+ return ret;
+}
+
+CameraService::DescriptorPtr CameraService::CameraClientManager::makeClientDescriptor(
+ const String8& key, const sp<BasicClient>& value, int32_t cost,
+ const std::set<String8>& conflictingKeys, int32_t priority, int32_t ownerId) {
+
+ return std::make_shared<resource_policy::ClientDescriptor<String8, sp<BasicClient>>>(
+ key, value, cost, conflictingKeys, priority, ownerId);
+}
+
+CameraService::DescriptorPtr CameraService::CameraClientManager::makeClientDescriptor(
+ const sp<BasicClient>& value, const CameraService::DescriptorPtr& partial) {
+ return makeClientDescriptor(partial->getKey(), value, partial->getCost(),
+ partial->getConflicting(), partial->getPriority(), partial->getOwnerId());
}
// ----------------------------------------------------------------------------
@@ -1633,15 +1934,13 @@
return NO_ERROR;
}
- result = String8::format("Camera module HAL API version: 0x%x\n",
- mModule->common.hal_api_version);
- result.appendFormat("Camera module API version: 0x%x\n",
- mModule->common.module_api_version);
- result.appendFormat("Camera module name: %s\n",
- mModule->common.name);
- result.appendFormat("Camera module author: %s\n",
- mModule->common.author);
- result.appendFormat("Number of camera devices: %d\n\n", mNumberOfCameras);
+ result = String8::format("Camera module HAL API version: 0x%x\n", mModule->getHalApiVersion());
+ result.appendFormat("Camera module API version: 0x%x\n", mModule->getModuleApiVersion());
+ result.appendFormat("Camera module name: %s\n", mModule->getModuleName());
+ result.appendFormat("Camera module author: %s\n", mModule->getModuleAuthor());
+ result.appendFormat("Number of camera devices: %d\n", mNumberOfCameras);
+ String8 activeClientString = mActiveClientManager.toString();
+ result.appendFormat("Active Camera Clients:\n%s", activeClientString.string());
sp<VendorTagDescriptor> desc = VendorTagDescriptor::getGlobalVendorTagDescriptor();
if (desc == NULL) {
@@ -1656,11 +1955,31 @@
desc->dump(fd, /*verbosity*/2, /*indentation*/4);
}
- for (int i = 0; i < mNumberOfCameras; i++) {
- result = String8::format("Camera %d static information:\n", i);
+ result = String8("Prior client events (most recent at top):\n");
+
+ for (const auto& msg : mEventLog) {
+ result.appendFormat("%s\n", msg.string());
+ }
+
+ if (mEventLog.size() == DEFAULT_EVICTION_LOG_LENGTH) {
+ result.append("...\n");
+ }
+
+ write(fd, result.string(), result.size());
+
+ bool stateLocked = tryLock(mCameraStatesLock);
+ if (!stateLocked) {
+ result = String8::format("CameraStates in use, may be deadlocked\n");
+ write(fd, result.string(), result.size());
+ }
+
+ for (auto& state : mCameraStates) {
+ String8 cameraId = state.first;
+ result = String8::format("Camera %s information:\n", cameraId.string());
camera_info info;
- status_t rc = mModule->get_camera_info(i, &info);
+ // TODO: Change getCameraInfo + HAL to use String cameraIds
+ status_t rc = mModule->getCameraInfo(cameraIdToInt(cameraId), &info);
if (rc != OK) {
result.appendFormat(" Error reading static information!\n");
write(fd, result.string(), result.size());
@@ -1669,13 +1988,24 @@
info.facing == CAMERA_FACING_BACK ? "BACK" : "FRONT");
result.appendFormat(" Orientation: %d\n", info.orientation);
int deviceVersion;
- if (mModule->common.module_api_version <
- CAMERA_MODULE_API_VERSION_2_0) {
+ if (mModule->getModuleApiVersion() < CAMERA_MODULE_API_VERSION_2_0) {
deviceVersion = CAMERA_DEVICE_API_VERSION_1_0;
} else {
deviceVersion = info.device_version;
}
- result.appendFormat(" Device version: 0x%x\n", deviceVersion);
+
+ auto conflicting = state.second->getConflicting();
+ result.appendFormat(" Resource Cost: %d\n", state.second->getCost());
+ result.appendFormat(" Conflicting Devices:");
+ for (auto& id : conflicting) {
+ result.appendFormat(" %s", cameraId.string());
+ }
+ if (conflicting.size() == 0) {
+ result.appendFormat(" NONE");
+ }
+ result.appendFormat("\n");
+
+ result.appendFormat(" Device version: %#x\n", deviceVersion);
if (deviceVersion >= CAMERA_DEVICE_API_VERSION_2_0) {
result.appendFormat(" Device static metadata:\n");
write(fd, result.string(), result.size());
@@ -1684,19 +2014,38 @@
} else {
write(fd, result.string(), result.size());
}
+
+ CameraParameters p = state.second->getShimParams();
+ if (!p.isEmpty()) {
+ result = String8::format(" Camera1 API shim is using parameters:\n ");
+ write(fd, result.string(), result.size());
+ p.dump(fd, args);
+ }
}
- sp<BasicClient> client = mClient[i].promote();
- if (client == 0) {
- result = String8::format(" Device is closed, no client instance\n");
+ auto clientDescriptor = mActiveClientManager.get(cameraId);
+ if (clientDescriptor == nullptr) {
+ result = String8::format(" Device %s is closed, no client instance\n",
+ cameraId.string());
write(fd, result.string(), result.size());
continue;
}
hasClient = true;
- result = String8::format(" Device is open. Client instance dump:\n");
+ result = String8::format(" Device %s is open. Client instance dump:\n\n",
+ cameraId.string());
+ result.appendFormat("Client priority level: %d\n", clientDescriptor->getPriority());
+ result.appendFormat("Client PID: %d\n", clientDescriptor->getOwnerId());
+
+ auto client = clientDescriptor->getValue();
+ result.appendFormat("Client package: %s\n",
+ String8(client->getPackageName()).string());
write(fd, result.string(), result.size());
+
client->dump(fd, args);
}
+
+ if (stateLocked) mCameraStatesLock.unlock();
+
if (!hasClient) {
result = String8::format("\nNo active camera clients yet.\n");
write(fd, result.string(), result.size());
@@ -1720,112 +2069,119 @@
write(fd, result.string(), result.size());
}
}
-
}
return NO_ERROR;
}
-/*virtual*/void CameraService::binderDied(
- const wp<IBinder> &who) {
+void CameraService::handleTorchClientBinderDied(const wp<IBinder> &who) {
+ Mutex::Autolock al(mTorchClientMapMutex);
+ for (size_t i = 0; i < mTorchClientMap.size(); i++) {
+ if (mTorchClientMap[i] == who) {
+ // turn off the torch mode that was turned on by dead client
+ String8 cameraId = mTorchClientMap.keyAt(i);
+ status_t res = mFlashlight->setTorchMode(cameraId, false);
+ if (res) {
+ ALOGE("%s: torch client died but couldn't turn off torch: "
+ "%s (%d)", __FUNCTION__, strerror(-res), res);
+ return;
+ }
+ mTorchClientMap.removeItemsAt(i);
+ break;
+ }
+ }
+}
+
+/*virtual*/void CameraService::binderDied(const wp<IBinder> &who) {
/**
* While tempting to promote the wp<IBinder> into a sp,
* it's actually not supported by the binder driver
*/
- ALOGV("java clients' binder died");
+ // check torch client
+ handleTorchClientBinderDied(who);
- sp<BasicClient> cameraClient = getClientByRemote(who);
-
- if (cameraClient == 0) {
- ALOGV("java clients' binder death already cleaned up (normal case)");
+ // check camera device client
+ if(!evictClientIdByRemote(who)) {
+ ALOGV("%s: Java client's binder death already cleaned up (normal case)", __FUNCTION__);
return;
}
- ALOGW("Disconnecting camera client %p since the binder for it "
- "died (this pid %d)", cameraClient.get(), getCallingPid());
-
- cameraClient->disconnect();
-
+ ALOGE("%s: Java client's binder died, removing it from the list of active clients",
+ __FUNCTION__);
}
-void CameraService::updateStatus(ICameraServiceListener::Status status,
- int32_t cameraId,
- const StatusVector *rejectSourceStates) {
- // do not lock mServiceLock here or can get into a deadlock from
- // connect() -> ProClient::disconnect -> updateStatus
- Mutex::Autolock lock(mStatusMutex);
-
- ICameraServiceListener::Status oldStatus = mStatusList[cameraId];
-
- mStatusList[cameraId] = status;
-
- if (oldStatus != status) {
- ALOGV("%s: Status has changed for camera ID %d from 0x%x to 0x%x",
- __FUNCTION__, cameraId, (uint32_t)oldStatus, (uint32_t)status);
-
- if (oldStatus == ICameraServiceListener::STATUS_NOT_PRESENT &&
- (status != ICameraServiceListener::STATUS_PRESENT &&
- status != ICameraServiceListener::STATUS_ENUMERATING)) {
-
- ALOGW("%s: From NOT_PRESENT can only transition into PRESENT"
- " or ENUMERATING", __FUNCTION__);
- mStatusList[cameraId] = oldStatus;
- return;
- }
-
- if (rejectSourceStates != NULL) {
- const StatusVector &rejectList = *rejectSourceStates;
- StatusVector::const_iterator it = rejectList.begin();
-
- /**
- * Sometimes we want to conditionally do a transition.
- * For example if a client disconnects, we want to go to PRESENT
- * only if we weren't already in NOT_PRESENT or ENUMERATING.
- */
- for (; it != rejectList.end(); ++it) {
- if (oldStatus == *it) {
- ALOGV("%s: Rejecting status transition for Camera ID %d, "
- " since the source state was was in one of the bad "
- " states.", __FUNCTION__, cameraId);
- mStatusList[cameraId] = oldStatus;
- return;
- }
- }
- }
-
- /**
- * ProClients lose their exclusive lock.
- * - Done before the CameraClient can initialize the HAL device,
- * since we want to be able to close it before they get to initialize
- */
- if (status == ICameraServiceListener::STATUS_NOT_AVAILABLE) {
- Vector<wp<ProClient> > proClients(mProClientList[cameraId]);
- Vector<wp<ProClient> >::const_iterator it;
-
- for (it = proClients.begin(); it != proClients.end(); ++it) {
- sp<ProClient> proCl = it->promote();
- if (proCl.get() != NULL) {
- proCl->onExclusiveLockStolen();
- }
- }
- }
-
- Vector<sp<ICameraServiceListener> >::const_iterator it;
- for (it = mListenerList.begin(); it != mListenerList.end(); ++it) {
- (*it)->onStatusChanged(status, cameraId);
- }
- }
+void CameraService::updateStatus(ICameraServiceListener::Status status, const String8& cameraId) {
+ updateStatus(status, cameraId, {});
}
-ICameraServiceListener::Status CameraService::getStatus(int cameraId) const {
- if (cameraId < 0 || cameraId >= MAX_CAMERAS) {
- ALOGE("%s: Invalid camera ID %d", __FUNCTION__, cameraId);
- return ICameraServiceListener::STATUS_UNKNOWN;
+void CameraService::updateStatus(ICameraServiceListener::Status status, const String8& cameraId,
+ std::initializer_list<ICameraServiceListener::Status> rejectSourceStates) {
+ // Do not lock mServiceLock here or can get into a deadlock from
+ // connect() -> disconnect -> updateStatus
+
+ auto state = getCameraState(cameraId);
+
+ if (state == nullptr) {
+ ALOGW("%s: Could not update the status for %s, no such device exists", __FUNCTION__,
+ cameraId.string());
+ return;
}
- Mutex::Autolock al(mStatusMutex);
- return mStatusList[cameraId];
+ // Update the status for this camera state, then send the onStatusChangedCallbacks to each
+ // of the listeners with both the mStatusStatus and mStatusListenerLock held
+ state->updateStatus(status, cameraId, rejectSourceStates, [this]
+ (const String8& cameraId, ICameraServiceListener::Status status) {
+
+ // Update torch status
+ if (status == ICameraServiceListener::STATUS_NOT_PRESENT ||
+ status == ICameraServiceListener::STATUS_NOT_AVAILABLE) {
+ // Update torch status to not available when the camera device becomes not present
+ // or not available.
+ onTorchStatusChanged(cameraId, ICameraServiceListener::TORCH_STATUS_NOT_AVAILABLE);
+ } else if (status == ICameraServiceListener::STATUS_PRESENT) {
+ // Update torch status to available when the camera device becomes present or
+ // available
+ onTorchStatusChanged(cameraId, ICameraServiceListener::TORCH_STATUS_AVAILABLE_OFF);
+ }
+
+ Mutex::Autolock lock(mStatusListenerLock);
+
+ for (auto& listener : mListenerList) {
+ // TODO: Refactor status listeners to use strings for Camera IDs and remove this.
+ int id = cameraIdToInt(cameraId);
+ if (id != -1) listener->onStatusChanged(status, id);
+ }
+ });
+}
+
+status_t CameraService::getTorchStatusLocked(
+ const String8& cameraId,
+ ICameraServiceListener::TorchStatus *status) const {
+ if (!status) {
+ return BAD_VALUE;
+ }
+ ssize_t index = mTorchStatusMap.indexOfKey(cameraId);
+ if (index == NAME_NOT_FOUND) {
+ // invalid camera ID or the camera doesn't have a flash unit
+ return NAME_NOT_FOUND;
+ }
+
+ *status = mTorchStatusMap.valueAt(index);
+ return OK;
+}
+
+status_t CameraService::setTorchStatusLocked(const String8& cameraId,
+ ICameraServiceListener::TorchStatus status) {
+ ssize_t index = mTorchStatusMap.indexOfKey(cameraId);
+ if (index == NAME_NOT_FOUND) {
+ return BAD_VALUE;
+ }
+ ICameraServiceListener::TorchStatus& item =
+ mTorchStatusMap.editValueAt(index);
+ item = status;
+
+ return OK;
}
}; // namespace android
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 126d8d9..ca1c504 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -17,6 +17,7 @@
#ifndef ANDROID_SERVERS_CAMERA_CAMERASERVICE_H
#define ANDROID_SERVERS_CAMERA_CAMERASERVICE_H
+#include <cutils/multiuser.h>
#include <utils/Vector.h>
#include <utils/KeyedVector.h>
#include <binder/AppOpsManager.h>
@@ -27,8 +28,6 @@
#include <camera/ICamera.h>
#include <camera/ICameraClient.h>
-#include <camera/IProCameraUser.h>
-#include <camera/IProCameraCallbacks.h>
#include <camera/camera2/ICameraDeviceUser.h>
#include <camera/camera2/ICameraDeviceCallbacks.h>
#include <camera/VendorTagDescriptor.h>
@@ -36,9 +35,17 @@
#include <camera/CameraParameters.h>
#include <camera/ICameraServiceListener.h>
+#include "CameraFlashlight.h"
-/* This needs to be increased if we can have more cameras */
-#define MAX_CAMERAS 2
+#include "common/CameraModule.h"
+#include "utils/AutoConditionLock.h"
+#include "utils/ClientManager.h"
+#include "utils/RingBuffer.h"
+
+#include <set>
+#include <string>
+#include <map>
+#include <memory>
namespace android {
@@ -58,6 +65,39 @@
class Client;
class BasicClient;
+ enum apiLevel {
+ API_1 = 1,
+ API_2 = 2
+ };
+
+ // Process States (mirrors frameworks/base/core/java/android/app/ActivityManager.java)
+ static const int PROCESS_STATE_NONEXISTENT = -1;
+ static const int PROCESS_STATE_PERSISTENT = 0;
+ static const int PROCESS_STATE_PERSISTENT_UI = 1;
+ static const int PROCESS_STATE_TOP = 2;
+ static const int PROCESS_STATE_IMPORTANT_FOREGROUND = 3;
+ static const int PROCESS_STATE_IMPORTANT_BACKGROUND = 4;
+ static const int PROCESS_STATE_BACKUP = 5;
+ static const int PROCESS_STATE_HEAVY_WEIGHT = 6;
+ static const int PROCESS_STATE_SERVICE = 7;
+ static const int PROCESS_STATE_RECEIVER = 8;
+ static const int PROCESS_STATE_HOME = 9;
+ static const int PROCESS_STATE_LAST_ACTIVITY = 10;
+ static const int PROCESS_STATE_CACHED_ACTIVITY = 11;
+ static const int PROCESS_STATE_CACHED_ACTIVITY_CLIENT = 12;
+ static const int PROCESS_STATE_CACHED_EMPTY = 13;
+
+ // 3 second busy timeout when other clients are connecting
+ static const nsecs_t DEFAULT_CONNECT_TIMEOUT_NS = 3000000000;
+
+ // Default number of messages to store in eviction log
+ static const size_t DEFAULT_EVICTION_LOG_LENGTH = 50;
+
+ enum {
+ // Default last user id
+ DEFAULT_LAST_USER_ID = 0,
+ };
+
// Implementation of BinderService<T>
static char const* getServiceName() { return "media.camera"; }
@@ -66,8 +106,11 @@
/////////////////////////////////////////////////////////////////////
// HAL Callbacks
- virtual void onDeviceStatusChanged(int cameraId,
- int newStatus);
+ virtual void onDeviceStatusChanged(camera_device_status_t cameraId,
+ camera_device_status_t newStatus);
+ virtual void onTorchStatusChanged(const String8& cameraId,
+ ICameraServiceListener::TorchStatus
+ newStatus);
/////////////////////////////////////////////////////////////////////
// ICameraService
@@ -88,11 +131,6 @@
/*out*/
sp<ICamera>& device);
- virtual status_t connectPro(const sp<IProCameraCallbacks>& cameraCb,
- int cameraId, const String16& clientPackageName, int clientUid,
- /*out*/
- sp<IProCameraUser>& device);
-
virtual status_t connectDevice(
const sp<ICameraDeviceCallbacks>& cameraCb,
int cameraId,
@@ -110,6 +148,11 @@
/*out*/
String16* parameters);
+ virtual status_t setTorchMode(const String16& cameraId, bool enabled,
+ const sp<IBinder>& clientBinder);
+
+ virtual void notifySystemEvent(int eventId, int arg0);
+
// OK = supports api of that version, -EOPNOTSUPP = does not support
virtual status_t supportsCameraApi(
int cameraId, int apiVersion);
@@ -122,7 +165,6 @@
/////////////////////////////////////////////////////////////////////
// Client functionality
- virtual void removeClientByRemote(const wp<IBinder>& remoteBinder);
enum sound_kind {
SOUND_SHUTTER = 0,
@@ -140,33 +182,36 @@
/////////////////////////////////////////////////////////////////////
// Shared utilities
- static status_t filterOpenErrorCode(status_t err);
static status_t filterGetInfoErrorCode(status_t err);
/////////////////////////////////////////////////////////////////////
// CameraClient functionality
- // returns plain pointer of client. Note that mClientLock should be acquired to
- // prevent the client from destruction. The result can be NULL.
- virtual BasicClient* getClientByIdUnsafe(int cameraId);
- virtual Mutex* getClientLockById(int cameraId);
-
class BasicClient : public virtual RefBase {
public:
- virtual status_t initialize(camera_module_t *module) = 0;
+ virtual status_t initialize(CameraModule *module) = 0;
virtual void disconnect();
// because we can't virtually inherit IInterface, which breaks
// virtual inheritance
virtual sp<IBinder> asBinderWrapper() = 0;
- // Return the remote callback binder object (e.g. IProCameraCallbacks)
+ // Return the remote callback binder object (e.g. ICameraDeviceCallbacks)
sp<IBinder> getRemote() {
return mRemoteBinder;
}
virtual status_t dump(int fd, const Vector<String16>& args) = 0;
+ // Return the package name for this client
+ virtual String16 getPackageName() const;
+
+ // Notify client about a fatal error
+ virtual void notifyError(ICameraDeviceCallbacks::CameraErrorCode errorCode,
+ const CaptureResultExtras& resultExtras) = 0;
+
+ // Get the PID of the application client using this
+ virtual int getClientPid() const;
protected:
BasicClient(const sp<CameraService>& cameraService,
const sp<IBinder>& remoteCallback,
@@ -193,6 +238,7 @@
pid_t mClientPid;
uid_t mClientUid; // immutable after constructor
pid_t mServicePid; // immutable after constructor
+ bool mDisconnected;
// - The app-side Binder interface to receive callbacks from us
sp<IBinder> mRemoteBinder; // immutable after constructor
@@ -201,10 +247,6 @@
status_t startCameraOps();
status_t finishCameraOps();
- // Notify client about a fatal error
- virtual void notifyError(
- ICameraDeviceCallbacks::CameraErrorCode errorCode,
- const CaptureResultExtras& resultExtras) = 0;
private:
AppOpsManager mAppOpsManager;
@@ -276,13 +318,11 @@
return asBinder(this);
}
- protected:
- static Mutex* getClientLockFromCookie(void* user);
- // convert client from cookie. Client lock should be acquired before getting Client.
- static Client* getClientFromCookie(void* user);
-
virtual void notifyError(ICameraDeviceCallbacks::CameraErrorCode errorCode,
const CaptureResultExtras& resultExtras);
+ protected:
+ // Convert client from cookie.
+ static sp<CameraService::Client> getClientFromCookie(void* user);
// Initialized in constructor
@@ -291,93 +331,232 @@
}; // class Client
- class ProClient : public BnProCameraUser, public BasicClient {
+ typedef std::shared_ptr<resource_policy::ClientDescriptor<String8,
+ sp<CameraService::BasicClient>>> DescriptorPtr;
+
+ /**
+ * A container class for managing active camera clients that are using HAL devices. Active
+ * clients are represented by ClientDescriptor objects that contain strong pointers to the
+ * actual BasicClient subclass binder interface implementation.
+ *
+ * This class manages the eviction behavior for the camera clients. See the parent class
+ * implementation in utils/ClientManager for the specifics of this behavior.
+ */
+ class CameraClientManager :
+ public resource_policy::ClientManager<String8, sp<CameraService::BasicClient>> {
public:
- typedef IProCameraCallbacks TCamCallbacks;
+ virtual ~CameraClientManager();
- ProClient(const sp<CameraService>& cameraService,
- const sp<IProCameraCallbacks>& remoteCallback,
- const String16& clientPackageName,
- int cameraId,
- int cameraFacing,
- int clientPid,
- uid_t clientUid,
- int servicePid);
+ /**
+ * Return a strong pointer to the active BasicClient for this camera ID, or an empty
+ * if none exists.
+ */
+ sp<CameraService::BasicClient> getCameraClient(const String8& id) const;
- virtual ~ProClient();
+ /**
+ * Return a string describing the current state.
+ */
+ String8 toString() const;
- const sp<IProCameraCallbacks>& getRemoteCallback() {
- return mRemoteCallback;
- }
+ /**
+ * Make a ClientDescriptor object wrapping the given BasicClient strong pointer.
+ */
+ static DescriptorPtr makeClientDescriptor(const String8& key, const sp<BasicClient>& value,
+ int32_t cost, const std::set<String8>& conflictingKeys, int32_t priority,
+ int32_t ownerId);
- /***
- IProCamera implementation
- ***/
- virtual status_t connect(const sp<IProCameraCallbacks>& callbacks)
- = 0;
- virtual status_t exclusiveTryLock() = 0;
- virtual status_t exclusiveLock() = 0;
- virtual status_t exclusiveUnlock() = 0;
+ /**
+ * Make a ClientDescriptor object wrapping the given BasicClient strong pointer with
+ * values intialized from a prior ClientDescriptor.
+ */
+ static DescriptorPtr makeClientDescriptor(const sp<BasicClient>& value,
+ const CameraService::DescriptorPtr& partial);
- virtual bool hasExclusiveLock() = 0;
-
- // Note that the callee gets a copy of the metadata.
- virtual int submitRequest(camera_metadata_t* metadata,
- bool streaming = false) = 0;
- virtual status_t cancelRequest(int requestId) = 0;
-
- // Callbacks from camera service
- virtual void onExclusiveLockStolen() = 0;
-
- protected:
- virtual void notifyError(ICameraDeviceCallbacks::CameraErrorCode errorCode,
- const CaptureResultExtras& resultExtras);
-
- sp<IProCameraCallbacks> mRemoteCallback;
- }; // class ProClient
+ }; // class CameraClientManager
private:
+ /**
+ * Container class for the state of each logical camera device, including: ID, status, and
+ * dependencies on other devices. The mapping of camera ID -> state saved in mCameraStates
+ * represents the camera devices advertised by the HAL (and any USB devices, when we add
+ * those).
+ *
+ * This container does NOT represent an active camera client. These are represented using
+ * the ClientDescriptors stored in mActiveClientManager.
+ */
+ class CameraState {
+ public:
+ /**
+ * Make a new CameraState and set the ID, cost, and conflicting devices using the values
+ * returned in the HAL's camera_info struct for each device.
+ */
+ CameraState(const String8& id, int cost, const std::set<String8>& conflicting);
+ virtual ~CameraState();
+
+ /**
+ * Return the status for this device.
+ *
+ * This method acquires mStatusLock.
+ */
+ ICameraServiceListener::Status getStatus() const;
+
+ /**
+ * This function updates the status for this camera device, unless the given status
+ * is in the given list of rejected status states, and execute the function passed in
+ * with a signature onStatusUpdateLocked(const String8&, ICameraServiceListener::Status)
+ * if the status has changed.
+ *
+ * This method is idempotent, and will not result in the function passed to
+ * onStatusUpdateLocked being called more than once for the same arguments.
+ * This method aquires mStatusLock.
+ */
+ template<class Func>
+ void updateStatus(ICameraServiceListener::Status status, const String8& cameraId,
+ std::initializer_list<ICameraServiceListener::Status> rejectSourceStates,
+ Func onStatusUpdatedLocked);
+
+ /**
+ * Return the last set CameraParameters object generated from the information returned by
+ * the HAL for this device (or an empty CameraParameters object if none has been set).
+ */
+ CameraParameters getShimParams() const;
+
+ /**
+ * Set the CameraParameters for this device.
+ */
+ void setShimParams(const CameraParameters& params);
+
+ /**
+ * Return the resource_cost advertised by the HAL for this device.
+ */
+ int getCost() const;
+
+ /**
+ * Return a set of the IDs of conflicting devices advertised by the HAL for this device.
+ */
+ std::set<String8> getConflicting() const;
+
+ /**
+ * Return the ID of this camera device.
+ */
+ String8 getId() const;
+
+ private:
+ const String8 mId;
+ ICameraServiceListener::Status mStatus; // protected by mStatusLock
+ const int mCost;
+ std::set<String8> mConflicting;
+ mutable Mutex mStatusLock;
+ CameraParameters mShimParams;
+ }; // class CameraState
+
// Delay-load the Camera HAL module
virtual void onFirstRef();
- // Step 1. Check if we can connect, before we acquire the service lock.
- status_t validateConnect(int cameraId,
- /*inout*/
- int& clientUid) const;
+ // Check if we can connect, before we acquire the service lock.
+ status_t validateConnectLocked(const String8& cameraId, /*inout*/int& clientUid) const;
- // Step 2. Check if we can connect, after we acquire the service lock.
- bool canConnectUnsafe(int cameraId,
- const String16& clientPackageName,
- const sp<IBinder>& remoteCallback,
- /*out*/
- sp<BasicClient> &client);
+ // Handle active client evictions, and update service state.
+ // Only call with with mServiceLock held.
+ status_t handleEvictionsLocked(const String8& cameraId, int clientPid,
+ apiLevel effectiveApiLevel, const sp<IBinder>& remoteCallback, const String8& packageName,
+ /*out*/
+ sp<BasicClient>* client,
+ std::shared_ptr<resource_policy::ClientDescriptor<String8, sp<BasicClient>>>* partial);
- // When connection is successful, initialize client and track its death
- status_t connectFinishUnsafe(const sp<BasicClient>& client,
- const sp<IBinder>& remoteCallback);
+ // Single implementation shared between the various connect calls
+ template<class CALLBACK, class CLIENT>
+ status_t connectHelper(const sp<CALLBACK>& cameraCb, const String8& cameraId, int halVersion,
+ const String16& clientPackageName, int clientUid, apiLevel effectiveApiLevel,
+ bool legacyMode, bool shimUpdateOnly, /*out*/sp<CLIENT>& device);
- virtual sp<BasicClient> getClientByRemote(const wp<IBinder>& cameraClient);
+ // Lock guarding camera service state
Mutex mServiceLock;
- // either a Client or CameraDeviceClient
- wp<BasicClient> mClient[MAX_CAMERAS]; // protected by mServiceLock
- Mutex mClientLock[MAX_CAMERAS]; // prevent Client destruction inside callbacks
+
+ // Condition to use with mServiceLock, used to handle simultaneous connect calls from clients
+ std::shared_ptr<WaitableMutexWrapper> mServiceLockWrapper;
+
+ // Return NO_ERROR if the device with a give ID can be connected to
+ status_t checkIfDeviceIsUsable(const String8& cameraId) const;
+
+ // Container for managing currently active application-layer clients
+ CameraClientManager mActiveClientManager;
+
+ // Mapping from camera ID -> state for each device, map is protected by mCameraStatesLock
+ std::map<String8, std::shared_ptr<CameraState>> mCameraStates;
+
+ // Mutex guarding mCameraStates map
+ mutable Mutex mCameraStatesLock;
+
+ // Circular buffer for storing event logging for dumps
+ RingBuffer<String8> mEventLog;
+
+ // UID of last user.
+ int mLastUserId;
+
+ /**
+ * Get the camera state for a given camera id.
+ *
+ * This acquires mCameraStatesLock.
+ */
+ std::shared_ptr<CameraService::CameraState> getCameraState(const String8& cameraId) const;
+
+ /**
+ * Evict client who's remote binder has died. Returns true if this client was in the active
+ * list and was disconnected.
+ *
+ * This method acquires mServiceLock.
+ */
+ bool evictClientIdByRemote(const wp<IBinder>& cameraClient);
+
+ /**
+ * Remove the given client from the active clients list; does not disconnect the client.
+ *
+ * This method acquires mServiceLock.
+ */
+ void removeByClient(const BasicClient* client);
+
+ /**
+ * Add new client to active clients list after conflicting clients have disconnected using the
+ * values set in the partial descriptor passed in to construct the actual client descriptor.
+ * This is typically called at the end of a connect call.
+ *
+ * This method must be called with mServiceLock held.
+ */
+ void finishConnectLocked(const sp<BasicClient>& client, const DescriptorPtr& desc);
+
+ /**
+ * Returns the integer corresponding to the given camera ID string, or -1 on failure.
+ */
+ static int cameraIdToInt(const String8& cameraId);
+
+ /**
+ * Remove a single client corresponding to the given camera id from the list of active clients.
+ * If none exists, return an empty strongpointer.
+ *
+ * This method must be called with mServiceLock held.
+ */
+ sp<CameraService::BasicClient> removeClientLocked(const String8& cameraId);
+
+ /**
+ * Handle a notification that the current device user has changed.
+ */
+ void doUserSwitch(int newUserId);
+
+ /**
+ * Add a event log message that a client has been disconnected.
+ */
+ void logDisconnected(const String8& cameraId, int clientPid, const String8& clientPackage);
+
+ /**
+ * Add a event log message that a client has been connected.
+ */
+ void logConnected(const String8& cameraId, int clientPid, const String8& clientPackage);
+
int mNumberOfCameras;
- typedef wp<ProClient> weak_pro_client_ptr;
- Vector<weak_pro_client_ptr> mProClientList[MAX_CAMERAS];
-
- // needs to be called with mServiceLock held
- sp<BasicClient> findClientUnsafe(const wp<IBinder>& cameraClient, int& outIndex);
- sp<ProClient> findProClientUnsafe(
- const wp<IBinder>& cameraCallbacksRemote);
-
- // atomics to record whether the hardware is allocated to some client.
- volatile int32_t mBusy[MAX_CAMERAS];
- void setCameraBusy(int cameraId);
- void setCameraFree(int cameraId);
-
// sounds
MediaPlayer* newMediaPlayer(const char *file);
@@ -385,45 +564,60 @@
sp<MediaPlayer> mSoundPlayer[NUM_SOUNDS];
int mSoundRef; // reference count (release all MediaPlayer when 0)
- camera_module_t *mModule;
+ CameraModule* mModule;
- Vector<sp<ICameraServiceListener> >
- mListenerList;
+ // Guarded by mStatusListenerMutex
+ std::vector<sp<ICameraServiceListener>> mListenerList;
+ Mutex mStatusListenerLock;
- // guard only mStatusList and the broadcasting of ICameraServiceListener
- mutable Mutex mStatusMutex;
- ICameraServiceListener::Status
- mStatusList[MAX_CAMERAS];
+ /**
+ * Update the status for the given camera id (if that device exists), and broadcast the
+ * status update to all current ICameraServiceListeners if the status has changed. Any
+ * statuses in rejectedSourceStates will be ignored.
+ *
+ * This method must be idempotent.
+ * This method acquires mStatusLock and mStatusListenerLock.
+ */
+ void updateStatus(ICameraServiceListener::Status status, const String8& cameraId,
+ std::initializer_list<ICameraServiceListener::Status> rejectedSourceStates);
+ void updateStatus(ICameraServiceListener::Status status, const String8& cameraId);
- // Read the current status (locks mStatusMutex)
- ICameraServiceListener::Status
- getStatus(int cameraId) const;
+ // flashlight control
+ sp<CameraFlashlight> mFlashlight;
+ // guard mTorchStatusMap
+ Mutex mTorchStatusMutex;
+ // guard mTorchClientMap
+ Mutex mTorchClientMapMutex;
+ // camera id -> torch status
+ KeyedVector<String8, ICameraServiceListener::TorchStatus> mTorchStatusMap;
+ // camera id -> torch client binder
+ // only store the last client that turns on each camera's torch mode
+ KeyedVector<String8, sp<IBinder> > mTorchClientMap;
- typedef Vector<ICameraServiceListener::Status> StatusVector;
- // Broadcast the new status if it changed (locks the service mutex)
- void updateStatus(
- ICameraServiceListener::Status status,
- int32_t cameraId,
- const StatusVector *rejectSourceStates = NULL);
+ // check and handle if torch client's process has died
+ void handleTorchClientBinderDied(const wp<IBinder> &who);
+
+ // handle torch mode status change and invoke callbacks. mTorchStatusMutex
+ // should be locked.
+ void onTorchStatusChangedLocked(const String8& cameraId,
+ ICameraServiceListener::TorchStatus newStatus);
+
+ // get a camera's torch status. mTorchStatusMutex should be locked.
+ status_t getTorchStatusLocked(const String8 &cameraId,
+ ICameraServiceListener::TorchStatus *status) const;
+
+ // set a camera's torch status. mTorchStatusMutex should be locked.
+ status_t setTorchStatusLocked(const String8 &cameraId,
+ ICameraServiceListener::TorchStatus status);
// IBinder::DeathRecipient implementation
virtual void binderDied(const wp<IBinder> &who);
// Helpers
- bool isValidCameraId(int cameraId);
-
bool setUpVendorTags();
/**
- * A mapping of camera ids to CameraParameters returned by that camera device.
- *
- * This cache is used to generate CameraCharacteristic metadata when using
- * the HAL1 shim.
- */
- KeyedVector<int, CameraParameters> mShimParams;
-
- /**
* Initialize and cache the metadata used by the HAL1 shim for a given cameraId.
*
* Returns OK on success, or a negative error code.
@@ -446,25 +640,192 @@
*/
status_t generateShimMetadata(int cameraId, /*out*/CameraMetadata* cameraInfo);
+ static int getCallingPid();
+
+ static int getCallingUid();
+
/**
- * Connect a new camera client. This should only be used while holding the
- * mutex for mServiceLock.
- *
- * Returns OK on success, or a negative error code.
+ * Get the current system time as a formatted string.
*/
- status_t connectHelperLocked(
- /*out*/
- sp<Client>& client,
- /*in*/
- const sp<ICameraClient>& cameraClient,
- int cameraId,
- const String16& clientPackageName,
- int clientUid,
- int callingPid,
- int halVersion = CAMERA_HAL_API_VERSION_UNSPECIFIED,
- bool legacyMode = false);
+ static String8 getFormattedCurrentTime();
+
+ /**
+ * Get the camera eviction priority from the current process state given by ActivityManager.
+ */
+ static int getCameraPriorityFromProcState(int procState);
+
+ static status_t makeClient(const sp<CameraService>& cameraService,
+ const sp<IInterface>& cameraCb, const String16& packageName, const String8& cameraId,
+ int facing, int clientPid, uid_t clientUid, int servicePid, bool legacyMode,
+ int halVersion, int deviceVersion, apiLevel effectiveApiLevel,
+ /*out*/sp<BasicClient>* client);
};
+template<class Func>
+void CameraService::CameraState::updateStatus(ICameraServiceListener::Status status,
+ const String8& cameraId,
+ std::initializer_list<ICameraServiceListener::Status> rejectSourceStates,
+ Func onStatusUpdatedLocked) {
+ Mutex::Autolock lock(mStatusLock);
+ ICameraServiceListener::Status oldStatus = mStatus;
+ mStatus = status;
+
+ if (oldStatus == status) {
+ return;
+ }
+
+ ALOGV("%s: Status has changed for camera ID %s from %#x to %#x", __FUNCTION__,
+ cameraId.string(), oldStatus, status);
+
+ if (oldStatus == ICameraServiceListener::STATUS_NOT_PRESENT &&
+ (status != ICameraServiceListener::STATUS_PRESENT &&
+ status != ICameraServiceListener::STATUS_ENUMERATING)) {
+
+ ALOGW("%s: From NOT_PRESENT can only transition into PRESENT or ENUMERATING",
+ __FUNCTION__);
+ mStatus = oldStatus;
+ return;
+ }
+
+ /**
+ * Sometimes we want to conditionally do a transition.
+ * For example if a client disconnects, we want to go to PRESENT
+ * only if we weren't already in NOT_PRESENT or ENUMERATING.
+ */
+ for (auto& rejectStatus : rejectSourceStates) {
+ if (oldStatus == rejectStatus) {
+ ALOGV("%s: Rejecting status transition for Camera ID %s, since the source "
+ "state was was in one of the bad states.", __FUNCTION__, cameraId.string());
+ mStatus = oldStatus;
+ return;
+ }
+ }
+
+ onStatusUpdatedLocked(cameraId, status);
+}
+
+
+template<class CALLBACK, class CLIENT>
+status_t CameraService::connectHelper(const sp<CALLBACK>& cameraCb, const String8& cameraId,
+ int halVersion, const String16& clientPackageName, int clientUid,
+ apiLevel effectiveApiLevel, bool legacyMode, bool shimUpdateOnly,
+ /*out*/sp<CLIENT>& device) {
+ status_t ret = NO_ERROR;
+ String8 clientName8(clientPackageName);
+ int clientPid = getCallingPid();
+
+ ALOGI("CameraService::connect call E (PID %d \"%s\", camera ID %s) for HAL version %d and "
+ "Camera API version %d", clientPid, clientName8.string(), cameraId.string(),
+ halVersion, static_cast<int>(effectiveApiLevel));
+
+ sp<CLIENT> client = nullptr;
+ {
+ // Acquire mServiceLock and prevent other clients from connecting
+ std::unique_ptr<AutoConditionLock> lock =
+ AutoConditionLock::waitAndAcquire(mServiceLockWrapper, DEFAULT_CONNECT_TIMEOUT_NS);
+
+ if (lock == nullptr) {
+ ALOGE("CameraService::connect X (PID %d) rejected (too many other clients connecting)."
+ , clientPid);
+ return -EBUSY;
+ }
+
+ // Enforce client permissions and do basic sanity checks
+ if((ret = validateConnectLocked(cameraId, /*inout*/clientUid)) != NO_ERROR) {
+ return ret;
+ }
+ mLastUserId = multiuser_get_user_id(clientUid);
+
+ // Check the shim parameters after acquiring lock, if they have already been updated and
+ // we were doing a shim update, return immediately
+ if (shimUpdateOnly) {
+ auto cameraState = getCameraState(cameraId);
+ if (cameraState != nullptr) {
+ if (!cameraState->getShimParams().isEmpty()) return NO_ERROR;
+ }
+ }
+
+ sp<BasicClient> clientTmp = nullptr;
+ std::shared_ptr<resource_policy::ClientDescriptor<String8, sp<BasicClient>>> partial;
+ if ((ret = handleEvictionsLocked(cameraId, clientPid, effectiveApiLevel,
+ IInterface::asBinder(cameraCb), clientName8, /*out*/&clientTmp,
+ /*out*/&partial)) != NO_ERROR) {
+ return ret;
+ }
+
+ if (clientTmp.get() != nullptr) {
+ // Handle special case for API1 MediaRecorder where the existing client is returned
+ device = static_cast<CLIENT*>(clientTmp.get());
+ return NO_ERROR;
+ }
+
+ // give flashlight a chance to close devices if necessary.
+ mFlashlight->prepareDeviceOpen(cameraId);
+
+ // TODO: Update getDeviceVersion + HAL interface to use strings for Camera IDs
+ int id = cameraIdToInt(cameraId);
+ if (id == -1) {
+ ALOGE("%s: Invalid camera ID %s, cannot get device version from HAL.", __FUNCTION__,
+ cameraId.string());
+ return BAD_VALUE;
+ }
+
+ int facing = -1;
+ int deviceVersion = getDeviceVersion(id, /*out*/&facing);
+ sp<BasicClient> tmp = nullptr;
+ if((ret = makeClient(this, cameraCb, clientPackageName, cameraId, facing, clientPid,
+ clientUid, getpid(), legacyMode, halVersion, deviceVersion, effectiveApiLevel,
+ /*out*/&tmp)) != NO_ERROR) {
+ return ret;
+ }
+ client = static_cast<CLIENT*>(tmp.get());
+
+ LOG_ALWAYS_FATAL_IF(client.get() == nullptr, "%s: CameraService in invalid state",
+ __FUNCTION__);
+
+ if ((ret = client->initialize(mModule)) != OK) {
+ ALOGE("%s: Could not initialize client from HAL module.", __FUNCTION__);
+ return ret;
+ }
+
+ sp<IBinder> remoteCallback = client->getRemote();
+ if (remoteCallback != nullptr) {
+ remoteCallback->linkToDeath(this);
+ }
+
+ // Update shim paremeters for legacy clients
+ if (effectiveApiLevel == API_1) {
+ // Assume we have always received a Client subclass for API1
+ sp<Client> shimClient = reinterpret_cast<Client*>(client.get());
+ String8 rawParams = shimClient->getParameters();
+ CameraParameters params(rawParams);
+
+ auto cameraState = getCameraState(cameraId);
+ if (cameraState != nullptr) {
+ cameraState->setShimParams(params);
+ } else {
+ ALOGE("%s: Cannot update shim parameters for camera %s, no such device exists.",
+ __FUNCTION__, cameraId.string());
+ }
+ }
+
+ if (shimUpdateOnly) {
+ // If only updating legacy shim parameters, immediately disconnect client
+ mServiceLock.unlock();
+ client->disconnect();
+ mServiceLock.lock();
+ } else {
+ // Otherwise, add client to active clients list
+ finishConnectLocked(client, partial);
+ }
+ } // lock is destroyed, allow further connect calls
+
+ // Important: release the mutex here so the client can call back into the service from its
+ // destructor (can be at the end of the call)
+ device = client;
+ return NO_ERROR;
+}
+
} // namespace android
#endif
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 0ed5586..f53f425 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -67,7 +67,7 @@
mLegacyMode = legacyMode;
}
-status_t Camera2Client::initialize(camera_module_t *module)
+status_t Camera2Client::initialize(CameraModule *module)
{
ATRACE_CALL();
ALOGV("%s: Initializing client for camera %d", __FUNCTION__, mCameraId);
@@ -121,7 +121,8 @@
}
case CAMERA_DEVICE_API_VERSION_3_0:
case CAMERA_DEVICE_API_VERSION_3_1:
- case CAMERA_DEVICE_API_VERSION_3_2: {
+ case CAMERA_DEVICE_API_VERSION_3_2:
+ case CAMERA_DEVICE_API_VERSION_3_3: {
sp<ZslProcessor3> zslProc =
new ZslProcessor3(this, mCaptureSequencer);
mZslProcessor = zslProc;
@@ -163,11 +164,9 @@
status_t Camera2Client::dump(int fd, const Vector<String16>& args) {
String8 result;
- result.appendFormat("Client2[%d] (%p) Client: %s PID: %d, dump:\n",
- mCameraId,
+ result.appendFormat("Client2[%d] (%p) PID: %d, dump:\n", mCameraId,
(getRemoteCallback() != NULL ?
(IInterface::asBinder(getRemoteCallback()).get()) : NULL),
- String8(mClientPackageName).string(),
mClientPid);
result.append(" State: ");
#define CASE_APPEND_ENUM(x) case x: result.append(#x "\n"); break;
@@ -1959,7 +1958,7 @@
return width * height * 2;
case HAL_PIXEL_FORMAT_RGBA_8888:
return width * height * 4;
- case HAL_PIXEL_FORMAT_RAW_SENSOR:
+ case HAL_PIXEL_FORMAT_RAW16:
return width * height * 2;
default:
ALOGE("%s: Unknown preview format: %x",
diff --git a/services/camera/libcameraservice/api1/Camera2Client.h b/services/camera/libcameraservice/api1/Camera2Client.h
index d68bb29..5a8241f 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.h
+++ b/services/camera/libcameraservice/api1/Camera2Client.h
@@ -94,7 +94,7 @@
virtual ~Camera2Client();
- status_t initialize(camera_module_t *module);
+ status_t initialize(CameraModule *module);
virtual status_t dump(int fd, const Vector<String16>& args);
diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index bbb2fe0..e552633 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -59,7 +59,7 @@
LOG1("CameraClient::CameraClient X (pid %d, id %d)", callingPid, cameraId);
}
-status_t CameraClient::initialize(camera_module_t *module) {
+status_t CameraClient::initialize(CameraModule *module) {
int callingPid = getCallingPid();
status_t res;
@@ -75,7 +75,7 @@
snprintf(camera_device_name, sizeof(camera_device_name), "%d", mCameraId);
mHardware = new CameraHardwareInterface(camera_device_name);
- res = mHardware->initialize(&module->common);
+ res = mHardware->initialize(module);
if (res != OK) {
ALOGE("%s: Camera %d: unable to initialize device: %s (%d)",
__FUNCTION__, mCameraId, strerror(-res), res);
@@ -99,12 +99,7 @@
// tear down the client
CameraClient::~CameraClient() {
- // this lock should never be NULL
- Mutex* lock = mCameraService->getClientLockById(mCameraId);
- lock->lock();
mDestructionStarted = true;
- // client will not be accessed from callback. should unlock to prevent dead-lock in disconnect
- lock->unlock();
int callingPid = getCallingPid();
LOG1("CameraClient::~CameraClient E (pid %d, this %p)", callingPid, this);
@@ -116,11 +111,11 @@
const size_t SIZE = 256;
char buffer[SIZE];
- size_t len = snprintf(buffer, SIZE, "Client[%d] (%p) PID: %d\n",
+ size_t len = snprintf(buffer, SIZE, "Client[%d] (%p) with UID %d\n",
mCameraId,
(getRemoteCallback() != NULL ?
IInterface::asBinder(getRemoteCallback()).get() : NULL),
- mClientPid);
+ mClientUid);
len = (len > SIZE - 1) ? SIZE - 1 : len;
write(fd, buffer, len);
@@ -677,6 +672,13 @@
LOG1("lockIfMessageWanted(%d): waited for %d ms",
msgType, sleepCount * CHECK_MESSAGE_INTERVAL);
}
+
+ // If messages are no longer enabled after acquiring lock, release and drop message
+ if ((mMsgEnabled & msgType) == 0) {
+ mLock.unlock();
+ break;
+ }
+
return true;
}
if (sleepCount++ == 0) {
@@ -702,26 +704,13 @@
// (others) c->dataCallback
// dataCallbackTimestamp
// (others) c->dataCallbackTimestamp
-//
-// NOTE: the *Callback functions grab mLock of the client before passing
-// control to handle* functions. So the handle* functions must release the
-// lock before calling the ICameraClient's callbacks, so those callbacks can
-// invoke methods in the Client class again (For example, the preview frame
-// callback may want to releaseRecordingFrame). The handle* functions must
-// release the lock after all accesses to member variables, so it must be
-// handled very carefully.
void CameraClient::notifyCallback(int32_t msgType, int32_t ext1,
int32_t ext2, void* user) {
LOG2("notifyCallback(%d)", msgType);
- Mutex* lock = getClientLockFromCookie(user);
- if (lock == NULL) return;
- Mutex::Autolock alock(*lock);
-
- CameraClient* client =
- static_cast<CameraClient*>(getClientFromCookie(user));
- if (client == NULL) return;
+ sp<CameraClient> client = static_cast<CameraClient*>(getClientFromCookie(user).get());
+ if (client.get() == nullptr) return;
if (!client->lockIfMessageWanted(msgType)) return;
@@ -740,13 +729,8 @@
const sp<IMemory>& dataPtr, camera_frame_metadata_t *metadata, void* user) {
LOG2("dataCallback(%d)", msgType);
- Mutex* lock = getClientLockFromCookie(user);
- if (lock == NULL) return;
- Mutex::Autolock alock(*lock);
-
- CameraClient* client =
- static_cast<CameraClient*>(getClientFromCookie(user));
- if (client == NULL) return;
+ sp<CameraClient> client = static_cast<CameraClient*>(getClientFromCookie(user).get());
+ if (client.get() == nullptr) return;
if (!client->lockIfMessageWanted(msgType)) return;
if (dataPtr == 0 && metadata == NULL) {
@@ -778,13 +762,8 @@
int32_t msgType, const sp<IMemory>& dataPtr, void* user) {
LOG2("dataCallbackTimestamp(%d)", msgType);
- Mutex* lock = getClientLockFromCookie(user);
- if (lock == NULL) return;
- Mutex::Autolock alock(*lock);
-
- CameraClient* client =
- static_cast<CameraClient*>(getClientFromCookie(user));
- if (client == NULL) return;
+ sp<CameraClient> client = static_cast<CameraClient*>(getClientFromCookie(user).get());
+ if (client.get() == nullptr) return;
if (!client->lockIfMessageWanted(msgType)) return;
diff --git a/services/camera/libcameraservice/api1/CameraClient.h b/services/camera/libcameraservice/api1/CameraClient.h
index 63a9d0f..95616b2 100644
--- a/services/camera/libcameraservice/api1/CameraClient.h
+++ b/services/camera/libcameraservice/api1/CameraClient.h
@@ -68,7 +68,7 @@
bool legacyMode = false);
~CameraClient();
- status_t initialize(camera_module_t *module);
+ status_t initialize(CameraModule *module);
status_t dump(int fd, const Vector<String16>& args);
diff --git a/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp b/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp
index eadaa00..5c8f750 100644
--- a/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/CallbackProcessor.cpp
@@ -154,8 +154,8 @@
params.previewWidth, params.previewHeight,
callbackFormat, params.previewFormat);
res = device->createStream(mCallbackWindow,
- params.previewWidth, params.previewHeight,
- callbackFormat, &mCallbackStreamId);
+ params.previewWidth, params.previewHeight, callbackFormat,
+ HAL_DATASPACE_JFIF, CAMERA3_STREAM_ROTATION_0, &mCallbackStreamId);
if (res != OK) {
ALOGE("%s: Camera %d: Can't create output stream for callbacks: "
"%s (%d)", __FUNCTION__, mId,
diff --git a/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp b/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
index 2772267..34798bf 100644
--- a/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/JpegProcessor.cpp
@@ -145,7 +145,8 @@
// Create stream for HAL production
res = device->createStream(mCaptureWindow,
params.pictureWidth, params.pictureHeight,
- HAL_PIXEL_FORMAT_BLOB, &mCaptureStreamId);
+ HAL_PIXEL_FORMAT_BLOB, HAL_DATASPACE_JFIF,
+ CAMERA3_STREAM_ROTATION_0, &mCaptureStreamId);
if (res != OK) {
ALOGE("%s: Camera %d: Can't create output stream for capture: "
"%s (%d)", __FUNCTION__, mId,
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index 4f4cfb0..6b0f8b5 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -65,15 +65,29 @@
const Size MAX_PREVIEW_SIZE = { MAX_PREVIEW_WIDTH, MAX_PREVIEW_HEIGHT };
// Treat the H.264 max size as the max supported video size.
MediaProfiles *videoEncoderProfiles = MediaProfiles::getInstance();
- int32_t maxVideoWidth = videoEncoderProfiles->getVideoEncoderParamByName(
- "enc.vid.width.max", VIDEO_ENCODER_H264);
- int32_t maxVideoHeight = videoEncoderProfiles->getVideoEncoderParamByName(
- "enc.vid.height.max", VIDEO_ENCODER_H264);
- const Size MAX_VIDEO_SIZE = {maxVideoWidth, maxVideoHeight};
+ Vector<video_encoder> encoders = videoEncoderProfiles->getVideoEncoders();
+ int32_t maxVideoWidth = 0;
+ int32_t maxVideoHeight = 0;
+ for (size_t i = 0; i < encoders.size(); i++) {
+ int width = videoEncoderProfiles->getVideoEncoderParamByName(
+ "enc.vid.width.max", encoders[i]);
+ int height = videoEncoderProfiles->getVideoEncoderParamByName(
+ "enc.vid.height.max", encoders[i]);
+ // Treat width/height separately here to handle the case where different
+ // profile might report max size of different aspect ratio
+ if (width > maxVideoWidth) {
+ maxVideoWidth = width;
+ }
+ if (height > maxVideoHeight) {
+ maxVideoHeight = height;
+ }
+ }
+ // This is just an upper bound and may not be an actually valid video size
+ const Size VIDEO_SIZE_UPPER_BOUND = {maxVideoWidth, maxVideoHeight};
res = getFilteredSizes(MAX_PREVIEW_SIZE, &availablePreviewSizes);
if (res != OK) return res;
- res = getFilteredSizes(MAX_VIDEO_SIZE, &availableVideoSizes);
+ res = getFilteredSizes(VIDEO_SIZE_UPPER_BOUND, &availableVideoSizes);
if (res != OK) return res;
// Select initial preview and video size that's under the initial bound and
@@ -182,9 +196,9 @@
supportedPreviewFormats +=
CameraParameters::PIXEL_FORMAT_YUV420SP;
break;
- // Not advertizing JPEG, RAW_SENSOR, etc, for preview formats
+ // Not advertizing JPEG, RAW16, etc, for preview formats
case HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED:
- case HAL_PIXEL_FORMAT_RAW_SENSOR:
+ case HAL_PIXEL_FORMAT_RAW16:
case HAL_PIXEL_FORMAT_BLOB:
addComma = false;
break;
@@ -2253,7 +2267,7 @@
case HAL_PIXEL_FORMAT_RGBA_8888: // RGBA8888
fmt = CameraParameters::PIXEL_FORMAT_RGBA8888;
break;
- case HAL_PIXEL_FORMAT_RAW_SENSOR:
+ case HAL_PIXEL_FORMAT_RAW16:
ALOGW("Raw sensor preview format requested.");
fmt = CameraParameters::PIXEL_FORMAT_BAYER_RGGB;
break;
diff --git a/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp b/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp
index 146d572..b6071f6 100644
--- a/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp
@@ -182,7 +182,8 @@
if (mPreviewStreamId == NO_STREAM) {
res = device->createStream(mPreviewWindow,
params.previewWidth, params.previewHeight,
- CAMERA2_HAL_PIXEL_FORMAT_OPAQUE, &mPreviewStreamId);
+ CAMERA2_HAL_PIXEL_FORMAT_OPAQUE, HAL_DATASPACE_UNKNOWN,
+ CAMERA3_STREAM_ROTATION_0, &mPreviewStreamId);
if (res != OK) {
ALOGE("%s: Camera %d: Unable to create preview stream: %s (%d)",
__FUNCTION__, mId, strerror(-res), res);
@@ -421,9 +422,12 @@
if (mRecordingStreamId == NO_STREAM) {
mRecordingFrameCount = 0;
+ // Selecting BT.709 colorspace by default
+ // TODO: Wire this in from encoder side
res = device->createStream(mRecordingWindow,
params.videoWidth, params.videoHeight,
- CAMERA2_HAL_PIXEL_FORMAT_OPAQUE, &mRecordingStreamId);
+ CAMERA2_HAL_PIXEL_FORMAT_OPAQUE, HAL_DATASPACE_BT709,
+ CAMERA3_STREAM_ROTATION_0, &mRecordingStreamId);
if (res != OK) {
ALOGE("%s: Camera %d: Can't create output stream for recording: "
"%s (%d)", __FUNCTION__, mId,
diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp b/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
index 186ce6c..a03f9c7 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
@@ -185,8 +185,8 @@
(int)CAMERA2_HAL_PIXEL_FORMAT_ZSL :
(int)HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;
res = device->createStream(mZslWindow,
- params.fastInfo.arrayWidth, params.fastInfo.arrayHeight,
- streamType, &mZslStreamId);
+ params.fastInfo.arrayWidth, params.fastInfo.arrayHeight, streamType,
+ HAL_DATASPACE_UNKNOWN, CAMERA3_STREAM_ROTATION_0, &mZslStreamId);
if (res != OK) {
ALOGE("%s: Camera %d: Can't create output stream for ZSL: "
"%s (%d)", __FUNCTION__, mId,
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 6a1ee44..8587e0e 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -71,7 +71,7 @@
ALOGI("CameraDeviceClient %d: Opened", cameraId);
}
-status_t CameraDeviceClient::initialize(camera_module_t *module)
+status_t CameraDeviceClient::initialize(CameraModule *module)
{
ATRACE_CALL();
status_t res;
@@ -314,17 +314,17 @@
return res;
}
-status_t CameraDeviceClient::createStream(int width, int height, int format,
- const sp<IGraphicBufferProducer>& bufferProducer)
+status_t CameraDeviceClient::createStream(const OutputConfiguration &outputConfiguration)
{
ATRACE_CALL();
- ALOGV("%s (w = %d, h = %d, f = 0x%x)", __FUNCTION__, width, height, format);
status_t res;
if ( (res = checkPid(__FUNCTION__) ) != OK) return res;
Mutex::Autolock icl(mBinderSerializationLock);
+
+ sp<IGraphicBufferProducer> bufferProducer = outputConfiguration.getGraphicBufferProducer();
if (bufferProducer == NULL) {
ALOGE("%s: bufferProducer must not be null", __FUNCTION__);
return BAD_VALUE;
@@ -370,7 +370,8 @@
sp<IBinder> binder = IInterface::asBinder(bufferProducer);
sp<ANativeWindow> anw = new Surface(bufferProducer, useAsync);
- // TODO: remove w,h,f since we are ignoring them
+ int width, height, format;
+ android_dataspace dataSpace;
if ((res = anw->query(anw.get(), NATIVE_WINDOW_WIDTH, &width)) != OK) {
ALOGE("%s: Camera %d: Failed to query Surface width", __FUNCTION__,
@@ -387,6 +388,12 @@
mCameraId);
return res;
}
+ if ((res = anw->query(anw.get(), NATIVE_WINDOW_DEFAULT_DATASPACE,
+ reinterpret_cast<int*>(&dataSpace))) != OK) {
+ ALOGE("%s: Camera %d: Failed to query Surface dataSpace", __FUNCTION__,
+ mCameraId);
+ return res;
+ }
// FIXME: remove this override since the default format should be
// IMPLEMENTATION_DEFINED. b/9487482
@@ -399,14 +406,17 @@
// Round dimensions to the nearest dimensions available for this format
if (flexibleConsumer && !CameraDeviceClient::roundBufferDimensionNearest(width, height,
- format, mDevice->info(), /*out*/&width, /*out*/&height)) {
+ format, dataSpace, mDevice->info(), /*out*/&width, /*out*/&height)) {
ALOGE("%s: No stream configurations with the format %#x defined, failed to create stream.",
__FUNCTION__, format);
return BAD_VALUE;
}
int streamId = -1;
- res = mDevice->createStream(anw, width, height, format, &streamId);
+ res = mDevice->createStream(anw, width, height, format, dataSpace,
+ static_cast<camera3_stream_rotation_t>
+ (outputConfiguration.getRotation()),
+ &streamId);
if (res == OK) {
mStreamMap.add(binder, streamId);
@@ -441,10 +451,12 @@
bool CameraDeviceClient::roundBufferDimensionNearest(int32_t width, int32_t height,
- int32_t format, const CameraMetadata& info,
+ int32_t format, android_dataspace dataSpace, const CameraMetadata& info,
/*out*/int32_t* outWidth, /*out*/int32_t* outHeight) {
camera_metadata_ro_entry streamConfigs =
+ (dataSpace == HAL_DATASPACE_DEPTH) ?
+ info.find(ANDROID_DEPTH_AVAILABLE_DEPTH_STREAM_CONFIGURATIONS) :
info.find(ANDROID_SCALER_AVAILABLE_STREAM_CONFIGURATIONS);
int32_t bestWidth = -1;
@@ -586,9 +598,7 @@
mCameraId,
(getRemoteCallback() != NULL ?
IInterface::asBinder(getRemoteCallback()).get() : NULL) );
- result.appendFormat(" Current client: %s (PID %d, UID %u)\n",
- String8(mClientPackageName).string(),
- mClientPid, mClientUid);
+ result.appendFormat(" Current client UID %u\n", mClientUid);
result.append(" State:\n");
result.appendFormat(" Request ID counter: %d\n", mRequestIdCounter);
@@ -635,9 +645,6 @@
}
}
-// TODO: refactor the code below this with IProCameraUser.
-// it's 100% copy-pasted, so lets not change it right now to make it easier.
-
void CameraDeviceClient::detachDevice() {
if (mDevice == 0) return;
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.h b/services/camera/libcameraservice/api2/CameraDeviceClient.h
index 84e46b7..a3dbb90 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.h
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.h
@@ -19,6 +19,7 @@
#include <camera/camera2/ICameraDeviceUser.h>
#include <camera/camera2/ICameraDeviceCallbacks.h>
+#include <camera/camera2/OutputConfiguration.h>
#include "CameraService.h"
#include "common/FrameProcessorBase.h"
@@ -83,11 +84,7 @@
// Returns -EBUSY if device is not idle
virtual status_t deleteStream(int streamId);
- virtual status_t createStream(
- int width,
- int height,
- int format,
- const sp<IGraphicBufferProducer>& bufferProducer);
+ virtual status_t createStream(const OutputConfiguration &outputConfiguration);
// Create a request object from a template.
virtual status_t createDefaultRequest(int templateId,
@@ -119,7 +116,7 @@
int servicePid);
virtual ~CameraDeviceClient();
- virtual status_t initialize(camera_module_t *module);
+ virtual status_t initialize(CameraModule *module);
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -161,7 +158,8 @@
// a width <= ROUNDING_WIDTH_CAP
static const int32_t ROUNDING_WIDTH_CAP = 1080;
static bool roundBufferDimensionNearest(int32_t width, int32_t height, int32_t format,
- const CameraMetadata& info, /*out*/int32_t* outWidth, /*out*/int32_t* outHeight);
+ android_dataspace dataSpace, const CameraMetadata& info,
+ /*out*/int32_t* outWidth, /*out*/int32_t* outHeight);
// IGraphicsBufferProducer binder -> Stream ID
KeyedVector<sp<IBinder>, int> mStreamMap;
diff --git a/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp b/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp
deleted file mode 100644
index 59e5083..0000000
--- a/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp
+++ /dev/null
@@ -1,444 +0,0 @@
-/*
- * Copyright (C) 2013 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 "ProCamera2Client"
-#define ATRACE_TAG ATRACE_TAG_CAMERA
-//#define LOG_NDEBUG 0
-
-#include <utils/Log.h>
-#include <utils/Trace.h>
-
-#include <cutils/properties.h>
-#include <gui/Surface.h>
-#include <gui/Surface.h>
-
-#include "api_pro/ProCamera2Client.h"
-#include "common/CameraDeviceBase.h"
-
-namespace android {
-using namespace camera2;
-
-// Interface used by CameraService
-
-ProCamera2Client::ProCamera2Client(const sp<CameraService>& cameraService,
- const sp<IProCameraCallbacks>& remoteCallback,
- const String16& clientPackageName,
- int cameraId,
- int cameraFacing,
- int clientPid,
- uid_t clientUid,
- int servicePid) :
- Camera2ClientBase(cameraService, remoteCallback, clientPackageName,
- cameraId, cameraFacing, clientPid, clientUid, servicePid)
-{
- ATRACE_CALL();
- ALOGI("ProCamera %d: Opened", cameraId);
-
- mExclusiveLock = false;
-}
-
-status_t ProCamera2Client::initialize(camera_module_t *module)
-{
- ATRACE_CALL();
- status_t res;
-
- res = Camera2ClientBase::initialize(module);
- if (res != OK) {
- return res;
- }
-
- String8 threadName;
- mFrameProcessor = new FrameProcessorBase(mDevice);
- threadName = String8::format("PC2-%d-FrameProc", mCameraId);
- mFrameProcessor->run(threadName.string());
-
- mFrameProcessor->registerListener(FRAME_PROCESSOR_LISTENER_MIN_ID,
- FRAME_PROCESSOR_LISTENER_MAX_ID,
- /*listener*/this);
-
- return OK;
-}
-
-ProCamera2Client::~ProCamera2Client() {
-}
-
-status_t ProCamera2Client::exclusiveTryLock() {
- ATRACE_CALL();
- ALOGV("%s", __FUNCTION__);
-
- Mutex::Autolock icl(mBinderSerializationLock);
- SharedCameraCallbacks::Lock l(mSharedCameraCallbacks);
-
- if (!mDevice.get()) return PERMISSION_DENIED;
-
- if (!mExclusiveLock) {
- mExclusiveLock = true;
-
- if (mRemoteCallback != NULL) {
- mRemoteCallback->onLockStatusChanged(
- IProCameraCallbacks::LOCK_ACQUIRED);
- }
-
- ALOGV("%s: exclusive lock acquired", __FUNCTION__);
-
- return OK;
- }
-
- // TODO: have a PERMISSION_DENIED case for when someone else owns the lock
-
- // don't allow recursive locking
- ALOGW("%s: exclusive lock already exists - recursive locking is not"
- "allowed", __FUNCTION__);
-
- return ALREADY_EXISTS;
-}
-
-status_t ProCamera2Client::exclusiveLock() {
- ATRACE_CALL();
- ALOGV("%s", __FUNCTION__);
-
- Mutex::Autolock icl(mBinderSerializationLock);
- SharedCameraCallbacks::Lock l(mSharedCameraCallbacks);
-
- if (!mDevice.get()) return PERMISSION_DENIED;
-
- /**
- * TODO: this should asynchronously 'wait' until the lock becomes available
- * if another client already has an exclusive lock.
- *
- * once we have proper sharing support this will need to do
- * more than just return immediately
- */
- if (!mExclusiveLock) {
- mExclusiveLock = true;
-
- if (mRemoteCallback != NULL) {
- mRemoteCallback->onLockStatusChanged(IProCameraCallbacks::LOCK_ACQUIRED);
- }
-
- ALOGV("%s: exclusive lock acquired", __FUNCTION__);
-
- return OK;
- }
-
- // don't allow recursive locking
- ALOGW("%s: exclusive lock already exists - recursive locking is not allowed"
- , __FUNCTION__);
- return ALREADY_EXISTS;
-}
-
-status_t ProCamera2Client::exclusiveUnlock() {
- ATRACE_CALL();
- ALOGV("%s", __FUNCTION__);
-
- Mutex::Autolock icl(mBinderSerializationLock);
- SharedCameraCallbacks::Lock l(mSharedCameraCallbacks);
-
- // don't allow unlocking if we have no lock
- if (!mExclusiveLock) {
- ALOGW("%s: cannot unlock, no lock was held in the first place",
- __FUNCTION__);
- return BAD_VALUE;
- }
-
- mExclusiveLock = false;
- if (mRemoteCallback != NULL ) {
- mRemoteCallback->onLockStatusChanged(
- IProCameraCallbacks::LOCK_RELEASED);
- }
- ALOGV("%s: exclusive lock released", __FUNCTION__);
-
- return OK;
-}
-
-bool ProCamera2Client::hasExclusiveLock() {
- Mutex::Autolock icl(mBinderSerializationLock);
- return mExclusiveLock;
-}
-
-void ProCamera2Client::onExclusiveLockStolen() {
- ALOGV("%s: ProClient lost exclusivity (id %d)",
- __FUNCTION__, mCameraId);
-
- Mutex::Autolock icl(mBinderSerializationLock);
- SharedCameraCallbacks::Lock l(mSharedCameraCallbacks);
-
- if (mExclusiveLock && mRemoteCallback.get() != NULL) {
- mRemoteCallback->onLockStatusChanged(
- IProCameraCallbacks::LOCK_STOLEN);
- }
-
- mExclusiveLock = false;
-
- //TODO: we should not need to detach the device, merely reset it.
- detachDevice();
-}
-
-status_t ProCamera2Client::submitRequest(camera_metadata_t* request,
- bool streaming) {
- ATRACE_CALL();
- ALOGV("%s", __FUNCTION__);
-
- Mutex::Autolock icl(mBinderSerializationLock);
-
- if (!mDevice.get()) return DEAD_OBJECT;
-
- if (!mExclusiveLock) {
- return PERMISSION_DENIED;
- }
-
- CameraMetadata metadata(request);
-
- if (!enforceRequestPermissions(metadata)) {
- return PERMISSION_DENIED;
- }
-
- if (streaming) {
- return mDevice->setStreamingRequest(metadata);
- } else {
- return mDevice->capture(metadata);
- }
-
- // unreachable. thx gcc for a useless warning
- return OK;
-}
-
-status_t ProCamera2Client::cancelRequest(int requestId) {
- (void)requestId;
- ATRACE_CALL();
- ALOGV("%s", __FUNCTION__);
-
- Mutex::Autolock icl(mBinderSerializationLock);
-
- if (!mDevice.get()) return DEAD_OBJECT;
-
- if (!mExclusiveLock) {
- return PERMISSION_DENIED;
- }
-
- // TODO: implement
- ALOGE("%s: not fully implemented yet", __FUNCTION__);
- return INVALID_OPERATION;
-}
-
-status_t ProCamera2Client::deleteStream(int streamId) {
- ATRACE_CALL();
- ALOGV("%s (streamId = 0x%x)", __FUNCTION__, streamId);
-
- status_t res;
- if ( (res = checkPid(__FUNCTION__) ) != OK) return res;
-
- Mutex::Autolock icl(mBinderSerializationLock);
-
- if (!mDevice.get()) return DEAD_OBJECT;
- mDevice->clearStreamingRequest();
-
- status_t code;
- if ((code = mDevice->waitUntilDrained()) != OK) {
- ALOGE("%s: waitUntilDrained failed with code 0x%x", __FUNCTION__, code);
- }
-
- return mDevice->deleteStream(streamId);
-}
-
-status_t ProCamera2Client::createStream(int width, int height, int format,
- const sp<IGraphicBufferProducer>& bufferProducer,
- /*out*/
- int* streamId)
-{
- if (streamId) {
- *streamId = -1;
- }
-
- ATRACE_CALL();
- ALOGV("%s (w = %d, h = %d, f = 0x%x)", __FUNCTION__, width, height, format);
-
- status_t res;
- if ( (res = checkPid(__FUNCTION__) ) != OK) return res;
-
- Mutex::Autolock icl(mBinderSerializationLock);
-
- if (!mDevice.get()) return DEAD_OBJECT;
-
- sp<IBinder> binder;
- sp<ANativeWindow> window;
- if (bufferProducer != 0) {
- binder = IInterface::asBinder(bufferProducer);
- window = new Surface(bufferProducer);
- }
-
- return mDevice->createStream(window, width, height, format,
- streamId);
-}
-
-// Create a request object from a template.
-// -- Caller owns the newly allocated metadata
-status_t ProCamera2Client::createDefaultRequest(int templateId,
- /*out*/
- camera_metadata** request)
-{
- ATRACE_CALL();
- ALOGV("%s (templateId = 0x%x)", __FUNCTION__, templateId);
-
- if (request) {
- *request = NULL;
- }
-
- status_t res;
- if ( (res = checkPid(__FUNCTION__) ) != OK) return res;
-
- Mutex::Autolock icl(mBinderSerializationLock);
-
- if (!mDevice.get()) return DEAD_OBJECT;
-
- CameraMetadata metadata;
- if ( (res = mDevice->createDefaultRequest(templateId, &metadata) ) == OK) {
- *request = metadata.release();
- }
-
- return res;
-}
-
-status_t ProCamera2Client::getCameraInfo(int cameraId,
- /*out*/
- camera_metadata** info)
-{
- if (cameraId != mCameraId) {
- return INVALID_OPERATION;
- }
-
- Mutex::Autolock icl(mBinderSerializationLock);
-
- if (!mDevice.get()) return DEAD_OBJECT;
-
- CameraMetadata deviceInfo = mDevice->info();
- *info = deviceInfo.release();
-
- return OK;
-}
-
-status_t ProCamera2Client::dump(int fd, const Vector<String16>& args) {
- String8 result;
- result.appendFormat("ProCamera2Client[%d] (%p) PID: %d, dump:\n",
- mCameraId,
- (getRemoteCallback() != NULL ?
- IInterface::asBinder(getRemoteCallback()).get() : NULL),
- mClientPid);
- result.append(" State:\n");
- write(fd, result.string(), result.size());
-
- // TODO: print dynamic/request section from most recent requests
- mFrameProcessor->dump(fd, args);
- return dumpDevice(fd, args);
-}
-
-// IProCameraUser interface
-
-void ProCamera2Client::detachDevice() {
- if (mDevice == 0) return;
-
- ALOGV("Camera %d: Stopping processors", mCameraId);
-
- mFrameProcessor->removeListener(FRAME_PROCESSOR_LISTENER_MIN_ID,
- FRAME_PROCESSOR_LISTENER_MAX_ID,
- /*listener*/this);
- mFrameProcessor->requestExit();
- ALOGV("Camera %d: Waiting for threads", mCameraId);
- mFrameProcessor->join();
- ALOGV("Camera %d: Disconnecting device", mCameraId);
-
- // WORKAROUND: HAL refuses to disconnect while there's streams in flight
- {
- mDevice->clearStreamingRequest();
-
- status_t code;
- if ((code = mDevice->waitUntilDrained()) != OK) {
- ALOGE("%s: waitUntilDrained failed with code 0x%x", __FUNCTION__,
- code);
- }
- }
-
- Camera2ClientBase::detachDevice();
-}
-
-void ProCamera2Client::onResultAvailable(const CaptureResult& result) {
- ATRACE_CALL();
- ALOGV("%s", __FUNCTION__);
-
- Mutex::Autolock icl(mBinderSerializationLock);
- SharedCameraCallbacks::Lock l(mSharedCameraCallbacks);
-
- if (mRemoteCallback != NULL) {
- CameraMetadata tmp(result.mMetadata);
- camera_metadata_t* meta = tmp.release();
- ALOGV("%s: meta = %p ", __FUNCTION__, meta);
- mRemoteCallback->onResultReceived(result.mResultExtras.requestId, meta);
- tmp.acquire(meta);
- }
-}
-
-bool ProCamera2Client::enforceRequestPermissions(CameraMetadata& metadata) {
-
- const int pid = IPCThreadState::self()->getCallingPid();
- const int selfPid = getpid();
- camera_metadata_entry_t entry;
-
- /**
- * Mixin default important security values
- * - android.led.transmit = defaulted ON
- */
- CameraMetadata staticInfo = mDevice->info();
- entry = staticInfo.find(ANDROID_LED_AVAILABLE_LEDS);
- for(size_t i = 0; i < entry.count; ++i) {
- uint8_t led = entry.data.u8[i];
-
- switch(led) {
- case ANDROID_LED_AVAILABLE_LEDS_TRANSMIT: {
- uint8_t transmitDefault = ANDROID_LED_TRANSMIT_ON;
- if (!metadata.exists(ANDROID_LED_TRANSMIT)) {
- metadata.update(ANDROID_LED_TRANSMIT,
- &transmitDefault, 1);
- }
- break;
- }
- }
- }
-
- // We can do anything!
- if (pid == selfPid) {
- return true;
- }
-
- /**
- * Permission check special fields in the request
- * - android.led.transmit = android.permission.CAMERA_DISABLE_TRANSMIT
- */
- entry = metadata.find(ANDROID_LED_TRANSMIT);
- if (entry.count > 0 && entry.data.u8[0] != ANDROID_LED_TRANSMIT_ON) {
- String16 permissionString =
- String16("android.permission.CAMERA_DISABLE_TRANSMIT_LED");
- if (!checkCallingPermission(permissionString)) {
- const int uid = IPCThreadState::self()->getCallingUid();
- ALOGE("Permission Denial: "
- "can't disable transmit LED pid=%d, uid=%d", pid, uid);
- return false;
- }
- }
-
- return true;
-}
-
-} // namespace android
diff --git a/services/camera/libcameraservice/api_pro/ProCamera2Client.h b/services/camera/libcameraservice/api_pro/ProCamera2Client.h
deleted file mode 100644
index 9d83122..0000000
--- a/services/camera/libcameraservice/api_pro/ProCamera2Client.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- * Copyright (C) 2013 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_SERVERS_CAMERA_PROCAMERA2CLIENT_H
-#define ANDROID_SERVERS_CAMERA_PROCAMERA2CLIENT_H
-
-#include "CameraService.h"
-#include "common/FrameProcessorBase.h"
-#include "common/Camera2ClientBase.h"
-#include "device2/Camera2Device.h"
-#include "camera/CaptureResult.h"
-
-namespace android {
-
-class IMemory;
-/**
- * Implements the binder IProCameraUser API,
- * meant for HAL2-level private API access.
- */
-class ProCamera2Client :
- public Camera2ClientBase<CameraService::ProClient>,
- public camera2::FrameProcessorBase::FilteredListener
-{
-public:
- /**
- * IProCameraUser interface (see IProCameraUser for details)
- */
- virtual status_t exclusiveTryLock();
- virtual status_t exclusiveLock();
- virtual status_t exclusiveUnlock();
-
- virtual bool hasExclusiveLock();
-
- // Note that the callee gets a copy of the metadata.
- virtual int submitRequest(camera_metadata_t* metadata,
- bool streaming = false);
- virtual status_t cancelRequest(int requestId);
-
- virtual status_t deleteStream(int streamId);
-
- virtual status_t createStream(
- int width,
- int height,
- int format,
- const sp<IGraphicBufferProducer>& bufferProducer,
- /*out*/
- int* streamId);
-
- // Create a request object from a template.
- // -- Caller owns the newly allocated metadata
- virtual status_t createDefaultRequest(int templateId,
- /*out*/
- camera_metadata** request);
-
- // Get the static metadata for the camera
- // -- Caller owns the newly allocated metadata
- virtual status_t getCameraInfo(int cameraId,
- /*out*/
- camera_metadata** info);
-
- /**
- * Interface used by CameraService
- */
-
- ProCamera2Client(const sp<CameraService>& cameraService,
- const sp<IProCameraCallbacks>& remoteCallback,
- const String16& clientPackageName,
- int cameraId,
- int cameraFacing,
- int clientPid,
- uid_t clientUid,
- int servicePid);
- virtual ~ProCamera2Client();
-
- virtual status_t initialize(camera_module_t *module);
-
- virtual status_t dump(int fd, const Vector<String16>& args);
-
- // Callbacks from camera service
- virtual void onExclusiveLockStolen();
-
- /**
- * Interface used by independent components of ProCamera2Client.
- */
-
-protected:
- /** FilteredListener implementation **/
- virtual void onResultAvailable(const CaptureResult& result);
-
- virtual void detachDevice();
-
-private:
- /** IProCameraUser interface-related private members */
-
- /** Preview callback related members */
- sp<camera2::FrameProcessorBase> mFrameProcessor;
- static const int32_t FRAME_PROCESSOR_LISTENER_MIN_ID = 0;
- static const int32_t FRAME_PROCESSOR_LISTENER_MAX_ID = 0x7fffffffL;
-
- /** Utility members */
- bool enforceRequestPermissions(CameraMetadata& metadata);
-
- // Whether or not we have an exclusive lock on the device
- // - if no we can't modify the request queue.
- // note that creating/deleting streams we own is still OK
- bool mExclusiveLock;
-};
-
-}; // namespace android
-
-#endif
diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.cpp b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
index 453c8bd..c0c2314 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.cpp
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
@@ -78,7 +78,7 @@
}
template <typename TClientBase>
-status_t Camera2ClientBase<TClientBase>::initialize(camera_module_t *module) {
+status_t Camera2ClientBase<TClientBase>::initialize(CameraModule *module) {
ATRACE_CALL();
ALOGV("%s: Initializing client for camera %d", __FUNCTION__,
TClientBase::mCameraId);
@@ -337,7 +337,6 @@
mRemoteCallback.clear();
}
-template class Camera2ClientBase<CameraService::ProClient>;
template class Camera2ClientBase<CameraService::Client>;
template class Camera2ClientBase<CameraDeviceClientBase>;
diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.h b/services/camera/libcameraservice/common/Camera2ClientBase.h
index e09c1b5..168ea0a 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.h
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.h
@@ -18,6 +18,7 @@
#define ANDROID_SERVERS_CAMERA_CAMERA2CLIENT_BASE_H
#include "common/CameraDeviceBase.h"
+#include "common/CameraModule.h"
#include "camera/CaptureResult.h"
namespace android {
@@ -35,7 +36,7 @@
typedef typename TClientBase::TCamCallbacks TCamCallbacks;
/**
- * Base binder interface (see ICamera/IProCameraUser for details)
+ * Base binder interface (see ICamera/ICameraDeviceUser for details)
*/
virtual status_t connect(const sp<TCamCallbacks>& callbacks);
virtual void disconnect();
@@ -55,7 +56,7 @@
int servicePid);
virtual ~Camera2ClientBase();
- virtual status_t initialize(camera_module_t *module);
+ virtual status_t initialize(CameraModule *module);
virtual status_t dump(int fd, const Vector<String16>& args);
/**
diff --git a/services/camera/libcameraservice/common/CameraDeviceBase.h b/services/camera/libcameraservice/common/CameraDeviceBase.h
index d26e20c..fe55b9e 100644
--- a/services/camera/libcameraservice/common/CameraDeviceBase.h
+++ b/services/camera/libcameraservice/common/CameraDeviceBase.h
@@ -29,6 +29,7 @@
#include "hardware/camera3.h"
#include "camera/CameraMetadata.h"
#include "camera/CaptureResult.h"
+#include "common/CameraModule.h"
namespace android {
@@ -45,7 +46,7 @@
*/
virtual int getId() const = 0;
- virtual status_t initialize(camera_module_t *module) = 0;
+ virtual status_t initialize(CameraModule *module) = 0;
virtual status_t disconnect() = 0;
virtual status_t dump(int fd, const Vector<String16> &args) = 0;
@@ -99,17 +100,14 @@
nsecs_t timeout) = 0;
/**
- * Create an output stream of the requested size and format.
+ * Create an output stream of the requested size, format, rotation and dataspace
*
- * If format is CAMERA2_HAL_PIXEL_FORMAT_OPAQUE, then the HAL device selects
- * an appropriate format; it can be queried with getStreamInfo.
- *
- * If format is HAL_PIXEL_FORMAT_COMPRESSED, the size parameter must be
- * equal to the size in bytes of the buffers to allocate for the stream. For
- * other formats, the size parameter is ignored.
+ * For HAL_PIXEL_FORMAT_BLOB formats, the width and height should be the
+ * logical dimensions of the buffer, not the number of bytes.
*/
virtual status_t createStream(sp<ANativeWindow> consumer,
- uint32_t width, uint32_t height, int format, int *id) = 0;
+ uint32_t width, uint32_t height, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation, int *id) = 0;
/**
* Create an input reprocess stream that uses buffers from an existing
diff --git a/services/camera/libcameraservice/common/CameraModule.cpp b/services/camera/libcameraservice/common/CameraModule.cpp
new file mode 100644
index 0000000..e5b12ae
--- /dev/null
+++ b/services/camera/libcameraservice/common/CameraModule.cpp
@@ -0,0 +1,163 @@
+/*
+ * Copyright (C) 2015 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 "CameraModule"
+//#define LOG_NDEBUG 0
+
+#include "CameraModule.h"
+
+namespace android {
+
+void CameraModule::deriveCameraCharacteristicsKeys(
+ uint32_t deviceVersion, CameraMetadata &chars) {
+ // HAL1 devices should not reach here
+ if (deviceVersion < CAMERA_DEVICE_API_VERSION_2_0) {
+ ALOGV("%s: Cannot derive keys for HAL version < 2.0");
+ return;
+ }
+
+ // Keys added in HAL3.3
+ if (deviceVersion < CAMERA_DEVICE_API_VERSION_3_3) {
+ Vector<uint8_t> controlModes;
+ uint8_t data = ANDROID_CONTROL_AE_LOCK_AVAILABLE_TRUE;
+ chars.update(ANDROID_CONTROL_AE_LOCK_AVAILABLE, &data, /*count*/1);
+ data = ANDROID_CONTROL_AWB_LOCK_AVAILABLE_TRUE;
+ chars.update(ANDROID_CONTROL_AWB_LOCK_AVAILABLE, &data, /*count*/1);
+ controlModes.push(ANDROID_CONTROL_MODE_OFF);
+ controlModes.push(ANDROID_CONTROL_MODE_AUTO);
+ camera_metadata_entry entry = chars.find(ANDROID_CONTROL_AVAILABLE_SCENE_MODES);
+ if (entry.count > 1 || entry.data.u8[0] != ANDROID_CONTROL_SCENE_MODE_DISABLED) {
+ controlModes.push(ANDROID_CONTROL_MODE_USE_SCENE_MODE);
+ }
+ chars.update(ANDROID_CONTROL_AVAILABLE_MODES, controlModes);
+ }
+ return;
+}
+
+CameraModule::CameraModule(camera_module_t *module) {
+ if (module == NULL) {
+ ALOGE("%s: camera hardware module must not be null", __FUNCTION__);
+ assert(0);
+ }
+
+ mModule = module;
+ mCameraInfoMap.setCapacity(getNumberOfCameras());
+}
+
+int CameraModule::getCameraInfo(int cameraId, struct camera_info *info) {
+ Mutex::Autolock lock(mCameraInfoLock);
+ if (cameraId < 0) {
+ ALOGE("%s: Invalid camera ID %d", __FUNCTION__, cameraId);
+ return -EINVAL;
+ }
+
+ // Only override static_camera_characteristics for API2 devices
+ int apiVersion = mModule->common.module_api_version;
+ if (apiVersion < CAMERA_MODULE_API_VERSION_2_0) {
+ return mModule->get_camera_info(cameraId, info);
+ }
+
+ ssize_t index = mCameraInfoMap.indexOfKey(cameraId);
+ if (index == NAME_NOT_FOUND) {
+ // Get camera info from raw module and cache it
+ camera_info rawInfo, cameraInfo;
+ int ret = mModule->get_camera_info(cameraId, &rawInfo);
+ if (ret != 0) {
+ return ret;
+ }
+ CameraMetadata m;
+ m = rawInfo.static_camera_characteristics;
+ deriveCameraCharacteristicsKeys(rawInfo.device_version, m);
+ mCameraCharacteristicsMap.add(cameraId, m);
+ cameraInfo = rawInfo;
+ cameraInfo.static_camera_characteristics =
+ mCameraCharacteristicsMap.valueFor(cameraId).getAndLock();
+ mCameraInfoMap.add(cameraId, cameraInfo);
+ index = mCameraInfoMap.indexOfKey(cameraId);
+ }
+
+ assert(index != NAME_NOT_FOUND);
+ // return the cached camera info
+ *info = mCameraInfoMap[index];
+ return 0;
+}
+
+int CameraModule::open(const char* id, struct hw_device_t** device) {
+ return filterOpenErrorCode(mModule->common.methods->open(&mModule->common, id, device));
+}
+
+int CameraModule::openLegacy(
+ const char* id, uint32_t halVersion, struct hw_device_t** device) {
+ return mModule->open_legacy(&mModule->common, id, halVersion, device);
+}
+
+int CameraModule::getNumberOfCameras() {
+ return mModule->get_number_of_cameras();
+}
+
+int CameraModule::setCallbacks(const camera_module_callbacks_t *callbacks) {
+ return mModule->set_callbacks(callbacks);
+}
+
+bool CameraModule::isVendorTagDefined() {
+ return mModule->get_vendor_tag_ops != NULL;
+}
+
+void CameraModule::getVendorTagOps(vendor_tag_ops_t* ops) {
+ if (mModule->get_vendor_tag_ops) {
+ mModule->get_vendor_tag_ops(ops);
+ }
+}
+
+int CameraModule::setTorchMode(const char* camera_id, bool enable) {
+ return mModule->set_torch_mode(camera_id, enable);
+}
+
+status_t CameraModule::filterOpenErrorCode(status_t err) {
+ switch(err) {
+ case NO_ERROR:
+ case -EBUSY:
+ case -EINVAL:
+ case -EUSERS:
+ return err;
+ default:
+ break;
+ }
+ return -ENODEV;
+}
+
+uint16_t CameraModule::getModuleApiVersion() {
+ return mModule->common.module_api_version;
+}
+
+const char* CameraModule::getModuleName() {
+ return mModule->common.name;
+}
+
+uint16_t CameraModule::getHalApiVersion() {
+ return mModule->common.hal_api_version;
+}
+
+const char* CameraModule::getModuleAuthor() {
+ return mModule->common.author;
+}
+
+void* CameraModule::getDso() {
+ return mModule->common.dso;
+}
+
+}; // namespace android
+
diff --git a/services/camera/libcameraservice/common/CameraModule.h b/services/camera/libcameraservice/common/CameraModule.h
new file mode 100644
index 0000000..e285b21
--- /dev/null
+++ b/services/camera/libcameraservice/common/CameraModule.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (C) 2015 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_SERVERS_CAMERA_CAMERAMODULE_H
+#define ANDROID_SERVERS_CAMERA_CAMERAMODULE_H
+
+#include <hardware/camera.h>
+#include <camera/CameraMetadata.h>
+#include <utils/Mutex.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+/**
+ * A wrapper class for HAL camera module.
+ *
+ * This class wraps camera_module_t returned from HAL to provide a wrapped
+ * get_camera_info implementation which CameraService generates some
+ * camera characteristics keys defined in newer HAL version on an older HAL.
+ */
+class CameraModule {
+public:
+ CameraModule(camera_module_t *module);
+
+ int getCameraInfo(int cameraId, struct camera_info *info);
+ int getNumberOfCameras(void);
+ int open(const char* id, struct hw_device_t** device);
+ int openLegacy(const char* id, uint32_t halVersion, struct hw_device_t** device);
+ int setCallbacks(const camera_module_callbacks_t *callbacks);
+ bool isVendorTagDefined();
+ void getVendorTagOps(vendor_tag_ops_t* ops);
+ int setTorchMode(const char* camera_id, bool enable);
+ uint16_t getModuleApiVersion();
+ const char* getModuleName();
+ uint16_t getHalApiVersion();
+ const char* getModuleAuthor();
+ // Only used by CameraModuleFixture native test. Do NOT use elsewhere.
+ void *getDso();
+
+private:
+ // Derive camera characteristics keys defined after HAL device version
+ static void deriveCameraCharacteristicsKeys(uint32_t deviceVersion, CameraMetadata &chars);
+ status_t filterOpenErrorCode(status_t err);
+
+ camera_module_t *mModule;
+ KeyedVector<int, camera_info> mCameraInfoMap;
+ KeyedVector<int, CameraMetadata> mCameraCharacteristicsMap;
+ Mutex mCameraInfoLock;
+};
+
+} // namespace android
+
+#endif
+
diff --git a/services/camera/libcameraservice/device1/CameraHardwareInterface.h b/services/camera/libcameraservice/device1/CameraHardwareInterface.h
index 1935c2b..7f14cd4 100644
--- a/services/camera/libcameraservice/device1/CameraHardwareInterface.h
+++ b/services/camera/libcameraservice/device1/CameraHardwareInterface.h
@@ -89,24 +89,22 @@
}
}
- status_t initialize(hw_module_t *module)
+ status_t initialize(CameraModule *module)
{
ALOGI("Opening camera %s", mName.string());
- camera_module_t *cameraModule = reinterpret_cast<camera_module_t *>(module);
camera_info info;
- status_t res = cameraModule->get_camera_info(atoi(mName.string()), &info);
+ status_t res = module->getCameraInfo(atoi(mName.string()), &info);
if (res != OK) return res;
int rc = OK;
- if (module->module_api_version >= CAMERA_MODULE_API_VERSION_2_3 &&
+ if (module->getModuleApiVersion() >= CAMERA_MODULE_API_VERSION_2_3 &&
info.device_version > CAMERA_DEVICE_API_VERSION_1_0) {
// Open higher version camera device as HAL1.0 device.
- rc = cameraModule->open_legacy(module, mName.string(),
- CAMERA_DEVICE_API_VERSION_1_0,
- (hw_device_t **)&mDevice);
+ rc = module->openLegacy(mName.string(),
+ CAMERA_DEVICE_API_VERSION_1_0,
+ (hw_device_t **)&mDevice);
} else {
- rc = CameraService::filterOpenErrorCode(module->methods->open(
- module, mName.string(), (hw_device_t **)&mDevice));
+ rc = module->open(mName.string(), (hw_device_t **)&mDevice);
}
if (rc != OK) {
ALOGE("Could not open camera %s: %d", mName.string(), rc);
diff --git a/services/camera/libcameraservice/device2/Camera2Device.cpp b/services/camera/libcameraservice/device2/Camera2Device.cpp
index d1158d6..878986b 100644
--- a/services/camera/libcameraservice/device2/Camera2Device.cpp
+++ b/services/camera/libcameraservice/device2/Camera2Device.cpp
@@ -53,7 +53,7 @@
return mId;
}
-status_t Camera2Device::initialize(camera_module_t *module)
+status_t Camera2Device::initialize(CameraModule *module)
{
ATRACE_CALL();
ALOGV("%s: Initializing device for camera %d", __FUNCTION__, mId);
@@ -68,8 +68,7 @@
camera2_device_t *device;
- res = CameraService::filterOpenErrorCode(module->common.methods->open(
- &module->common, name, reinterpret_cast<hw_device_t**>(&device)));
+ res = module->open(name, reinterpret_cast<hw_device_t**>(&device));
if (res != OK) {
ALOGE("%s: Could not open camera %d: %s (%d)", __FUNCTION__,
@@ -87,7 +86,7 @@
}
camera_info info;
- res = module->get_camera_info(mId, &info);
+ res = module->getCameraInfo(mId, &info);
if (res != OK ) return res;
if (info.device_version != device->common.version) {
@@ -242,7 +241,8 @@
}
status_t Camera2Device::createStream(sp<ANativeWindow> consumer,
- uint32_t width, uint32_t height, int format, int *id) {
+ uint32_t width, uint32_t height, int format,
+ android_dataspace /*dataSpace*/, camera3_stream_rotation_t rotation, int *id) {
ATRACE_CALL();
status_t res;
ALOGV("%s: E", __FUNCTION__);
diff --git a/services/camera/libcameraservice/device2/Camera2Device.h b/services/camera/libcameraservice/device2/Camera2Device.h
index 4def8ae..9b32fa6 100644
--- a/services/camera/libcameraservice/device2/Camera2Device.h
+++ b/services/camera/libcameraservice/device2/Camera2Device.h
@@ -43,7 +43,7 @@
* CameraDevice interface
*/
virtual int getId() const;
- virtual status_t initialize(camera_module_t *module);
+ virtual status_t initialize(CameraModule *module);
virtual status_t disconnect();
virtual status_t dump(int fd, const Vector<String16>& args);
virtual const CameraMetadata& info() const;
@@ -57,7 +57,8 @@
virtual status_t clearStreamingRequest(int64_t *lastFrameNumber = NULL);
virtual status_t waitUntilRequestReceived(int32_t requestId, nsecs_t timeout);
virtual status_t createStream(sp<ANativeWindow> consumer,
- uint32_t width, uint32_t height, int format, int *id);
+ uint32_t width, uint32_t height, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation, int *id);
virtual status_t createReprocessStreamFromStream(int outputId, int *id);
virtual status_t getStreamInfo(int id,
uint32_t *width, uint32_t *height, uint32_t *format);
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 53e6fa9..8236788 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -86,7 +86,7 @@
* CameraDeviceBase interface
*/
-status_t Camera3Device::initialize(camera_module_t *module)
+status_t Camera3Device::initialize(CameraModule *module)
{
ATRACE_CALL();
Mutex::Autolock il(mInterfaceLock);
@@ -106,9 +106,8 @@
camera3_device_t *device;
ATRACE_BEGIN("camera3->open");
- res = CameraService::filterOpenErrorCode(module->common.methods->open(
- &module->common, deviceName.string(),
- reinterpret_cast<hw_device_t**>(&device)));
+ res = module->open(deviceName.string(),
+ reinterpret_cast<hw_device_t**>(&device));
ATRACE_END();
if (res != OK) {
@@ -127,7 +126,7 @@
}
camera_info info;
- res = CameraService::filterGetInfoErrorCode(module->get_camera_info(
+ res = CameraService::filterGetInfoErrorCode(module->getCameraInfo(
mId, &info));
if (res != OK) return res;
@@ -802,12 +801,13 @@
}
status_t Camera3Device::createStream(sp<ANativeWindow> consumer,
- uint32_t width, uint32_t height, int format, int *id) {
+ uint32_t width, uint32_t height, int format, android_dataspace dataSpace,
+ camera3_stream_rotation_t rotation, int *id) {
ATRACE_CALL();
Mutex::Autolock il(mInterfaceLock);
Mutex::Autolock l(mLock);
- ALOGV("Camera %d: Creating new stream %d: %d x %d, format %d",
- mId, mNextStreamId, width, height, format);
+ ALOGV("Camera %d: Creating new stream %d: %d x %d, format %d, dataspace %d rotation %d",
+ mId, mNextStreamId, width, height, format, dataSpace, rotation);
status_t res;
bool wasActive = false;
@@ -847,10 +847,10 @@
}
newStream = new Camera3OutputStream(mNextStreamId, consumer,
- width, height, jpegBufferSize, format);
+ width, height, jpegBufferSize, format, dataSpace, rotation);
} else {
newStream = new Camera3OutputStream(mNextStreamId, consumer,
- width, height, format);
+ width, height, format, dataSpace, rotation);
}
newStream->setStatusTracker(mStatusTracker);
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index ec8dc10..a77548d 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -73,7 +73,7 @@
virtual int getId() const;
// Transitions to idle state on success.
- virtual status_t initialize(camera_module_t *module);
+ virtual status_t initialize(CameraModule *module);
virtual status_t disconnect();
virtual status_t dump(int fd, const Vector<String16> &args);
virtual const CameraMetadata& info() const;
@@ -95,7 +95,8 @@
// If adding streams while actively capturing, will pause device before adding
// stream, reconfiguring device, and unpausing.
virtual status_t createStream(sp<ANativeWindow> consumer,
- uint32_t width, uint32_t height, int format, int *id);
+ uint32_t width, uint32_t height, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation, int *id);
virtual status_t createInputStream(
uint32_t width, uint32_t height, int format,
int *id);
diff --git a/services/camera/libcameraservice/device3/Camera3DummyStream.cpp b/services/camera/libcameraservice/device3/Camera3DummyStream.cpp
index 6656b09..01edfff 100644
--- a/services/camera/libcameraservice/device3/Camera3DummyStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3DummyStream.cpp
@@ -28,7 +28,7 @@
Camera3DummyStream::Camera3DummyStream(int id) :
Camera3IOStreamBase(id, CAMERA3_STREAM_OUTPUT, DUMMY_WIDTH, DUMMY_HEIGHT,
- /*maxSize*/0, DUMMY_FORMAT) {
+ /*maxSize*/0, DUMMY_FORMAT, DUMMY_DATASPACE, DUMMY_ROTATION) {
}
diff --git a/services/camera/libcameraservice/device3/Camera3DummyStream.h b/services/camera/libcameraservice/device3/Camera3DummyStream.h
index 3e42623..d023c57 100644
--- a/services/camera/libcameraservice/device3/Camera3DummyStream.h
+++ b/services/camera/libcameraservice/device3/Camera3DummyStream.h
@@ -75,6 +75,8 @@
static const int DUMMY_WIDTH = 320;
static const int DUMMY_HEIGHT = 240;
static const int DUMMY_FORMAT = HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;
+ static const android_dataspace DUMMY_DATASPACE = HAL_DATASPACE_UNKNOWN;
+ static const camera3_stream_rotation_t DUMMY_ROTATION = CAMERA3_STREAM_ROTATION_0;
static const uint32_t DUMMY_USAGE = GRALLOC_USAGE_HW_COMPOSER;
/**
diff --git a/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp b/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
index cc66459..8696413 100644
--- a/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
+++ b/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
@@ -30,9 +30,10 @@
namespace camera3 {
Camera3IOStreamBase::Camera3IOStreamBase(int id, camera3_stream_type_t type,
- uint32_t width, uint32_t height, size_t maxSize, int format) :
+ uint32_t width, uint32_t height, size_t maxSize, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation) :
Camera3Stream(id, type,
- width, height, maxSize, format),
+ width, height, maxSize, format, dataSpace, rotation),
mTotalBufferCount(0),
mHandoutTotalBufferCount(0),
mHandoutOutputBufferCount(0),
diff --git a/services/camera/libcameraservice/device3/Camera3IOStreamBase.h b/services/camera/libcameraservice/device3/Camera3IOStreamBase.h
index a35c290..abcf2b1 100644
--- a/services/camera/libcameraservice/device3/Camera3IOStreamBase.h
+++ b/services/camera/libcameraservice/device3/Camera3IOStreamBase.h
@@ -33,7 +33,8 @@
public Camera3Stream {
protected:
Camera3IOStreamBase(int id, camera3_stream_type_t type,
- uint32_t width, uint32_t height, size_t maxSize, int format);
+ uint32_t width, uint32_t height, size_t maxSize, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation);
public:
diff --git a/services/camera/libcameraservice/device3/Camera3InputStream.cpp b/services/camera/libcameraservice/device3/Camera3InputStream.cpp
index 9c1e28b..6bf671e 100644
--- a/services/camera/libcameraservice/device3/Camera3InputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3InputStream.cpp
@@ -29,8 +29,8 @@
Camera3InputStream::Camera3InputStream(int id,
uint32_t width, uint32_t height, int format) :
- Camera3IOStreamBase(id, CAMERA3_STREAM_INPUT, width, height,
- /*maxSize*/0, format) {
+ Camera3IOStreamBase(id, CAMERA3_STREAM_INPUT, width, height, /*maxSize*/0,
+ format, HAL_DATASPACE_UNKNOWN, CAMERA3_STREAM_ROTATION_0) {
if (format == HAL_PIXEL_FORMAT_BLOB) {
ALOGE("%s: Bad format, BLOB not supported", __FUNCTION__);
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
index 77ad503..0c739e9 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
@@ -33,9 +33,10 @@
Camera3OutputStream::Camera3OutputStream(int id,
sp<ANativeWindow> consumer,
- uint32_t width, uint32_t height, int format) :
+ uint32_t width, uint32_t height, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation) :
Camera3IOStreamBase(id, CAMERA3_STREAM_OUTPUT, width, height,
- /*maxSize*/0, format),
+ /*maxSize*/0, format, dataSpace, rotation),
mConsumer(consumer),
mTransform(0),
mTraceFirstBuffer(true) {
@@ -48,9 +49,10 @@
Camera3OutputStream::Camera3OutputStream(int id,
sp<ANativeWindow> consumer,
- uint32_t width, uint32_t height, size_t maxSize, int format) :
+ uint32_t width, uint32_t height, size_t maxSize, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation) :
Camera3IOStreamBase(id, CAMERA3_STREAM_OUTPUT, width, height, maxSize,
- format),
+ format, dataSpace, rotation),
mConsumer(consumer),
mTransform(0),
mTraceFirstBuffer(true) {
@@ -69,10 +71,12 @@
Camera3OutputStream::Camera3OutputStream(int id, camera3_stream_type_t type,
uint32_t width, uint32_t height,
- int format) :
+ int format,
+ android_dataspace dataSpace,
+ camera3_stream_rotation_t rotation) :
Camera3IOStreamBase(id, type, width, height,
/*maxSize*/0,
- format),
+ format, dataSpace, rotation),
mTransform(0) {
// Subclasses expected to initialize mConsumer themselves
@@ -153,33 +157,9 @@
ALOG_ASSERT(output, "Expected output to be true");
status_t res;
- sp<Fence> releaseFence;
- /**
- * Fence management - calculate Release Fence
- */
- if (buffer.status == CAMERA3_BUFFER_STATUS_ERROR) {
- if (buffer.release_fence != -1) {
- ALOGE("%s: Stream %d: HAL should not set release_fence(%d) when "
- "there is an error", __FUNCTION__, mId, buffer.release_fence);
- close(buffer.release_fence);
- }
-
- /**
- * Reassign release fence as the acquire fence in case of error
- */
- releaseFence = new Fence(buffer.acquire_fence);
- } else {
- res = native_window_set_buffers_timestamp(mConsumer.get(), timestamp);
- if (res != OK) {
- ALOGE("%s: Stream %d: Error setting timestamp: %s (%d)",
- __FUNCTION__, mId, strerror(-res), res);
- return res;
- }
-
- releaseFence = new Fence(buffer.release_fence);
- }
-
+ // Fence management - always honor release fence from HAL
+ sp<Fence> releaseFence = new Fence(buffer.release_fence);
int anwReleaseFence = releaseFence->dup();
/**
@@ -213,6 +193,13 @@
mTraceFirstBuffer = false;
}
+ res = native_window_set_buffers_timestamp(mConsumer.get(), timestamp);
+ if (res != OK) {
+ ALOGE("%s: Stream %d: Error setting timestamp: %s (%d)",
+ __FUNCTION__, mId, strerror(-res), res);
+ return res;
+ }
+
res = currentConsumer->queueBuffer(currentConsumer.get(),
container_of(buffer.buffer, ANativeWindowBuffer, handle),
anwReleaseFence);
@@ -323,6 +310,14 @@
return res;
}
+ res = native_window_set_buffers_data_space(mConsumer.get(),
+ camera3_stream::data_space);
+ if (res != OK) {
+ ALOGE("%s: Unable to configure stream dataspace %#x for stream %d",
+ __FUNCTION__, camera3_stream::data_space, mId);
+ return res;
+ }
+
int maxConsumerBuffers;
res = mConsumer->query(mConsumer.get(),
NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS, &maxConsumerBuffers);
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.h b/services/camera/libcameraservice/device3/Camera3OutputStream.h
index be278c5..12b2ebb 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.h
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.h
@@ -39,14 +39,16 @@
* Set up a stream for formats that have 2 dimensions, such as RAW and YUV.
*/
Camera3OutputStream(int id, sp<ANativeWindow> consumer,
- uint32_t width, uint32_t height, int format);
+ uint32_t width, uint32_t height, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation);
/**
* Set up a stream for formats that have a variable buffer size for the same
* dimensions, such as compressed JPEG.
*/
Camera3OutputStream(int id, sp<ANativeWindow> consumer,
- uint32_t width, uint32_t height, size_t maxSize, int format);
+ uint32_t width, uint32_t height, size_t maxSize, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation);
virtual ~Camera3OutputStream();
@@ -64,7 +66,8 @@
protected:
Camera3OutputStream(int id, camera3_stream_type_t type,
- uint32_t width, uint32_t height, int format);
+ uint32_t width, uint32_t height, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation);
/**
* Note that we release the lock briefly in this function
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.cpp b/services/camera/libcameraservice/device3/Camera3Stream.cpp
index 3c0e908..4acbce3 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Stream.cpp
@@ -46,7 +46,8 @@
Camera3Stream::Camera3Stream(int id,
camera3_stream_type type,
- uint32_t width, uint32_t height, size_t maxSize, int format) :
+ uint32_t width, uint32_t height, size_t maxSize, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation) :
camera3_stream(),
mId(id),
mName(String8::format("Camera3Stream[%d]", id)),
@@ -58,6 +59,8 @@
camera3_stream::width = width;
camera3_stream::height = height;
camera3_stream::format = format;
+ camera3_stream::data_space = dataSpace;
+ camera3_stream::rotation = rotation;
camera3_stream::usage = 0;
camera3_stream::max_buffers = 0;
camera3_stream::priv = NULL;
@@ -84,6 +87,10 @@
return camera3_stream::format;
}
+android_dataspace Camera3Stream::getDataSpace() const {
+ return camera3_stream::data_space;
+}
+
camera3_stream* Camera3Stream::startConfiguration() {
ATRACE_CALL();
Mutex::Autolock l(mLock);
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.h b/services/camera/libcameraservice/device3/Camera3Stream.h
index d0e1337..aba27fe 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.h
+++ b/services/camera/libcameraservice/device3/Camera3Stream.h
@@ -119,9 +119,10 @@
/**
* Get the stream's dimensions and format
*/
- uint32_t getWidth() const;
- uint32_t getHeight() const;
- int getFormat() const;
+ uint32_t getWidth() const;
+ uint32_t getHeight() const;
+ int getFormat() const;
+ android_dataspace getDataSpace() const;
/**
* Start the stream configuration process. Returns a handle to the stream's
@@ -264,7 +265,8 @@
mutable Mutex mLock;
Camera3Stream(int id, camera3_stream_type type,
- uint32_t width, uint32_t height, size_t maxSize, int format);
+ uint32_t width, uint32_t height, size_t maxSize, int format,
+ android_dataspace dataSpace, camera3_stream_rotation_t rotation);
/**
* Interface to be implemented by derived classes
diff --git a/services/camera/libcameraservice/device3/Camera3ZslStream.cpp b/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
index 81330ea..10d7f2e 100644
--- a/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
@@ -114,7 +114,8 @@
int bufferCount) :
Camera3OutputStream(id, CAMERA3_STREAM_BIDIRECTIONAL,
width, height,
- HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED),
+ HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED,
+ HAL_DATASPACE_UNKNOWN, CAMERA3_STREAM_ROTATION_0),
mDepth(bufferCount) {
sp<IGraphicBufferProducer> producer;
diff --git a/services/camera/libcameraservice/utils/AutoConditionLock.cpp b/services/camera/libcameraservice/utils/AutoConditionLock.cpp
new file mode 100644
index 0000000..c8ee965
--- /dev/null
+++ b/services/camera/libcameraservice/utils/AutoConditionLock.cpp
@@ -0,0 +1,90 @@
+/*
+ * Copyright (C) 2015 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 "AutoConditionLock.h"
+
+namespace android {
+
+WaitableMutexWrapper::WaitableMutexWrapper(Mutex* mutex) : mMutex{mutex}, mState{false} {}
+
+WaitableMutexWrapper::~WaitableMutexWrapper() {}
+
+// Locks manager-owned mutex
+AutoConditionLock::AutoConditionLock(const std::shared_ptr<WaitableMutexWrapper>& manager) :
+ mManager{manager}, mAutoLock{manager->mMutex} {}
+
+// Unlocks manager-owned mutex
+AutoConditionLock::~AutoConditionLock() {
+ // Unset the condition and wake everyone up before releasing lock
+ mManager->mState = false;
+ mManager->mCondition.broadcast();
+}
+
+std::unique_ptr<AutoConditionLock> AutoConditionLock::waitAndAcquire(
+ const std::shared_ptr<WaitableMutexWrapper>& manager, nsecs_t waitTime) {
+
+ if (manager == nullptr || manager->mMutex == nullptr) {
+ // Bad input, return null
+ return std::unique_ptr<AutoConditionLock>{nullptr};
+ }
+
+ // Acquire scoped lock
+ std::unique_ptr<AutoConditionLock> scopedLock(new AutoConditionLock(manager));
+
+ // Figure out what time in the future we should hit the timeout
+ nsecs_t failTime = systemTime(SYSTEM_TIME_MONOTONIC) + waitTime;
+
+ // Wait until we timeout, or success
+ while(manager->mState) {
+ status_t ret = manager->mCondition.waitRelative(*(manager->mMutex), waitTime);
+ if (ret != NO_ERROR) {
+ // Timed out or whatever, return null
+ return std::unique_ptr<AutoConditionLock>{nullptr};
+ }
+ waitTime = failTime - systemTime(SYSTEM_TIME_MONOTONIC);
+ }
+
+ // Set the condition and return
+ manager->mState = true;
+ return scopedLock;
+}
+
+std::unique_ptr<AutoConditionLock> AutoConditionLock::waitAndAcquire(
+ const std::shared_ptr<WaitableMutexWrapper>& manager) {
+
+ if (manager == nullptr || manager->mMutex == nullptr) {
+ // Bad input, return null
+ return std::unique_ptr<AutoConditionLock>{nullptr};
+ }
+
+ // Acquire scoped lock
+ std::unique_ptr<AutoConditionLock> scopedLock(new AutoConditionLock(manager));
+
+ // Wait until we timeout, or success
+ while(manager->mState) {
+ status_t ret = manager->mCondition.wait(*(manager->mMutex));
+ if (ret != NO_ERROR) {
+ // Timed out or whatever, return null
+ return std::unique_ptr<AutoConditionLock>{nullptr};
+ }
+ }
+
+ // Set the condition and return
+ manager->mState = true;
+ return scopedLock;
+}
+
+}; // namespace android
diff --git a/services/camera/libcameraservice/utils/AutoConditionLock.h b/services/camera/libcameraservice/utils/AutoConditionLock.h
new file mode 100644
index 0000000..9a3eafc
--- /dev/null
+++ b/services/camera/libcameraservice/utils/AutoConditionLock.h
@@ -0,0 +1,99 @@
+/*
+ * Copyright (C) 2015 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_SERVICE_UTILS_SCOPED_CONDITION_H
+#define ANDROID_SERVICE_UTILS_SCOPED_CONDITION_H
+
+#include <utils/Timers.h>
+#include <utils/Condition.h>
+#include <utils/Errors.h>
+#include <utils/Mutex.h>
+
+#include <memory>
+
+namespace android {
+
+/**
+ * WaitableMutexWrapper can be used with AutoConditionLock to construct scoped locks for the
+ * wrapped Mutex with timeouts for lock acquisition.
+ */
+class WaitableMutexWrapper {
+ friend class AutoConditionLock;
+public:
+ /**
+ * Construct the ConditionManger with the given Mutex.
+ */
+ WaitableMutexWrapper(Mutex* mutex);
+
+ virtual ~WaitableMutexWrapper();
+private:
+ Mutex* mMutex;
+ bool mState;
+ Condition mCondition;
+};
+
+/**
+ * AutoConditionLock is a scoped lock similar to Mutex::Autolock, but allows timeouts to be
+ * specified for lock acquisition.
+ *
+ * AutoConditionLock is used with a WaitableMutexWrapper to lock/unlock the WaitableMutexWrapper's
+ * wrapped Mutex, and wait/set/signal the WaitableMutexWrapper's wrapped condition. To use this,
+ * call AutoConditionLock::waitAndAcquire to get an instance. This will:
+ * - Lock the given WaitableMutexWrapper's mutex.
+ * - Wait for the WaitableMutexWrapper's condition to become false, or timeout.
+ * - Set the WaitableMutexWrapper's condition to true.
+ *
+ * When the AutoConditionLock goes out of scope and is destroyed, this will:
+ * - Set the WaitableMutexWrapper's condition to false.
+ * - Signal threads waiting on this condition to wakeup.
+ * - Release WaitableMutexWrapper's mutex.
+ */
+class AutoConditionLock final {
+public:
+ AutoConditionLock() = delete;
+ AutoConditionLock(const AutoConditionLock& other) = delete;
+ AutoConditionLock & operator=(const AutoConditionLock&) = delete;
+
+ ~AutoConditionLock();
+
+ /**
+ * Make a new AutoConditionLock from a given WaitableMutexWrapper, waiting up to waitTime
+ * nanoseconds to acquire the WaitableMutexWrapper's wrapped lock.
+ *
+ * Return an empty unique_ptr if this fails, or a timeout occurs.
+ */
+ static std::unique_ptr<AutoConditionLock> waitAndAcquire(
+ const std::shared_ptr<WaitableMutexWrapper>& manager, nsecs_t waitTime);
+
+ /**
+ * Make a new AutoConditionLock from a given WaitableMutexWrapper, waiting indefinitely to
+ * acquire the WaitableMutexWrapper's wrapped lock.
+ *
+ * Return an empty unique_ptr if this fails.
+ */
+ static std::unique_ptr<AutoConditionLock> waitAndAcquire(
+ const std::shared_ptr<WaitableMutexWrapper>& manager);
+private:
+ AutoConditionLock(const std::shared_ptr<WaitableMutexWrapper>& manager);
+
+ std::shared_ptr<WaitableMutexWrapper> mManager;
+ Mutex::Autolock mAutoLock;
+};
+
+}; // namespace android
+
+#endif // ANDROID_SERVICE_UTILS_SCOPED_CONDITION_H
diff --git a/services/camera/libcameraservice/utils/ClientManager.h b/services/camera/libcameraservice/utils/ClientManager.h
new file mode 100644
index 0000000..ad5486d
--- /dev/null
+++ b/services/camera/libcameraservice/utils/ClientManager.h
@@ -0,0 +1,539 @@
+/*
+ * Copyright (C) 2015 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_SERVICE_UTILS_EVICTION_POLICY_MANAGER_H
+#define ANDROID_SERVICE_UTILS_EVICTION_POLICY_MANAGER_H
+
+#include <utils/Mutex.h>
+
+#include <algorithm>
+#include <utility>
+#include <vector>
+#include <set>
+#include <map>
+#include <memory>
+
+namespace android {
+namespace resource_policy {
+
+// --------------------------------------------------------------------------------
+
+/**
+ * The ClientDescriptor class is a container for a given key/value pair identifying a shared
+ * resource, and the corresponding cost, priority, owner ID, and conflicting keys list used
+ * in determining eviction behavior.
+ *
+ * Aside from the priority, these values are immutable once the ClientDescriptor has been
+ * constructed.
+ */
+template<class KEY, class VALUE>
+class ClientDescriptor final {
+public:
+ ClientDescriptor(const KEY& key, const VALUE& value, int32_t cost,
+ const std::set<KEY>& conflictingKeys, int32_t priority, int32_t ownerId);
+ ClientDescriptor(KEY&& key, VALUE&& value, int32_t cost, std::set<KEY>&& conflictingKeys,
+ int32_t priority, int32_t ownerId);
+
+ ~ClientDescriptor();
+
+ /**
+ * Return the key for this descriptor.
+ */
+ const KEY& getKey() const;
+
+ /**
+ * Return the value for this descriptor.
+ */
+ const VALUE& getValue() const;
+
+ /**
+ * Return the cost for this descriptor.
+ */
+ int32_t getCost() const;
+
+ /**
+ * Return the priority for this descriptor.
+ */
+ int32_t getPriority() const;
+
+ /**
+ * Return the owner ID for this descriptor.
+ */
+ int32_t getOwnerId() const;
+
+ /**
+ * Return true if the given key is in this descriptor's conflicting keys list.
+ */
+ bool isConflicting(const KEY& key) const;
+
+ /**
+ * Return the set of all conflicting keys for this descriptor.
+ */
+ std::set<KEY> getConflicting() const;
+
+ /**
+ * Set the proirity for this descriptor.
+ */
+ void setPriority(int32_t priority);
+
+ // This class is ordered by key
+ template<class K, class V>
+ friend bool operator < (const ClientDescriptor<K, V>& a, const ClientDescriptor<K, V>& b);
+
+private:
+ KEY mKey;
+ VALUE mValue;
+ int32_t mCost;
+ std::set<KEY> mConflicting;
+ int32_t mPriority;
+ int32_t mOwnerId;
+}; // class ClientDescriptor
+
+template<class K, class V>
+bool operator < (const ClientDescriptor<K, V>& a, const ClientDescriptor<K, V>& b) {
+ return a.mKey < b.mKey;
+}
+
+template<class KEY, class VALUE>
+ClientDescriptor<KEY, VALUE>::ClientDescriptor(const KEY& key, const VALUE& value, int32_t cost,
+ const std::set<KEY>& conflictingKeys, int32_t priority, int32_t ownerId) : mKey{key},
+ mValue{value}, mCost{cost}, mConflicting{conflictingKeys}, mPriority{priority},
+ mOwnerId{ownerId} {}
+
+template<class KEY, class VALUE>
+ClientDescriptor<KEY, VALUE>::ClientDescriptor(KEY&& key, VALUE&& value, int32_t cost,
+ std::set<KEY>&& conflictingKeys, int32_t priority, int32_t ownerId) :
+ mKey{std::forward<KEY>(key)}, mValue{std::forward<VALUE>(value)}, mCost{cost},
+ mConflicting{std::forward<std::set<KEY>>(conflictingKeys)}, mPriority{priority},
+ mOwnerId{ownerId} {}
+
+template<class KEY, class VALUE>
+ClientDescriptor<KEY, VALUE>::~ClientDescriptor() {}
+
+template<class KEY, class VALUE>
+const KEY& ClientDescriptor<KEY, VALUE>::getKey() const {
+ return mKey;
+}
+
+template<class KEY, class VALUE>
+const VALUE& ClientDescriptor<KEY, VALUE>::getValue() const {
+ return mValue;
+}
+
+template<class KEY, class VALUE>
+int32_t ClientDescriptor<KEY, VALUE>::getCost() const {
+ return mCost;
+}
+
+template<class KEY, class VALUE>
+int32_t ClientDescriptor<KEY, VALUE>::getPriority() const {
+ return mPriority;
+}
+
+template<class KEY, class VALUE>
+int32_t ClientDescriptor<KEY, VALUE>::getOwnerId() const {
+ return mOwnerId;
+}
+
+template<class KEY, class VALUE>
+bool ClientDescriptor<KEY, VALUE>::isConflicting(const KEY& key) const {
+ if (key == mKey) return true;
+ for (const auto& x : mConflicting) {
+ if (key == x) return true;
+ }
+ return false;
+}
+
+template<class KEY, class VALUE>
+std::set<KEY> ClientDescriptor<KEY, VALUE>::getConflicting() const {
+ return mConflicting;
+}
+
+template<class KEY, class VALUE>
+void ClientDescriptor<KEY, VALUE>::setPriority(int32_t priority) {
+ mPriority = priority;
+}
+
+// --------------------------------------------------------------------------------
+
+/**
+ * The ClientManager class wraps an LRU-ordered list of active clients and implements eviction
+ * behavior for handling shared resource access.
+ *
+ * When adding a new descriptor, eviction behavior is as follows:
+ * - Keys are unique, adding a descriptor with the same key as an existing descriptor will
+ * result in the lower-priority of the two being removed. Priority ties result in the
+ * LRU descriptor being evicted (this means the incoming descriptor be added in this case).
+ * - Any descriptors with keys that are in the incoming descriptor's 'conflicting keys' list
+ * will be removed if they have an equal or lower priority than the incoming descriptor;
+ * if any have a higher priority, the incoming descriptor is removed instead.
+ * - If the sum of all descriptors' costs, including the incoming descriptor's, is more than
+ * the max cost allowed for this ClientManager, descriptors with non-zero cost, equal or lower
+ * priority, and a different owner will be evicted in LRU order until either the cost is less
+ * than the max cost, or all descriptors meeting this criteria have been evicted and the
+ * incoming descriptor has the highest priority. Otherwise, the incoming descriptor is
+ * removed instead.
+ */
+template<class KEY, class VALUE>
+class ClientManager {
+public:
+ // The default maximum "cost" allowed before evicting
+ static constexpr int32_t DEFAULT_MAX_COST = 100;
+
+ ClientManager();
+ ClientManager(int32_t totalCost);
+
+ /**
+ * Add a given ClientDescriptor to the managed list. ClientDescriptors for clients that
+ * are evicted by this action are returned in a vector.
+ *
+ * This may return the ClientDescriptor passed in if it would be evicted.
+ */
+ std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>> addAndEvict(
+ const std::shared_ptr<ClientDescriptor<KEY, VALUE>>& client);
+
+ /**
+ * Given a map containing owner (pid) -> priority mappings, update the priority of each
+ * ClientDescriptor with an owner in this mapping.
+ */
+ void updatePriorities(const std::map<int32_t,int32_t>& ownerPriorityList);
+
+ /**
+ * Remove all ClientDescriptors.
+ */
+ void removeAll();
+
+ /**
+ * Remove and return the ClientDescriptor with a given key.
+ */
+ std::shared_ptr<ClientDescriptor<KEY, VALUE>> remove(const KEY& key);
+
+ /**
+ * Remove the given ClientDescriptor.
+ */
+ void remove(const std::shared_ptr<ClientDescriptor<KEY, VALUE>>& value);
+
+ /**
+ * Return a vector of the ClientDescriptors that would be evicted by adding the given
+ * ClientDescriptor.
+ *
+ * This may return the ClientDescriptor passed in if it would be evicted.
+ */
+ std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>> wouldEvict(
+ const std::shared_ptr<ClientDescriptor<KEY, VALUE>>& client) const;
+
+ /**
+ * Return a vector of active ClientDescriptors that prevent this client from being added.
+ */
+ std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>> getIncompatibleClients(
+ const std::shared_ptr<ClientDescriptor<KEY, VALUE>>& client) const;
+
+ /**
+ * Return a vector containing all currently active ClientDescriptors.
+ */
+ std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>> getAll() const;
+
+ /**
+ * Return a vector containing all keys of currently active ClientDescriptors.
+ */
+ std::vector<KEY> getAllKeys() const;
+
+ /**
+ * Return a vector of the owner tags of all currently active ClientDescriptors (duplicates
+ * will be removed).
+ */
+ std::vector<int32_t> getAllOwners() const;
+
+ /**
+ * Return the ClientDescriptor corresponding to the given key, or an empty shared pointer
+ * if none exists.
+ */
+ std::shared_ptr<ClientDescriptor<KEY, VALUE>> get(const KEY& key) const;
+
+protected:
+ ~ClientManager();
+
+private:
+
+ /**
+ * Return a vector of the ClientDescriptors that would be evicted by adding the given
+ * ClientDescriptor. If returnIncompatibleClients is set to true, instead, return the
+ * vector of ClientDescriptors that are higher priority than the incoming client and
+ * either conflict with this client, or contribute to the resource cost if that would
+ * prevent the incoming client from being added.
+ *
+ * This may return the ClientDescriptor passed in.
+ */
+ std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>> wouldEvictLocked(
+ const std::shared_ptr<ClientDescriptor<KEY, VALUE>>& client,
+ bool returnIncompatibleClients = false) const;
+
+ int64_t getCurrentCostLocked() const;
+
+ mutable Mutex mLock;
+ int32_t mMaxCost;
+ // LRU ordered, most recent at end
+ std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>> mClients;
+}; // class ClientManager
+
+template<class KEY, class VALUE>
+ClientManager<KEY, VALUE>::ClientManager() :
+ ClientManager(DEFAULT_MAX_COST) {}
+
+template<class KEY, class VALUE>
+ClientManager<KEY, VALUE>::ClientManager(int32_t totalCost) : mMaxCost(totalCost) {}
+
+template<class KEY, class VALUE>
+ClientManager<KEY, VALUE>::~ClientManager() {}
+
+template<class KEY, class VALUE>
+std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>> ClientManager<KEY, VALUE>::wouldEvict(
+ const std::shared_ptr<ClientDescriptor<KEY, VALUE>>& client) const {
+ Mutex::Autolock lock(mLock);
+ return wouldEvictLocked(client);
+}
+
+template<class KEY, class VALUE>
+std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>>
+ClientManager<KEY, VALUE>::getIncompatibleClients(
+ const std::shared_ptr<ClientDescriptor<KEY, VALUE>>& client) const {
+ Mutex::Autolock lock(mLock);
+ return wouldEvictLocked(client, /*returnIncompatibleClients*/true);
+}
+
+template<class KEY, class VALUE>
+std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>>
+ClientManager<KEY, VALUE>::wouldEvictLocked(
+ const std::shared_ptr<ClientDescriptor<KEY, VALUE>>& client,
+ bool returnIncompatibleClients) const {
+
+ std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>> evictList;
+
+ // Disallow null clients, return input
+ if (client == nullptr) {
+ evictList.push_back(client);
+ return evictList;
+ }
+
+ const KEY& key = client->getKey();
+ int32_t cost = client->getCost();
+ int32_t priority = client->getPriority();
+ int32_t owner = client->getOwnerId();
+
+ int64_t totalCost = getCurrentCostLocked() + cost;
+
+ // Determine the MRU of the owners tied for having the highest priority
+ int32_t highestPriorityOwner = owner;
+ int32_t highestPriority = priority;
+ for (const auto& i : mClients) {
+ int32_t curPriority = i->getPriority();
+ if (curPriority >= highestPriority) {
+ highestPriority = curPriority;
+ highestPriorityOwner = i->getOwnerId();
+ }
+ }
+
+ if (highestPriority == priority) {
+ // Switch back owner if the incoming client has the highest priority, as it is MRU
+ highestPriorityOwner = owner;
+ }
+
+ // Build eviction list of clients to remove
+ for (const auto& i : mClients) {
+ const KEY& curKey = i->getKey();
+ int32_t curCost = i->getCost();
+ int32_t curPriority = i->getPriority();
+ int32_t curOwner = i->getOwnerId();
+
+ bool conflicting = (curKey == key || i->isConflicting(key) ||
+ client->isConflicting(curKey));
+
+ if (!returnIncompatibleClients) {
+ // Find evicted clients
+
+ if (conflicting && curPriority > priority) {
+ // Pre-existing conflicting client with higher priority exists
+ evictList.clear();
+ evictList.push_back(client);
+ return evictList;
+ } else if (conflicting || ((totalCost > mMaxCost && curCost > 0) &&
+ (curPriority <= priority) &&
+ !(highestPriorityOwner == owner && owner == curOwner))) {
+ // Add a pre-existing client to the eviction list if:
+ // - We are adding a client with higher priority that conflicts with this one.
+ // - The total cost including the incoming client's is more than the allowable
+ // maximum, and the client has a non-zero cost, lower priority, and a different
+ // owner than the incoming client when the incoming client has the
+ // highest priority.
+ evictList.push_back(i);
+ totalCost -= curCost;
+ }
+ } else {
+ // Find clients preventing the incoming client from being added
+
+ if (curPriority > priority && (conflicting || (totalCost > mMaxCost && curCost > 0))) {
+ // Pre-existing conflicting client with higher priority exists
+ evictList.push_back(i);
+ }
+ }
+ }
+
+ // Immediately return the incompatible clients if we are calculating these instead
+ if (returnIncompatibleClients) {
+ return evictList;
+ }
+
+ // If the total cost is too high, return the input unless the input has the highest priority
+ if (totalCost > mMaxCost && highestPriorityOwner != owner) {
+ evictList.clear();
+ evictList.push_back(client);
+ return evictList;
+ }
+
+ return evictList;
+
+}
+
+template<class KEY, class VALUE>
+std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>> ClientManager<KEY, VALUE>::addAndEvict(
+ const std::shared_ptr<ClientDescriptor<KEY, VALUE>>& client) {
+ Mutex::Autolock lock(mLock);
+ auto evicted = wouldEvictLocked(client);
+ auto it = evicted.begin();
+ if (it != evicted.end() && *it == client) {
+ return evicted;
+ }
+
+ auto iter = evicted.cbegin();
+
+ // Remove evicted clients from list
+ mClients.erase(std::remove_if(mClients.begin(), mClients.end(),
+ [&iter] (std::shared_ptr<ClientDescriptor<KEY, VALUE>>& curClientPtr) {
+ if (curClientPtr->getKey() == (*iter)->getKey()) {
+ iter++;
+ return true;
+ }
+ return false;
+ }), mClients.end());
+
+ mClients.push_back(client);
+
+ return evicted;
+}
+
+template<class KEY, class VALUE>
+std::vector<std::shared_ptr<ClientDescriptor<KEY, VALUE>>>
+ClientManager<KEY, VALUE>::getAll() const {
+ Mutex::Autolock lock(mLock);
+ return mClients;
+}
+
+template<class KEY, class VALUE>
+std::vector<KEY> ClientManager<KEY, VALUE>::getAllKeys() const {
+ Mutex::Autolock lock(mLock);
+ std::vector<KEY> keys(mClients.size());
+ for (const auto& i : mClients) {
+ keys.push_back(i->getKey());
+ }
+ return keys;
+}
+
+template<class KEY, class VALUE>
+std::vector<int32_t> ClientManager<KEY, VALUE>::getAllOwners() const {
+ Mutex::Autolock lock(mLock);
+ std::set<int32_t> owners;
+ for (const auto& i : mClients) {
+ owners.emplace(i->getOwnerId());
+ }
+ return std::vector<int32_t>(owners.begin(), owners.end());
+}
+
+template<class KEY, class VALUE>
+void ClientManager<KEY, VALUE>::updatePriorities(
+ const std::map<int32_t,int32_t>& ownerPriorityList) {
+ Mutex::Autolock lock(mLock);
+ for (auto& i : mClients) {
+ auto j = ownerPriorityList.find(i->getOwnerId());
+ if (j != ownerPriorityList.end()) {
+ i->setPriority(j->second);
+ }
+ }
+}
+
+template<class KEY, class VALUE>
+std::shared_ptr<ClientDescriptor<KEY, VALUE>> ClientManager<KEY, VALUE>::get(
+ const KEY& key) const {
+ Mutex::Autolock lock(mLock);
+ for (const auto& i : mClients) {
+ if (i->getKey() == key) return i;
+ }
+ return std::shared_ptr<ClientDescriptor<KEY, VALUE>>(nullptr);
+}
+
+template<class KEY, class VALUE>
+void ClientManager<KEY, VALUE>::removeAll() {
+ Mutex::Autolock lock(mLock);
+ mClients.clear();
+}
+
+template<class KEY, class VALUE>
+std::shared_ptr<ClientDescriptor<KEY, VALUE>> ClientManager<KEY, VALUE>::remove(const KEY& key) {
+ Mutex::Autolock lock(mLock);
+
+ std::shared_ptr<ClientDescriptor<KEY, VALUE>> ret;
+
+ // Remove evicted clients from list
+ mClients.erase(std::remove_if(mClients.begin(), mClients.end(),
+ [&key, &ret] (std::shared_ptr<ClientDescriptor<KEY, VALUE>>& curClientPtr) {
+ if (curClientPtr->getKey() == key) {
+ ret = curClientPtr;
+ return true;
+ }
+ return false;
+ }), mClients.end());
+
+ return ret;
+}
+
+template<class KEY, class VALUE>
+void ClientManager<KEY, VALUE>::remove(
+ const std::shared_ptr<ClientDescriptor<KEY, VALUE>>& value) {
+ Mutex::Autolock lock(mLock);
+ // Remove evicted clients from list
+ mClients.erase(std::remove_if(mClients.begin(), mClients.end(),
+ [&value] (std::shared_ptr<ClientDescriptor<KEY, VALUE>>& curClientPtr) {
+ if (curClientPtr == value) {
+ return true;
+ }
+ return false;
+ }), mClients.end());
+}
+
+template<class KEY, class VALUE>
+int64_t ClientManager<KEY, VALUE>::getCurrentCostLocked() const {
+ int64_t totalCost = 0;
+ for (const auto& x : mClients) {
+ totalCost += x->getCost();
+ }
+ return totalCost;
+}
+
+// --------------------------------------------------------------------------------
+
+}; // namespace resource_policy
+}; // namespace android
+
+#endif // ANDROID_SERVICE_UTILS_EVICTION_POLICY_MANAGER_H
diff --git a/services/camera/libcameraservice/utils/RingBuffer.h b/services/camera/libcameraservice/utils/RingBuffer.h
new file mode 100644
index 0000000..df7c00e
--- /dev/null
+++ b/services/camera/libcameraservice/utils/RingBuffer.h
@@ -0,0 +1,361 @@
+/*
+ * Copyright (C) 2015 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_SERVICE_UTILS_RING_BUFFER_H
+#define ANDROID_SERVICE_UTILS_RING_BUFFER_H
+
+#include <utils/Log.h>
+#include <cutils/compiler.h>
+
+#include <iterator>
+#include <utility>
+#include <vector>
+
+namespace android {
+
+/**
+ * A RingBuffer class that maintains an array of objects that can grow up to a certain size.
+ * Elements added to the RingBuffer are inserted in the logical front of the buffer, and
+ * invalidate all current iterators for that RingBuffer object.
+ */
+template <class T>
+class RingBuffer final {
+public:
+
+ /**
+ * Construct a RingBuffer that can grow up to the given length.
+ */
+ RingBuffer(size_t length);
+
+ /**
+ * Forward iterator to this class. Implements an std:forward_iterator.
+ */
+ class iterator : public std::iterator<std::forward_iterator_tag, T> {
+ public:
+ iterator(T* ptr, size_t size, size_t pos, size_t ctr);
+
+ iterator& operator++();
+
+ iterator operator++(int);
+
+ bool operator==(const iterator& rhs);
+
+ bool operator!=(const iterator& rhs);
+
+ T& operator*();
+
+ T* operator->();
+
+ private:
+ T* mPtr;
+ size_t mSize;
+ size_t mPos;
+ size_t mCtr;
+ };
+
+ /**
+ * Constant forward iterator to this class. Implements an std:forward_iterator.
+ */
+ class const_iterator : public std::iterator<std::forward_iterator_tag, T> {
+ public:
+ const_iterator(const T* ptr, size_t size, size_t pos, size_t ctr);
+
+ const_iterator& operator++();
+
+ const_iterator operator++(int);
+
+ bool operator==(const const_iterator& rhs);
+
+ bool operator!=(const const_iterator& rhs);
+
+ const T& operator*();
+
+ const T* operator->();
+
+ private:
+ const T* mPtr;
+ size_t mSize;
+ size_t mPos;
+ size_t mCtr;
+ };
+
+ /**
+ * Adds item to the front of this RingBuffer. If the RingBuffer is at its maximum length,
+ * this will result in the last element being replaced (this is done using the element's
+ * assignment operator).
+ *
+ * All current iterators are invalidated.
+ */
+ void add(const T& item);
+
+ /**
+ * Moves item to the front of this RingBuffer. Following a call to this, item should no
+ * longer be used. If the RingBuffer is at its maximum length, this will result in the
+ * last element being replaced (this is done using the element's assignment operator).
+ *
+ * All current iterators are invalidated.
+ */
+ void add(T&& item);
+
+ /**
+ * Construct item in-place in the front of this RingBuffer using the given arguments. If
+ * the RingBuffer is at its maximum length, this will result in the last element being
+ * replaced (this is done using the element's assignment operator).
+ *
+ * All current iterators are invalidated.
+ */
+ template <class... Args>
+ void emplace(Args&&... args);
+
+ /**
+ * Get an iterator to the front of this RingBuffer.
+ */
+ iterator begin();
+
+ /**
+ * Get an iterator to the end of this RingBuffer.
+ */
+ iterator end();
+
+ /**
+ * Get a const_iterator to the front of this RingBuffer.
+ */
+ const_iterator begin() const;
+
+ /**
+ * Get a const_iterator to the end of this RingBuffer.
+ */
+ const_iterator end() const;
+
+ /**
+ * Return a reference to the element at a given index. If the index is out of range for
+ * this ringbuffer, [0, size), the behavior for this is undefined.
+ */
+ T& operator[](size_t index);
+
+ /**
+ * Return a const reference to the element at a given index. If the index is out of range
+ * for this ringbuffer, [0, size), the behavior for this is undefined.
+ */
+ const T& operator[](size_t index) const;
+
+ /**
+ * Return the current size of this RingBuffer.
+ */
+ size_t size() const;
+
+ /**
+ * Remove all elements from this RingBuffer and set the size to 0.
+ */
+ void clear();
+
+private:
+ size_t mFrontIdx;
+ size_t mMaxBufferSize;
+ std::vector<T> mBuffer;
+}; // class RingBuffer
+
+
+template <class T>
+RingBuffer<T>::RingBuffer(size_t length) : mFrontIdx{0}, mMaxBufferSize{length} {}
+
+template <class T>
+RingBuffer<T>::iterator::iterator(T* ptr, size_t size, size_t pos, size_t ctr) :
+ mPtr{ptr}, mSize{size}, mPos{pos}, mCtr{ctr} {}
+
+template <class T>
+typename RingBuffer<T>::iterator& RingBuffer<T>::iterator::operator++() {
+ ++mCtr;
+
+ if (CC_UNLIKELY(mCtr == mSize)) {
+ mPos = mSize;
+ return *this;
+ }
+
+ mPos = ((CC_UNLIKELY(mPos == 0)) ? mSize - 1 : mPos - 1);
+ return *this;
+}
+
+template <class T>
+typename RingBuffer<T>::iterator RingBuffer<T>::iterator::operator++(int) {
+ iterator tmp{mPtr, mSize, mPos, mCtr};
+ ++(*this);
+ return tmp;
+}
+
+template <class T>
+bool RingBuffer<T>::iterator::operator==(const iterator& rhs) {
+ return (mPtr + mPos) == (rhs.mPtr + rhs.mPos);
+}
+
+template <class T>
+bool RingBuffer<T>::iterator::operator!=(const iterator& rhs) {
+ return (mPtr + mPos) != (rhs.mPtr + rhs.mPos);
+}
+
+template <class T>
+T& RingBuffer<T>::iterator::operator*() {
+ return *(mPtr + mPos);
+}
+
+template <class T>
+T* RingBuffer<T>::iterator::operator->() {
+ return mPtr + mPos;
+}
+
+template <class T>
+RingBuffer<T>::const_iterator::const_iterator(const T* ptr, size_t size, size_t pos, size_t ctr) :
+ mPtr{ptr}, mSize{size}, mPos{pos}, mCtr{ctr} {}
+
+template <class T>
+typename RingBuffer<T>::const_iterator& RingBuffer<T>::const_iterator::operator++() {
+ ++mCtr;
+
+ if (CC_UNLIKELY(mCtr == mSize)) {
+ mPos = mSize;
+ return *this;
+ }
+
+ mPos = ((CC_UNLIKELY(mPos == 0)) ? mSize - 1 : mPos - 1);
+ return *this;
+}
+
+template <class T>
+typename RingBuffer<T>::const_iterator RingBuffer<T>::const_iterator::operator++(int) {
+ const_iterator tmp{mPtr, mSize, mPos, mCtr};
+ ++(*this);
+ return tmp;
+}
+
+template <class T>
+bool RingBuffer<T>::const_iterator::operator==(const const_iterator& rhs) {
+ return (mPtr + mPos) == (rhs.mPtr + rhs.mPos);
+}
+
+template <class T>
+bool RingBuffer<T>::const_iterator::operator!=(const const_iterator& rhs) {
+ return (mPtr + mPos) != (rhs.mPtr + rhs.mPos);
+}
+
+template <class T>
+const T& RingBuffer<T>::const_iterator::operator*() {
+ return *(mPtr + mPos);
+}
+
+template <class T>
+const T* RingBuffer<T>::const_iterator::operator->() {
+ return mPtr + mPos;
+}
+
+template <class T>
+void RingBuffer<T>::add(const T& item) {
+ if (mBuffer.size() < mMaxBufferSize) {
+ mBuffer.push_back(item);
+ mFrontIdx = ((mFrontIdx + 1) % mMaxBufferSize);
+ return;
+ }
+
+ mBuffer[mFrontIdx] = item;
+ mFrontIdx = ((mFrontIdx + 1) % mMaxBufferSize);
+}
+
+template <class T>
+void RingBuffer<T>::add(T&& item) {
+ if (mBuffer.size() != mMaxBufferSize) {
+ mBuffer.push_back(std::forward<T>(item));
+ mFrontIdx = ((mFrontIdx + 1) % mMaxBufferSize);
+ return;
+ }
+
+ // Only works for types with move assignment operator
+ mBuffer[mFrontIdx] = std::forward<T>(item);
+ mFrontIdx = ((mFrontIdx + 1) % mMaxBufferSize);
+}
+
+template <class T>
+template <class... Args>
+void RingBuffer<T>::emplace(Args&&... args) {
+ if (mBuffer.size() != mMaxBufferSize) {
+ mBuffer.emplace_back(std::forward<Args>(args)...);
+ mFrontIdx = ((mFrontIdx + 1) % mMaxBufferSize);
+ return;
+ }
+
+ // Only works for types with move assignment operator
+ mBuffer[mFrontIdx] = T(std::forward<Args>(args)...);
+ mFrontIdx = ((mFrontIdx + 1) % mMaxBufferSize);
+}
+
+template <class T>
+typename RingBuffer<T>::iterator RingBuffer<T>::begin() {
+ size_t tmp = (mBuffer.size() == 0) ? 0 : mBuffer.size() - 1;
+ return iterator(mBuffer.data(), mBuffer.size(), (mFrontIdx == 0) ? tmp : mFrontIdx - 1, 0);
+}
+
+template <class T>
+typename RingBuffer<T>::iterator RingBuffer<T>::end() {
+ size_t s = mBuffer.size();
+ return iterator(mBuffer.data(), s, s, s);
+}
+
+template <class T>
+typename RingBuffer<T>::const_iterator RingBuffer<T>::begin() const {
+ size_t tmp = (mBuffer.size() == 0) ? 0 : mBuffer.size() - 1;
+ return const_iterator(mBuffer.data(), mBuffer.size(),
+ (mFrontIdx == 0) ? tmp : mFrontIdx - 1, 0);
+}
+
+template <class T>
+typename RingBuffer<T>::const_iterator RingBuffer<T>::end() const {
+ size_t s = mBuffer.size();
+ return const_iterator(mBuffer.data(), s, s, s);
+}
+
+template <class T>
+T& RingBuffer<T>::operator[](size_t index) {
+ LOG_ALWAYS_FATAL_IF(index >= mBuffer.size(), "Index %zu out of bounds, size is %zu.",
+ index, mBuffer.size());
+ size_t pos = (index >= mFrontIdx) ?
+ mBuffer.size() - 1 - (index - mFrontIdx) : mFrontIdx - 1 - index;
+ return mBuffer[pos];
+}
+
+template <class T>
+const T& RingBuffer<T>::operator[](size_t index) const {
+ LOG_ALWAYS_FATAL_IF(index >= mBuffer.size(), "Index %zu out of bounds, size is %zu.",
+ index, mBuffer.size());
+ size_t pos = (index >= mFrontIdx) ?
+ mBuffer.size() - 1 - (index - mFrontIdx) : mFrontIdx - 1 - index;
+ return mBuffer[pos];
+}
+
+template <class T>
+size_t RingBuffer<T>::size() const {
+ return mBuffer.size();
+}
+
+template <class T>
+void RingBuffer<T>::clear() {
+ mBuffer.clear();
+ mFrontIdx = 0;
+}
+
+}; // namespace android
+
+#endif // ANDROID_SERVICE_UTILS_RING_BUFFER_H
+
+
diff --git a/services/medialog/Android.mk b/services/medialog/Android.mk
index 95f2fef..03438bf 100644
--- a/services/medialog/Android.mk
+++ b/services/medialog/Android.mk
@@ -10,4 +10,6 @@
LOCAL_32_BIT_ONLY := true
+LOCAL_C_INCLUDES := $(call include-path-for, audio-utils)
+
include $(BUILD_SHARED_LIBRARY)
diff --git a/services/mediaresourcemanager/Android.mk b/services/mediaresourcemanager/Android.mk
new file mode 100644
index 0000000..84218cf
--- /dev/null
+++ b/services/mediaresourcemanager/Android.mk
@@ -0,0 +1,18 @@
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := ResourceManagerService.cpp
+
+LOCAL_SHARED_LIBRARIES := libmedia libstagefright libbinder libutils liblog
+
+LOCAL_MODULE:= libresourcemanagerservice
+
+LOCAL_32_BIT_ONLY := true
+
+LOCAL_C_INCLUDES += \
+ $(TOPDIR)frameworks/av/include
+
+include $(BUILD_SHARED_LIBRARY)
+
+include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/services/mediaresourcemanager/ResourceManagerService.cpp b/services/mediaresourcemanager/ResourceManagerService.cpp
new file mode 100644
index 0000000..7296d47
--- /dev/null
+++ b/services/mediaresourcemanager/ResourceManagerService.cpp
@@ -0,0 +1,345 @@
+/*
+**
+** Copyright 2015, 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 "ResourceManagerService"
+#include <utils/Log.h>
+
+#include <binder/IServiceManager.h>
+#include <dirent.h>
+#include <media/stagefright/ProcessInfo.h>
+#include <string.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/time.h>
+#include <unistd.h>
+
+#include "ResourceManagerService.h"
+
+namespace android {
+
+template <typename T>
+static String8 getString(const Vector<T> &items) {
+ String8 itemsStr;
+ for (size_t i = 0; i < items.size(); ++i) {
+ itemsStr.appendFormat("%s ", items[i].toString().string());
+ }
+ return itemsStr;
+}
+
+static bool hasResourceType(String8 type, Vector<MediaResource> resources) {
+ for (size_t i = 0; i < resources.size(); ++i) {
+ if (resources[i].mType == type) {
+ return true;
+ }
+ }
+ return false;
+}
+
+static bool hasResourceType(String8 type, ResourceInfos infos) {
+ for (size_t i = 0; i < infos.size(); ++i) {
+ if (hasResourceType(type, infos[i].resources)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+static ResourceInfos& getResourceInfosForEdit(
+ int pid,
+ PidResourceInfosMap& map) {
+ ssize_t index = map.indexOfKey(pid);
+ if (index < 0) {
+ // new pid
+ ResourceInfos infosForPid;
+ map.add(pid, infosForPid);
+ }
+
+ return map.editValueFor(pid);
+}
+
+static ResourceInfo& getResourceInfoForEdit(
+ int64_t clientId,
+ const sp<IResourceManagerClient> client,
+ ResourceInfos& infos) {
+ for (size_t i = 0; i < infos.size(); ++i) {
+ if (infos[i].clientId == clientId) {
+ return infos.editItemAt(i);
+ }
+ }
+ ResourceInfo info;
+ info.clientId = clientId;
+ info.client = client;
+ infos.push_back(info);
+ return infos.editItemAt(infos.size() - 1);
+}
+
+ResourceManagerService::ResourceManagerService()
+ : mProcessInfo(new ProcessInfo()),
+ mSupportsMultipleSecureCodecs(true),
+ mSupportsSecureWithNonSecureCodec(true) {}
+
+ResourceManagerService::ResourceManagerService(sp<ProcessInfoInterface> processInfo)
+ : mProcessInfo(processInfo),
+ mSupportsMultipleSecureCodecs(true),
+ mSupportsSecureWithNonSecureCodec(true) {}
+
+ResourceManagerService::~ResourceManagerService() {}
+
+void ResourceManagerService::config(const Vector<MediaResourcePolicy> &policies) {
+ ALOGV("config(%s)", getString(policies).string());
+
+ Mutex::Autolock lock(mLock);
+ for (size_t i = 0; i < policies.size(); ++i) {
+ String8 type = policies[i].mType;
+ uint64_t value = policies[i].mValue;
+ if (type == kPolicySupportsMultipleSecureCodecs) {
+ mSupportsMultipleSecureCodecs = (value != 0);
+ } else if (type == kPolicySupportsSecureWithNonSecureCodec) {
+ mSupportsSecureWithNonSecureCodec = (value != 0);
+ }
+ }
+}
+
+void ResourceManagerService::addResource(
+ int pid,
+ int64_t clientId,
+ const sp<IResourceManagerClient> client,
+ const Vector<MediaResource> &resources) {
+ ALOGV("addResource(pid %d, clientId %lld, resources %s)",
+ pid, (long long) clientId, getString(resources).string());
+
+ Mutex::Autolock lock(mLock);
+ ResourceInfos& infos = getResourceInfosForEdit(pid, mMap);
+ ResourceInfo& info = getResourceInfoForEdit(clientId, client, infos);
+ info.resources.appendVector(resources);
+}
+
+void ResourceManagerService::removeResource(int64_t clientId) {
+ ALOGV("removeResource(%lld)", (long long) clientId);
+
+ Mutex::Autolock lock(mLock);
+ bool found = false;
+ for (size_t i = 0; i < mMap.size(); ++i) {
+ ResourceInfos &infos = mMap.editValueAt(i);
+ for (size_t j = 0; j < infos.size();) {
+ if (infos[j].clientId == clientId) {
+ j = infos.removeAt(j);
+ found = true;
+ } else {
+ ++j;
+ }
+ }
+ if (found) {
+ break;
+ }
+ }
+ if (!found) {
+ ALOGV("didn't find client");
+ }
+}
+
+bool ResourceManagerService::reclaimResource(
+ int callingPid, const Vector<MediaResource> &resources) {
+ ALOGV("reclaimResource(callingPid %d, resources %s)",
+ callingPid, getString(resources).string());
+
+ Vector<sp<IResourceManagerClient>> clients;
+ {
+ Mutex::Autolock lock(mLock);
+ // first pass to handle secure/non-secure codec conflict
+ for (size_t i = 0; i < resources.size(); ++i) {
+ String8 type = resources[i].mType;
+ if (type == kResourceSecureCodec) {
+ if (!mSupportsMultipleSecureCodecs) {
+ if (!getAllClients_l(callingPid, String8(kResourceSecureCodec), &clients)) {
+ return false;
+ }
+ }
+ if (!mSupportsSecureWithNonSecureCodec) {
+ if (!getAllClients_l(callingPid, String8(kResourceNonSecureCodec), &clients)) {
+ return false;
+ }
+ }
+ } else if (type == kResourceNonSecureCodec) {
+ if (!mSupportsSecureWithNonSecureCodec) {
+ if (!getAllClients_l(callingPid, String8(kResourceSecureCodec), &clients)) {
+ return false;
+ }
+ }
+ }
+ }
+
+ if (clients.size() == 0) {
+ // if no secure/non-secure codec conflict, run second pass to handle other resources.
+ for (size_t i = 0; i < resources.size(); ++i) {
+ String8 type = resources[i].mType;
+ if (type == kResourceGraphicMemory) {
+ sp<IResourceManagerClient> client;
+ if (!getLowestPriorityBiggestClient_l(callingPid, type, &client)) {
+ return false;
+ }
+ clients.push_back(client);
+ }
+ }
+ }
+ }
+
+ if (clients.size() == 0) {
+ return false;
+ }
+
+ for (size_t i = 0; i < clients.size(); ++i) {
+ ALOGV("reclaimResource from client %p", clients[i].get());
+ if (!clients[i]->reclaimResource()) {
+ return false;
+ }
+ }
+ return true;
+}
+
+bool ResourceManagerService::getAllClients_l(
+ int callingPid, const String8 &type, Vector<sp<IResourceManagerClient>> *clients) {
+ Vector<sp<IResourceManagerClient>> temp;
+ for (size_t i = 0; i < mMap.size(); ++i) {
+ ResourceInfos &infos = mMap.editValueAt(i);
+ for (size_t j = 0; j < infos.size(); ++j) {
+ if (hasResourceType(type, infos[j].resources)) {
+ if (!isCallingPriorityHigher_l(callingPid, mMap.keyAt(i))) {
+ // some higher/equal priority process owns the resource,
+ // this request can't be fulfilled.
+ ALOGE("getAllClients_l: can't reclaim resource %s from pid %d",
+ type.string(), mMap.keyAt(i));
+ return false;
+ }
+ temp.push_back(infos[j].client);
+ }
+ }
+ }
+ if (temp.size() == 0) {
+ ALOGV("getAllClients_l: didn't find any resource %s", type.string());
+ return true;
+ }
+ clients->appendVector(temp);
+ return true;
+}
+
+bool ResourceManagerService::getLowestPriorityBiggestClient_l(
+ int callingPid, const String8 &type, sp<IResourceManagerClient> *client) {
+ int lowestPriorityPid;
+ int lowestPriority;
+ int callingPriority;
+ if (!mProcessInfo->getPriority(callingPid, &callingPriority)) {
+ ALOGE("getLowestPriorityBiggestClient_l: can't get process priority for pid %d",
+ callingPid);
+ return false;
+ }
+ if (!getLowestPriorityPid_l(type, &lowestPriorityPid, &lowestPriority)) {
+ return false;
+ }
+ if (lowestPriority <= callingPriority) {
+ ALOGE("getLowestPriorityBiggestClient_l: lowest priority %d vs caller priority %d",
+ lowestPriority, callingPriority);
+ return false;
+ }
+
+ if (!getBiggestClient_l(lowestPriorityPid, type, client)) {
+ return false;
+ }
+ return true;
+}
+
+bool ResourceManagerService::getLowestPriorityPid_l(
+ const String8 &type, int *lowestPriorityPid, int *lowestPriority) {
+ int pid = -1;
+ int priority = -1;
+ for (size_t i = 0; i < mMap.size(); ++i) {
+ if (mMap.valueAt(i).size() == 0) {
+ // no client on this process.
+ continue;
+ }
+ if (!hasResourceType(type, mMap.valueAt(i))) {
+ // doesn't have the requested resource type
+ continue;
+ }
+ int tempPid = mMap.keyAt(i);
+ int tempPriority;
+ if (!mProcessInfo->getPriority(tempPid, &tempPriority)) {
+ ALOGV("getLowestPriorityPid_l: can't get priority of pid %d, skipped", tempPid);
+ // TODO: remove this pid from mMap?
+ continue;
+ }
+ if (pid == -1 || tempPriority > priority) {
+ // initial the value
+ pid = tempPid;
+ priority = tempPriority;
+ }
+ }
+ if (pid != -1) {
+ *lowestPriorityPid = pid;
+ *lowestPriority = priority;
+ }
+ return (pid != -1);
+}
+
+bool ResourceManagerService::isCallingPriorityHigher_l(int callingPid, int pid) {
+ int callingPidPriority;
+ if (!mProcessInfo->getPriority(callingPid, &callingPidPriority)) {
+ return false;
+ }
+
+ int priority;
+ if (!mProcessInfo->getPriority(pid, &priority)) {
+ return false;
+ }
+
+ return (callingPidPriority < priority);
+}
+
+bool ResourceManagerService::getBiggestClient_l(
+ int pid, const String8 &type, sp<IResourceManagerClient> *client) {
+ ssize_t index = mMap.indexOfKey(pid);
+ if (index < 0) {
+ ALOGE("getBiggestClient_l: can't find resource info for pid %d", pid);
+ return false;
+ }
+
+ sp<IResourceManagerClient> clientTemp;
+ uint64_t largestValue = 0;
+ const ResourceInfos &infos = mMap.valueAt(index);
+ for (size_t i = 0; i < infos.size(); ++i) {
+ Vector<MediaResource> resources = infos[i].resources;
+ for (size_t j = 0; j < resources.size(); ++j) {
+ if (resources[j].mType == type) {
+ if (resources[j].mValue > largestValue) {
+ largestValue = resources[j].mValue;
+ clientTemp = infos[i].client;
+ }
+ }
+ }
+ }
+
+ if (clientTemp == NULL) {
+ ALOGE("getBiggestClient_l: can't find resource type %s for pid %d", type.string(), pid);
+ return false;
+ }
+
+ *client = clientTemp;
+ return true;
+}
+
+} // namespace android
diff --git a/services/mediaresourcemanager/ResourceManagerService.h b/services/mediaresourcemanager/ResourceManagerService.h
new file mode 100644
index 0000000..2ed9bf8
--- /dev/null
+++ b/services/mediaresourcemanager/ResourceManagerService.h
@@ -0,0 +1,106 @@
+/*
+**
+** Copyright 2015, 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_RESOURCEMANAGERSERVICE_H
+#define ANDROID_RESOURCEMANAGERSERVICE_H
+
+#include <arpa/inet.h>
+#include <binder/BinderService.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/String8.h>
+#include <utils/threads.h>
+#include <utils/Vector.h>
+
+#include <media/IResourceManagerService.h>
+
+namespace android {
+
+struct ProcessInfoInterface;
+
+struct ResourceInfo {
+ int64_t clientId;
+ sp<IResourceManagerClient> client;
+ Vector<MediaResource> resources;
+};
+
+typedef Vector<ResourceInfo> ResourceInfos;
+typedef KeyedVector<int, ResourceInfos> PidResourceInfosMap;
+
+class ResourceManagerService
+ : public BinderService<ResourceManagerService>,
+ public BnResourceManagerService
+{
+public:
+ static char const *getServiceName() { return "media.resource_manager"; }
+
+ ResourceManagerService();
+ ResourceManagerService(sp<ProcessInfoInterface> processInfo);
+
+ // IResourceManagerService interface
+ virtual void config(const Vector<MediaResourcePolicy> &policies);
+
+ virtual void addResource(
+ int pid,
+ int64_t clientId,
+ const sp<IResourceManagerClient> client,
+ const Vector<MediaResource> &resources);
+
+ virtual void removeResource(int64_t clientId);
+
+ virtual bool reclaimResource(int callingPid, const Vector<MediaResource> &resources);
+
+protected:
+ virtual ~ResourceManagerService();
+
+private:
+ friend class ResourceManagerServiceTest;
+
+ // Gets the list of all the clients who own the specified resource type.
+ // Returns false if any client belongs to a process with higher priority than the
+ // calling process. The clients will remain unchanged if returns false.
+ bool getAllClients_l(int callingPid, const String8 &type,
+ Vector<sp<IResourceManagerClient>> *clients);
+
+ // Gets the client who owns specified resource type from lowest possible priority process.
+ // Returns false if the calling process priority is not higher than the lowest process
+ // priority. The client will remain unchanged if returns false.
+ bool getLowestPriorityBiggestClient_l(int callingPid, const String8 &type,
+ sp<IResourceManagerClient> *client);
+
+ // Gets lowest priority process that has the specified resource type.
+ // Returns false if failed. The output parameters will remain unchanged if failed.
+ bool getLowestPriorityPid_l(const String8 &type, int *pid, int *priority);
+
+ // Gets the client who owns biggest piece of specified resource type from pid.
+ // Returns false if failed. The client will remain unchanged if failed.
+ bool getBiggestClient_l(int pid, const String8 &type, sp<IResourceManagerClient> *client);
+
+ bool isCallingPriorityHigher_l(int callingPid, int pid);
+
+ mutable Mutex mLock;
+ sp<ProcessInfoInterface> mProcessInfo;
+ PidResourceInfosMap mMap;
+ bool mSupportsMultipleSecureCodecs;
+ bool mSupportsSecureWithNonSecureCodec;
+};
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif // ANDROID_RESOURCEMANAGERSERVICE_H
diff --git a/services/mediaresourcemanager/test/Android.mk b/services/mediaresourcemanager/test/Android.mk
new file mode 100644
index 0000000..228b62a
--- /dev/null
+++ b/services/mediaresourcemanager/test/Android.mk
@@ -0,0 +1,25 @@
+# Build the unit tests.
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := ResourceManagerService_test
+
+LOCAL_MODULE_TAGS := tests
+
+LOCAL_SRC_FILES := \
+ ResourceManagerService_test.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+ libbinder \
+ liblog \
+ libmedia \
+ libresourcemanagerservice \
+ libutils \
+
+LOCAL_C_INCLUDES := \
+ frameworks/av/include \
+ frameworks/av/services/mediaresourcemanager \
+
+LOCAL_32_BIT_ONLY := true
+
+include $(BUILD_NATIVE_TEST)
diff --git a/services/mediaresourcemanager/test/ResourceManagerService_test.cpp b/services/mediaresourcemanager/test/ResourceManagerService_test.cpp
new file mode 100644
index 0000000..b73e1bc
--- /dev/null
+++ b/services/mediaresourcemanager/test/ResourceManagerService_test.cpp
@@ -0,0 +1,464 @@
+/*
+ * Copyright (C) 2015 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 "ResourceManagerService_test"
+#include <utils/Log.h>
+
+#include <gtest/gtest.h>
+
+#include "ResourceManagerService.h"
+#include <media/IResourceManagerService.h>
+#include <media/MediaResource.h>
+#include <media/MediaResourcePolicy.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/ProcessInfoInterface.h>
+
+namespace android {
+
+struct TestProcessInfo : public ProcessInfoInterface {
+ TestProcessInfo() {}
+ virtual ~TestProcessInfo() {}
+
+ virtual bool getPriority(int pid, int *priority) {
+ // For testing, use pid as priority.
+ // Lower the value higher the priority.
+ *priority = pid;
+ return true;
+ }
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(TestProcessInfo);
+};
+
+struct TestClient : public BnResourceManagerClient {
+ TestClient(sp<ResourceManagerService> service)
+ : mReclaimed(false), mService(service) {}
+
+ virtual bool reclaimResource() {
+ sp<IResourceManagerClient> client(this);
+ mService->removeResource((int64_t) client.get());
+ mReclaimed = true;
+ return true;
+ }
+
+ bool reclaimed() const {
+ return mReclaimed;
+ }
+
+ void reset() {
+ mReclaimed = false;
+ }
+
+protected:
+ virtual ~TestClient() {}
+
+private:
+ bool mReclaimed;
+ sp<ResourceManagerService> mService;
+ DISALLOW_EVIL_CONSTRUCTORS(TestClient);
+};
+
+static const int kTestPid1 = 30;
+static const int kTestPid2 = 20;
+
+class ResourceManagerServiceTest : public ::testing::Test {
+public:
+ ResourceManagerServiceTest()
+ : mService(new ResourceManagerService(new TestProcessInfo)),
+ mTestClient1(new TestClient(mService)),
+ mTestClient2(new TestClient(mService)),
+ mTestClient3(new TestClient(mService)) {
+ }
+
+protected:
+ static bool isEqualResources(const Vector<MediaResource> &resources1,
+ const Vector<MediaResource> &resources2) {
+ if (resources1.size() != resources2.size()) {
+ return false;
+ }
+ for (size_t i = 0; i < resources1.size(); ++i) {
+ if (resources1[i] != resources2[i]) {
+ return false;
+ }
+ }
+ return true;
+ }
+
+ static void expectEqResourceInfo(const ResourceInfo &info, sp<IResourceManagerClient> client,
+ const Vector<MediaResource> &resources) {
+ EXPECT_EQ(client, info.client);
+ EXPECT_TRUE(isEqualResources(resources, info.resources));
+ }
+
+ void verifyClients(bool c1, bool c2, bool c3) {
+ TestClient *client1 = static_cast<TestClient*>(mTestClient1.get());
+ TestClient *client2 = static_cast<TestClient*>(mTestClient2.get());
+ TestClient *client3 = static_cast<TestClient*>(mTestClient3.get());
+
+ EXPECT_EQ(c1, client1->reclaimed());
+ EXPECT_EQ(c2, client2->reclaimed());
+ EXPECT_EQ(c3, client3->reclaimed());
+
+ client1->reset();
+ client2->reset();
+ client3->reset();
+ }
+
+ void addResource() {
+ // kTestPid1 mTestClient1
+ Vector<MediaResource> resources1;
+ resources1.push_back(MediaResource(String8(kResourceSecureCodec), 1));
+ mService->addResource(kTestPid1, (int64_t) mTestClient1.get(), mTestClient1, resources1);
+ resources1.push_back(MediaResource(String8(kResourceGraphicMemory), 200));
+ Vector<MediaResource> resources11;
+ resources11.push_back(MediaResource(String8(kResourceGraphicMemory), 200));
+ mService->addResource(kTestPid1, (int64_t) mTestClient1.get(), mTestClient1, resources11);
+
+ // kTestPid2 mTestClient2
+ Vector<MediaResource> resources2;
+ resources2.push_back(MediaResource(String8(kResourceNonSecureCodec), 1));
+ resources2.push_back(MediaResource(String8(kResourceGraphicMemory), 300));
+ mService->addResource(kTestPid2, (int64_t) mTestClient2.get(), mTestClient2, resources2);
+
+ // kTestPid2 mTestClient3
+ Vector<MediaResource> resources3;
+ mService->addResource(kTestPid2, (int64_t) mTestClient3.get(), mTestClient3, resources3);
+ resources3.push_back(MediaResource(String8(kResourceSecureCodec), 1));
+ resources3.push_back(MediaResource(String8(kResourceGraphicMemory), 100));
+ mService->addResource(kTestPid2, (int64_t) mTestClient3.get(), mTestClient3, resources3);
+
+ const PidResourceInfosMap &map = mService->mMap;
+ EXPECT_EQ(2u, map.size());
+ ssize_t index1 = map.indexOfKey(kTestPid1);
+ ASSERT_GE(index1, 0);
+ const ResourceInfos &infos1 = map[index1];
+ EXPECT_EQ(1u, infos1.size());
+ expectEqResourceInfo(infos1[0], mTestClient1, resources1);
+
+ ssize_t index2 = map.indexOfKey(kTestPid2);
+ ASSERT_GE(index2, 0);
+ const ResourceInfos &infos2 = map[index2];
+ EXPECT_EQ(2u, infos2.size());
+ expectEqResourceInfo(infos2[0], mTestClient2, resources2);
+ expectEqResourceInfo(infos2[1], mTestClient3, resources3);
+ }
+
+ void testConfig() {
+ EXPECT_TRUE(mService->mSupportsMultipleSecureCodecs);
+ EXPECT_TRUE(mService->mSupportsSecureWithNonSecureCodec);
+
+ Vector<MediaResourcePolicy> policies1;
+ policies1.push_back(MediaResourcePolicy(String8(kPolicySupportsMultipleSecureCodecs), 1));
+ policies1.push_back(
+ MediaResourcePolicy(String8(kPolicySupportsSecureWithNonSecureCodec), 0));
+ mService->config(policies1);
+ EXPECT_TRUE(mService->mSupportsMultipleSecureCodecs);
+ EXPECT_FALSE(mService->mSupportsSecureWithNonSecureCodec);
+
+ Vector<MediaResourcePolicy> policies2;
+ policies2.push_back(MediaResourcePolicy(String8(kPolicySupportsMultipleSecureCodecs), 0));
+ policies2.push_back(
+ MediaResourcePolicy(String8(kPolicySupportsSecureWithNonSecureCodec), 1));
+ mService->config(policies2);
+ EXPECT_FALSE(mService->mSupportsMultipleSecureCodecs);
+ EXPECT_TRUE(mService->mSupportsSecureWithNonSecureCodec);
+ }
+
+ void testRemoveResource() {
+ addResource();
+
+ mService->removeResource((int64_t) mTestClient2.get());
+
+ const PidResourceInfosMap &map = mService->mMap;
+ EXPECT_EQ(2u, map.size());
+ const ResourceInfos &infos1 = map.valueFor(kTestPid1);
+ const ResourceInfos &infos2 = map.valueFor(kTestPid2);
+ EXPECT_EQ(1u, infos1.size());
+ EXPECT_EQ(1u, infos2.size());
+ // mTestClient2 has been removed.
+ EXPECT_EQ(mTestClient3, infos2[0].client);
+ }
+
+ void testGetAllClients() {
+ addResource();
+
+ String8 type = String8(kResourceSecureCodec);
+ String8 unknowType = String8("unknowType");
+ Vector<sp<IResourceManagerClient> > clients;
+ int lowPriorityPid = 100;
+ EXPECT_FALSE(mService->getAllClients_l(lowPriorityPid, type, &clients));
+ int midPriorityPid = 25;
+ EXPECT_FALSE(mService->getAllClients_l(lowPriorityPid, type, &clients));
+ int highPriorityPid = 10;
+ EXPECT_TRUE(mService->getAllClients_l(10, unknowType, &clients));
+ EXPECT_TRUE(mService->getAllClients_l(10, type, &clients));
+
+ EXPECT_EQ(2u, clients.size());
+ EXPECT_EQ(mTestClient3, clients[0]);
+ EXPECT_EQ(mTestClient1, clients[1]);
+ }
+
+ void testReclaimResourceSecure() {
+ Vector<MediaResource> resources;
+ resources.push_back(MediaResource(String8(kResourceSecureCodec), 1));
+ resources.push_back(MediaResource(String8(kResourceGraphicMemory), 150));
+
+ // ### secure codec can't coexist and secure codec can coexist with non-secure codec ###
+ {
+ addResource();
+ mService->mSupportsMultipleSecureCodecs = false;
+ mService->mSupportsSecureWithNonSecureCodec = true;
+
+ // priority too low
+ EXPECT_FALSE(mService->reclaimResource(40, resources));
+ EXPECT_FALSE(mService->reclaimResource(25, resources));
+
+ // reclaim all secure codecs
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(true, false, true);
+
+ // call again should reclaim one largest graphic memory from lowest process
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(false, true, false);
+
+ // nothing left
+ EXPECT_FALSE(mService->reclaimResource(10, resources));
+ }
+
+ // ### secure codecs can't coexist and secure codec can't coexist with non-secure codec ###
+ {
+ addResource();
+ mService->mSupportsMultipleSecureCodecs = false;
+ mService->mSupportsSecureWithNonSecureCodec = false;
+
+ // priority too low
+ EXPECT_FALSE(mService->reclaimResource(40, resources));
+ EXPECT_FALSE(mService->reclaimResource(25, resources));
+
+ // reclaim all secure and non-secure codecs
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(true, true, true);
+
+ // nothing left
+ EXPECT_FALSE(mService->reclaimResource(10, resources));
+ }
+
+
+ // ### secure codecs can coexist but secure codec can't coexist with non-secure codec ###
+ {
+ addResource();
+ mService->mSupportsMultipleSecureCodecs = true;
+ mService->mSupportsSecureWithNonSecureCodec = false;
+
+ // priority too low
+ EXPECT_FALSE(mService->reclaimResource(40, resources));
+ EXPECT_FALSE(mService->reclaimResource(25, resources));
+
+ // reclaim all non-secure codecs
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(false, true, false);
+
+ // call again should reclaim one largest graphic memory from lowest process
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(true, false, false);
+
+ // call again should reclaim another largest graphic memory from lowest process
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(false, false, true);
+
+ // nothing left
+ EXPECT_FALSE(mService->reclaimResource(10, resources));
+ }
+
+ // ### secure codecs can coexist and secure codec can coexist with non-secure codec ###
+ {
+ addResource();
+ mService->mSupportsMultipleSecureCodecs = true;
+ mService->mSupportsSecureWithNonSecureCodec = true;
+
+ // priority too low
+ EXPECT_FALSE(mService->reclaimResource(40, resources));
+
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ // one largest graphic memory from lowest process got reclaimed
+ verifyClients(true, false, false);
+
+ // call again should reclaim another graphic memory from lowest process
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(false, true, false);
+
+ // call again should reclaim another graphic memory from lowest process
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(false, false, true);
+
+ // nothing left
+ EXPECT_FALSE(mService->reclaimResource(10, resources));
+ }
+ }
+
+ void testReclaimResourceNonSecure() {
+ Vector<MediaResource> resources;
+ resources.push_back(MediaResource(String8(kResourceNonSecureCodec), 1));
+ resources.push_back(MediaResource(String8(kResourceGraphicMemory), 150));
+
+ // ### secure codec can't coexist with non-secure codec ###
+ {
+ addResource();
+ mService->mSupportsSecureWithNonSecureCodec = false;
+
+ // priority too low
+ EXPECT_FALSE(mService->reclaimResource(40, resources));
+ EXPECT_FALSE(mService->reclaimResource(25, resources));
+
+ // reclaim all secure codecs
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(true, false, true);
+
+ // call again should reclaim one graphic memory from lowest process
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(false, true, false);
+
+ // nothing left
+ EXPECT_FALSE(mService->reclaimResource(10, resources));
+ }
+
+
+ // ### secure codec can coexist with non-secure codec ###
+ {
+ addResource();
+ mService->mSupportsSecureWithNonSecureCodec = true;
+
+ // priority too low
+ EXPECT_FALSE(mService->reclaimResource(40, resources));
+
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ // one largest graphic memory from lowest process got reclaimed
+ verifyClients(true, false, false);
+
+ // call again should reclaim another graphic memory from lowest process
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(false, true, false);
+
+ // call again should reclaim another graphic memory from lowest process
+ EXPECT_TRUE(mService->reclaimResource(10, resources));
+ verifyClients(false, false, true);
+
+ // nothing left
+ EXPECT_FALSE(mService->reclaimResource(10, resources));
+ }
+ }
+
+ void testGetLowestPriorityBiggestClient() {
+ String8 type = String8(kResourceGraphicMemory);
+ sp<IResourceManagerClient> client;
+ EXPECT_FALSE(mService->getLowestPriorityBiggestClient_l(10, type, &client));
+
+ addResource();
+
+ EXPECT_FALSE(mService->getLowestPriorityBiggestClient_l(100, type, &client));
+ EXPECT_TRUE(mService->getLowestPriorityBiggestClient_l(10, type, &client));
+
+ // kTestPid1 is the lowest priority process with kResourceGraphicMemory.
+ // mTestClient1 has the largest kResourceGraphicMemory within kTestPid1.
+ EXPECT_EQ(mTestClient1, client);
+ }
+
+ void testGetLowestPriorityPid() {
+ int pid;
+ int priority;
+ TestProcessInfo processInfo;
+
+ String8 type = String8(kResourceGraphicMemory);
+ EXPECT_FALSE(mService->getLowestPriorityPid_l(type, &pid, &priority));
+
+ addResource();
+
+ EXPECT_TRUE(mService->getLowestPriorityPid_l(type, &pid, &priority));
+ EXPECT_EQ(kTestPid1, pid);
+ int priority1;
+ processInfo.getPriority(kTestPid1, &priority1);
+ EXPECT_EQ(priority1, priority);
+
+ type = String8(kResourceNonSecureCodec);
+ EXPECT_TRUE(mService->getLowestPriorityPid_l(type, &pid, &priority));
+ EXPECT_EQ(kTestPid2, pid);
+ int priority2;
+ processInfo.getPriority(kTestPid2, &priority2);
+ EXPECT_EQ(priority2, priority);
+ }
+
+ void testGetBiggestClient() {
+ String8 type = String8(kResourceGraphicMemory);
+ sp<IResourceManagerClient> client;
+ EXPECT_FALSE(mService->getBiggestClient_l(kTestPid2, type, &client));
+
+ addResource();
+
+ EXPECT_TRUE(mService->getBiggestClient_l(kTestPid2, type, &client));
+ EXPECT_EQ(mTestClient2, client);
+ }
+
+ void testIsCallingPriorityHigher() {
+ EXPECT_FALSE(mService->isCallingPriorityHigher_l(101, 100));
+ EXPECT_FALSE(mService->isCallingPriorityHigher_l(100, 100));
+ EXPECT_TRUE(mService->isCallingPriorityHigher_l(99, 100));
+ }
+
+ sp<ResourceManagerService> mService;
+ sp<IResourceManagerClient> mTestClient1;
+ sp<IResourceManagerClient> mTestClient2;
+ sp<IResourceManagerClient> mTestClient3;
+};
+
+TEST_F(ResourceManagerServiceTest, config) {
+ testConfig();
+}
+
+TEST_F(ResourceManagerServiceTest, addResource) {
+ addResource();
+}
+
+TEST_F(ResourceManagerServiceTest, removeResource) {
+ testRemoveResource();
+}
+
+TEST_F(ResourceManagerServiceTest, reclaimResource) {
+ testReclaimResourceSecure();
+ testReclaimResourceNonSecure();
+}
+
+TEST_F(ResourceManagerServiceTest, getAllClients_l) {
+ testGetAllClients();
+}
+
+TEST_F(ResourceManagerServiceTest, getLowestPriorityBiggestClient_l) {
+ testGetLowestPriorityBiggestClient();
+}
+
+TEST_F(ResourceManagerServiceTest, getLowestPriorityPid_l) {
+ testGetLowestPriorityPid();
+}
+
+TEST_F(ResourceManagerServiceTest, getBiggestClient_l) {
+ testGetBiggestClient();
+}
+
+TEST_F(ResourceManagerServiceTest, isCallingPriorityHigher_l) {
+ testIsCallingPriorityHigher();
+}
+
+} // namespace android
diff --git a/services/radio/Android.mk b/services/radio/Android.mk
new file mode 100644
index 0000000..9ee5666
--- /dev/null
+++ b/services/radio/Android.mk
@@ -0,0 +1,36 @@
+# Copyright 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.
+
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+
+LOCAL_SRC_FILES:= \
+ RadioService.cpp
+
+LOCAL_SHARED_LIBRARIES:= \
+ libui \
+ liblog \
+ libutils \
+ libbinder \
+ libcutils \
+ libmedia \
+ libhardware \
+ libradio \
+ libradio_metadata
+
+LOCAL_MODULE:= libradioservice
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/services/radio/RadioRegions.h b/services/radio/RadioRegions.h
new file mode 100644
index 0000000..3335b8a
--- /dev/null
+++ b/services/radio/RadioRegions.h
@@ -0,0 +1,225 @@
+/*
+ * Copyright (C) 2015 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_RADIO_REGIONS_H
+#define ANDROID_HARDWARE_RADIO_REGIONS_H
+
+namespace android {
+
+#define RADIO_BAND_LOWER_FM_ITU1 87500
+#define RADIO_BAND_UPPER_FM_ITU1 108000
+#define RADIO_BAND_SPACING_FM_ITU1 100
+
+#define RADIO_BAND_LOWER_FM_ITU2 87900
+#define RADIO_BAND_UPPER_FM_ITU2 107900
+#define RADIO_BAND_SPACING_FM_ITU2 200
+
+#define RADIO_BAND_LOWER_FM_JAPAN 76000
+#define RADIO_BAND_UPPER_FM_JAPAN 90000
+#define RADIO_BAND_SPACING_FM_JAPAN 100
+
+#define RADIO_BAND_LOWER_FM_OIRT 65800
+#define RADIO_BAND_UPPER_FM_OIRT 74000
+#define RADIO_BAND_SPACING_FM_OIRT 10
+
+#define RADIO_BAND_LOWER_LW 153
+#define RADIO_BAND_UPPER_LW 279
+#define RADIO_BAND_SPACING_LW 9
+
+#define RADIO_BAND_LOWER_MW_IUT1 531
+#define RADIO_BAND_UPPER_MW_ITU1 1611
+#define RADIO_BAND_SPACING_MW_ITU1 9
+
+#define RADIO_BAND_LOWER_MW_IUT2 540
+#define RADIO_BAND_UPPER_MW_ITU2 1610
+#define RADIO_BAND_SPACING_MW_ITU2 10
+
+#define RADIO_BAND_LOWER_SW 2300
+#define RADIO_BAND_UPPER_SW 26100
+#define RADIO_BAND_SPACING_SW 5
+
+
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+#endif
+
+const radio_band_config_t sKnownRegionConfigs[] = {
+ { // FM ITU 1
+ RADIO_REGION_ITU_1,
+ {
+ RADIO_BAND_FM,
+ false,
+ RADIO_BAND_LOWER_FM_ITU1,
+ RADIO_BAND_UPPER_FM_ITU1,
+ 1,
+ {RADIO_BAND_SPACING_FM_ITU1},
+ {
+ RADIO_DEEMPHASIS_50,
+ true,
+ RADIO_RDS_WORLD,
+ true,
+ true,
+ }
+ }
+ },
+ { // FM Americas
+ RADIO_REGION_ITU_2,
+ {
+ RADIO_BAND_FM,
+ false,
+ RADIO_BAND_LOWER_FM_ITU2,
+ RADIO_BAND_UPPER_FM_ITU2,
+ 1,
+ {RADIO_BAND_SPACING_FM_ITU2},
+ {
+ RADIO_DEEMPHASIS_75,
+ true,
+ RADIO_RDS_US,
+ true,
+ true,
+ }
+ }
+ },
+ { // FM Japan
+ RADIO_REGION_JAPAN,
+ {
+ RADIO_BAND_FM,
+ false,
+ RADIO_BAND_LOWER_FM_JAPAN,
+ RADIO_BAND_UPPER_FM_JAPAN,
+ 1,
+ {RADIO_BAND_SPACING_FM_JAPAN},
+ {
+ RADIO_DEEMPHASIS_50,
+ true,
+ RADIO_RDS_WORLD,
+ true,
+ true,
+ }
+ }
+ },
+ { // FM Korea
+ RADIO_REGION_KOREA,
+ {
+ RADIO_BAND_FM,
+ false,
+ RADIO_BAND_LOWER_FM_ITU1,
+ RADIO_BAND_UPPER_FM_ITU1,
+ 1,
+ {RADIO_BAND_SPACING_FM_ITU1},
+ {
+ RADIO_DEEMPHASIS_75,
+ true,
+ RADIO_RDS_WORLD,
+ true,
+ true,
+ }
+ }
+ },
+ { // FM OIRT
+ RADIO_REGION_OIRT,
+ {
+ RADIO_BAND_FM,
+ false,
+ RADIO_BAND_LOWER_FM_OIRT,
+ RADIO_BAND_UPPER_FM_OIRT,
+ 1,
+ {RADIO_BAND_SPACING_FM_OIRT},
+ {
+ RADIO_DEEMPHASIS_50,
+ true,
+ RADIO_RDS_WORLD,
+ true,
+ true,
+ }
+ }
+ },
+ { // FM US HD radio
+ RADIO_REGION_ITU_2,
+ {
+ RADIO_BAND_FM_HD,
+ false,
+ RADIO_BAND_LOWER_FM_ITU2,
+ RADIO_BAND_UPPER_FM_ITU2,
+ 1,
+ {RADIO_BAND_SPACING_FM_ITU2},
+ {
+ RADIO_DEEMPHASIS_75,
+ true,
+ RADIO_RDS_US,
+ true,
+ true,
+ }
+ }
+ },
+ { // AM LW
+ RADIO_REGION_ITU_1,
+ {
+ RADIO_BAND_AM,
+ false,
+ RADIO_BAND_LOWER_LW,
+ RADIO_BAND_UPPER_LW,
+ 1,
+ {RADIO_BAND_SPACING_LW},
+ {
+ }
+ }
+ },
+ { // AM SW
+ RADIO_REGION_ITU_1,
+ {
+ RADIO_BAND_AM,
+ false,
+ RADIO_BAND_LOWER_SW,
+ RADIO_BAND_UPPER_SW,
+ 1,
+ {RADIO_BAND_SPACING_SW},
+ {
+ }
+ }
+ },
+ { // AM MW ITU1
+ RADIO_REGION_ITU_1,
+ {
+ RADIO_BAND_AM,
+ false,
+ RADIO_BAND_LOWER_MW_IUT1,
+ RADIO_BAND_UPPER_MW_ITU1,
+ 1,
+ {RADIO_BAND_SPACING_MW_ITU1},
+ {
+ }
+ }
+ },
+ { // AM MW ITU2
+ RADIO_REGION_ITU_2,
+ {
+ RADIO_BAND_AM,
+ false,
+ RADIO_BAND_LOWER_MW_IUT2,
+ RADIO_BAND_UPPER_MW_ITU2,
+ 1,
+ {RADIO_BAND_SPACING_MW_ITU2},
+ {
+ }
+ }
+ }
+};
+
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_RADIO_REGIONS_H
diff --git a/services/radio/RadioService.cpp b/services/radio/RadioService.cpp
new file mode 100644
index 0000000..a6c2bdf
--- /dev/null
+++ b/services/radio/RadioService.cpp
@@ -0,0 +1,901 @@
+/*
+ * Copyright (C) 2015 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 "RadioService"
+//#define LOG_NDEBUG 0
+
+#include <stdio.h>
+#include <string.h>
+#include <sys/types.h>
+#include <pthread.h>
+
+#include <system/audio.h>
+#include <system/audio_policy.h>
+#include <system/radio.h>
+#include <system/radio_metadata.h>
+#include <cutils/atomic.h>
+#include <cutils/properties.h>
+#include <hardware/hardware.h>
+#include <utils/Errors.h>
+#include <utils/Log.h>
+#include <binder/IServiceManager.h>
+#include <binder/MemoryBase.h>
+#include <binder/MemoryHeapBase.h>
+#include <hardware/radio.h>
+#include <media/AudioSystem.h>
+#include "RadioService.h"
+#include "RadioRegions.h"
+
+namespace android {
+
+static const char kRadioTunerAudioDeviceName[] = "Radio tuner source";
+
+RadioService::RadioService()
+ : BnRadioService(), mNextUniqueId(1)
+{
+ ALOGI("%s", __FUNCTION__);
+}
+
+void RadioService::onFirstRef()
+{
+ const hw_module_t *mod;
+ int rc;
+ struct radio_hw_device *dev;
+
+ ALOGI("%s", __FUNCTION__);
+
+ rc = hw_get_module_by_class(RADIO_HARDWARE_MODULE_ID, RADIO_HARDWARE_MODULE_ID_FM, &mod);
+ if (rc != 0) {
+ ALOGE("couldn't load radio module %s.%s (%s)",
+ RADIO_HARDWARE_MODULE_ID, "primary", strerror(-rc));
+ return;
+ }
+ rc = radio_hw_device_open(mod, &dev);
+ if (rc != 0) {
+ ALOGE("couldn't open radio hw device in %s.%s (%s)",
+ RADIO_HARDWARE_MODULE_ID, "primary", strerror(-rc));
+ return;
+ }
+ if (dev->common.version != RADIO_DEVICE_API_VERSION_CURRENT) {
+ ALOGE("wrong radio hw device version %04x", dev->common.version);
+ return;
+ }
+
+ struct radio_hal_properties halProperties;
+ rc = dev->get_properties(dev, &halProperties);
+ if (rc != 0) {
+ ALOGE("could not read implementation properties");
+ return;
+ }
+
+ radio_properties_t properties;
+ properties.handle =
+ (radio_handle_t)android_atomic_inc(&mNextUniqueId);
+
+ ALOGI("loaded default module %s, handle %d", properties.product, properties.handle);
+
+ convertProperties(&properties, &halProperties);
+ sp<Module> module = new Module(dev, properties);
+ mModules.add(properties.handle, module);
+}
+
+RadioService::~RadioService()
+{
+ for (size_t i = 0; i < mModules.size(); i++) {
+ radio_hw_device_close(mModules.valueAt(i)->hwDevice());
+ }
+}
+
+status_t RadioService::listModules(struct radio_properties *properties,
+ uint32_t *numModules)
+{
+ ALOGV("listModules");
+
+ AutoMutex lock(mServiceLock);
+ if (numModules == NULL || (*numModules != 0 && properties == NULL)) {
+ return BAD_VALUE;
+ }
+ size_t maxModules = *numModules;
+ *numModules = mModules.size();
+ for (size_t i = 0; i < mModules.size() && i < maxModules; i++) {
+ properties[i] = mModules.valueAt(i)->properties();
+ }
+ return NO_ERROR;
+}
+
+status_t RadioService::attach(radio_handle_t handle,
+ const sp<IRadioClient>& client,
+ const struct radio_band_config *config,
+ bool withAudio,
+ sp<IRadio>& radio)
+{
+ ALOGV("%s %d config %p withAudio %d", __FUNCTION__, handle, config, withAudio);
+
+ AutoMutex lock(mServiceLock);
+ radio.clear();
+ if (client == 0) {
+ return BAD_VALUE;
+ }
+ ssize_t index = mModules.indexOfKey(handle);
+ if (index < 0) {
+ return BAD_VALUE;
+ }
+ sp<Module> module = mModules.valueAt(index);
+
+ if (config == NULL) {
+ config = module->getDefaultConfig();
+ if (config == NULL) {
+ return INVALID_OPERATION;
+ }
+ }
+ ALOGV("%s region %d type %d", __FUNCTION__, config->region, config->band.type);
+
+ radio = module->addClient(client, config, withAudio);
+
+ if (radio == 0) {
+ NO_INIT;
+ }
+ return NO_ERROR;
+}
+
+
+static const int kDumpLockRetries = 50;
+static const int kDumpLockSleep = 60000;
+
+static bool tryLock(Mutex& mutex)
+{
+ bool locked = false;
+ for (int i = 0; i < kDumpLockRetries; ++i) {
+ if (mutex.tryLock() == NO_ERROR) {
+ locked = true;
+ break;
+ }
+ usleep(kDumpLockSleep);
+ }
+ return locked;
+}
+
+status_t RadioService::dump(int fd, const Vector<String16>& args __unused) {
+ String8 result;
+ if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+ result.appendFormat("Permission Denial: can't dump RadioService");
+ write(fd, result.string(), result.size());
+ } else {
+ bool locked = tryLock(mServiceLock);
+ // failed to lock - RadioService is probably deadlocked
+ if (!locked) {
+ result.append("RadioService may be deadlocked\n");
+ write(fd, result.string(), result.size());
+ }
+
+ if (locked) mServiceLock.unlock();
+ }
+ return NO_ERROR;
+}
+
+status_t RadioService::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) {
+ return BnRadioService::onTransact(code, data, reply, flags);
+}
+
+
+// static
+void RadioService::callback(radio_hal_event_t *halEvent, void *cookie)
+{
+ CallbackThread *callbackThread = (CallbackThread *)cookie;
+ if (callbackThread == NULL) {
+ return;
+ }
+ callbackThread->sendEvent(halEvent);
+}
+
+/* static */
+void RadioService::convertProperties(radio_properties_t *properties,
+ const radio_hal_properties_t *halProperties)
+{
+ memset(properties, 0, sizeof(struct radio_properties));
+ properties->class_id = halProperties->class_id;
+ strlcpy(properties->implementor, halProperties->implementor,
+ RADIO_STRING_LEN_MAX);
+ strlcpy(properties->product, halProperties->product,
+ RADIO_STRING_LEN_MAX);
+ strlcpy(properties->version, halProperties->version,
+ RADIO_STRING_LEN_MAX);
+ strlcpy(properties->serial, halProperties->serial,
+ RADIO_STRING_LEN_MAX);
+ properties->num_tuners = halProperties->num_tuners;
+ properties->num_audio_sources = halProperties->num_audio_sources;
+ properties->supports_capture = halProperties->supports_capture;
+
+ for (size_t i = 0; i < ARRAY_SIZE(sKnownRegionConfigs); i++) {
+ const radio_hal_band_config_t *band = &sKnownRegionConfigs[i].band;
+ size_t j;
+ for (j = 0; j < halProperties->num_bands; j++) {
+ const radio_hal_band_config_t *halBand = &halProperties->bands[j];
+ size_t k;
+ if (band->type != halBand->type) continue;
+ if (band->lower_limit < halBand->lower_limit) continue;
+ if (band->upper_limit > halBand->upper_limit) continue;
+ for (k = 0; k < halBand->num_spacings; k++) {
+ if (band->spacings[0] == halBand->spacings[k]) break;
+ }
+ if (k == halBand->num_spacings) continue;
+ if (band->type == RADIO_BAND_AM) break;
+ if ((band->fm.deemphasis & halBand->fm.deemphasis) == 0) continue;
+ if (halBand->fm.rds == 0) break;
+ if ((band->fm.rds & halBand->fm.rds) != 0) break;
+ }
+ if (j == halProperties->num_bands) continue;
+
+ ALOGI("convertProperties() Adding band type %d region %d",
+ sKnownRegionConfigs[i].band.type , sKnownRegionConfigs[i].region);
+
+ memcpy(&properties->bands[properties->num_bands++],
+ &sKnownRegionConfigs[i],
+ sizeof(radio_band_config_t));
+ }
+}
+
+#undef LOG_TAG
+#define LOG_TAG "RadioService::CallbackThread"
+
+RadioService::CallbackThread::CallbackThread(const wp<ModuleClient>& moduleClient)
+ : mModuleClient(moduleClient), mMemoryDealer(new MemoryDealer(1024 * 1024, "RadioService"))
+{
+}
+
+RadioService::CallbackThread::~CallbackThread()
+{
+ mEventQueue.clear();
+}
+
+void RadioService::CallbackThread::onFirstRef()
+{
+ run("RadioService cbk", ANDROID_PRIORITY_URGENT_AUDIO);
+}
+
+bool RadioService::CallbackThread::threadLoop()
+{
+ while (!exitPending()) {
+ sp<IMemory> eventMemory;
+ sp<ModuleClient> moduleClient;
+ {
+ Mutex::Autolock _l(mCallbackLock);
+ while (mEventQueue.isEmpty() && !exitPending()) {
+ ALOGV("CallbackThread::threadLoop() sleep");
+ mCallbackCond.wait(mCallbackLock);
+ ALOGV("CallbackThread::threadLoop() wake up");
+ }
+ if (exitPending()) {
+ break;
+ }
+ eventMemory = mEventQueue[0];
+ mEventQueue.removeAt(0);
+ moduleClient = mModuleClient.promote();
+ }
+ if (moduleClient != 0) {
+ moduleClient->onCallbackEvent(eventMemory);
+ eventMemory.clear();
+ }
+ }
+ return false;
+}
+
+void RadioService::CallbackThread::exit()
+{
+ Mutex::Autolock _l(mCallbackLock);
+ requestExit();
+ mCallbackCond.broadcast();
+}
+
+sp<IMemory> RadioService::CallbackThread::prepareEvent(radio_hal_event_t *halEvent)
+{
+ sp<IMemory> eventMemory;
+
+ size_t headerSize =
+ (sizeof(struct radio_event) + sizeof(unsigned int) - 1) /sizeof(unsigned int);
+ size_t metadataSize = 0;
+ switch (halEvent->type) {
+ case RADIO_EVENT_TUNED:
+ case RADIO_EVENT_AF_SWITCH:
+ if (radio_metadata_check(halEvent->info.metadata) == 0) {
+ metadataSize = radio_metadata_get_size(halEvent->info.metadata);
+ }
+ break;
+ case RADIO_EVENT_METADATA:
+ if (radio_metadata_check(halEvent->metadata) != 0) {
+ return eventMemory;
+ }
+ metadataSize = radio_metadata_get_size(halEvent->metadata);
+ break;
+ default:
+ break;
+ }
+ size_t size = headerSize + metadataSize;
+ eventMemory = mMemoryDealer->allocate(size);
+ if (eventMemory == 0 || eventMemory->pointer() == NULL) {
+ eventMemory.clear();
+ return eventMemory;
+ }
+ struct radio_event *event = (struct radio_event *)eventMemory->pointer();
+ event->type = halEvent->type;
+ event->status = halEvent->status;
+
+ switch (event->type) {
+ case RADIO_EVENT_CONFIG:
+ event->config.band = halEvent->config;
+ break;
+ case RADIO_EVENT_TUNED:
+ case RADIO_EVENT_AF_SWITCH:
+ event->info = halEvent->info;
+ if (metadataSize != 0) {
+ memcpy((char *)event + headerSize, halEvent->info.metadata, metadataSize);
+ // replace meta data pointer by offset while in shared memory so that receiving side
+ // can restore the pointer in destination process.
+ event->info.metadata = (radio_metadata_t *)headerSize;
+ }
+ break;
+ case RADIO_EVENT_TA:
+ case RADIO_EVENT_ANTENNA:
+ case RADIO_EVENT_CONTROL:
+ event->on = halEvent->on;
+ break;
+ case RADIO_EVENT_METADATA:
+ memcpy((char *)event + headerSize, halEvent->metadata, metadataSize);
+ // replace meta data pointer by offset while in shared memory so that receiving side
+ // can restore the pointer in destination process.
+ event->metadata = (radio_metadata_t *)headerSize;
+ break;
+ case RADIO_EVENT_HW_FAILURE:
+ default:
+ break;
+ }
+
+ return eventMemory;
+}
+
+void RadioService::CallbackThread::sendEvent(radio_hal_event_t *event)
+ {
+ sp<IMemory> eventMemory = prepareEvent(event);
+ if (eventMemory == 0) {
+ return;
+ }
+
+ AutoMutex lock(mCallbackLock);
+ mEventQueue.add(eventMemory);
+ mCallbackCond.signal();
+ ALOGV("%s DONE", __FUNCTION__);
+}
+
+
+#undef LOG_TAG
+#define LOG_TAG "RadioService::Module"
+
+RadioService::Module::Module(radio_hw_device* hwDevice, radio_properties properties)
+ : mHwDevice(hwDevice), mProperties(properties), mMute(true)
+{
+}
+
+RadioService::Module::~Module() {
+ mModuleClients.clear();
+}
+
+status_t RadioService::Module::dump(int fd __unused, const Vector<String16>& args __unused) {
+ String8 result;
+ return NO_ERROR;
+}
+
+sp<RadioService::ModuleClient> RadioService::Module::addClient(const sp<IRadioClient>& client,
+ const struct radio_band_config *config,
+ bool audio)
+{
+ ALOGV("addClient() %p config %p product %s", this, config, mProperties.product);
+ AutoMutex lock(mLock);
+ sp<ModuleClient> moduleClient;
+ int ret;
+
+ for (size_t i = 0; i < mModuleClients.size(); i++) {
+ if (mModuleClients[i]->client() == client) {
+ // client already connected: reject
+ return moduleClient;
+ }
+ }
+ moduleClient = new ModuleClient(this, client, config, audio);
+
+ struct radio_hal_band_config halConfig;
+ halConfig = config->band;
+
+ // Tuner preemption logic:
+ // There is a limited amount of tuners and a limited amount of radio audio sources per module.
+ // The minimum is one tuner and one audio source.
+ // The numbers of tuners and sources are indicated in the module properties.
+ // NOTE: current framework implementation only supports one radio audio source.
+ // It is possible to open more than one tuner at a time but only one tuner can be connected
+ // to the radio audio source (AUDIO_DEVICE_IN_FM_TUNER).
+ // The base rule is that a newly connected tuner always wins, i.e. always gets a tuner
+ // and can use the audio source if requested.
+ // If another client is preempted, it is notified by a callback with RADIO_EVENT_CONTROL
+ // indicating loss of control.
+ // - If the newly connected client requests the audio source (audio == true):
+ // - if an audio source is available
+ // no problem
+ // - if not:
+ // the oldest client in the list using audio is preempted.
+ // - If the newly connected client does not request the audio source (audio == false):
+ // - if a tuner is available
+ // no problem
+ // - if not:
+ // The oldest client not using audio is preempted first and if none is found the
+ // the oldest client using audio is preempted.
+ // Each time a tuner using the audio source is opened or closed, the audio policy manager is
+ // notified of the connection or disconnection of AUDIO_DEVICE_IN_FM_TUNER.
+
+ sp<ModuleClient> oldestTuner;
+ sp<ModuleClient> oldestAudio;
+ size_t allocatedTuners = 0;
+ size_t allocatedAudio = 0;
+ for (size_t i = 0; i < mModuleClients.size(); i++) {
+ if (mModuleClients[i]->getTuner() != NULL) {
+ if (mModuleClients[i]->audio()) {
+ if (oldestAudio == 0) {
+ oldestAudio = mModuleClients[i];
+ }
+ allocatedAudio++;
+ } else {
+ if (oldestTuner == 0) {
+ oldestTuner = mModuleClients[i];
+ }
+ allocatedTuners++;
+ }
+ }
+ }
+
+ const struct radio_tuner *halTuner;
+ sp<ModuleClient> preemtedClient;
+ if (audio) {
+ if (allocatedAudio >= mProperties.num_audio_sources) {
+ ALOG_ASSERT(oldestAudio != 0, "addClient() allocatedAudio/oldestAudio mismatch");
+ preemtedClient = oldestAudio;
+ }
+ } else {
+ if (allocatedAudio + allocatedTuners >= mProperties.num_tuners) {
+ if (allocatedTuners != 0) {
+ ALOG_ASSERT(oldestTuner != 0, "addClient() allocatedTuners/oldestTuner mismatch");
+ preemtedClient = oldestTuner;
+ } else {
+ ALOG_ASSERT(oldestAudio != 0, "addClient() allocatedAudio/oldestAudio mismatch");
+ preemtedClient = oldestAudio;
+ }
+ }
+ }
+ if (preemtedClient != 0) {
+ halTuner = preemtedClient->getTuner();
+ preemtedClient->setTuner(NULL);
+ mHwDevice->close_tuner(mHwDevice, halTuner);
+ if (preemtedClient->audio()) {
+ notifyDeviceConnection(false, "");
+ }
+ }
+
+ ret = mHwDevice->open_tuner(mHwDevice, &halConfig, audio,
+ RadioService::callback, moduleClient->callbackThread().get(),
+ &halTuner);
+ if (ret == 0) {
+ ALOGV("addClient() setTuner %p", halTuner);
+ moduleClient->setTuner(halTuner);
+ mModuleClients.add(moduleClient);
+ if (audio) {
+ notifyDeviceConnection(true, "");
+ }
+ } else {
+ moduleClient.clear();
+ }
+
+
+ ALOGV("addClient() DONE moduleClient %p", moduleClient.get());
+
+ return moduleClient;
+}
+
+void RadioService::Module::removeClient(const sp<ModuleClient>& moduleClient) {
+ ALOGV("removeClient()");
+ AutoMutex lock(mLock);
+ int ret;
+ ssize_t index = -1;
+
+ for (size_t i = 0; i < mModuleClients.size(); i++) {
+ if (mModuleClients[i] == moduleClient) {
+ index = i;
+ break;
+ }
+ }
+ if (index == -1) {
+ return;
+ }
+
+ mModuleClients.removeAt(index);
+ const struct radio_tuner *halTuner = moduleClient->getTuner();
+ if (halTuner == NULL) {
+ return;
+ }
+
+ mHwDevice->close_tuner(mHwDevice, halTuner);
+ if (moduleClient->audio()) {
+ notifyDeviceConnection(false, "");
+ }
+
+ mMute = true;
+
+ if (mModuleClients.isEmpty()) {
+ return;
+ }
+
+ // Tuner reallocation logic:
+ // When a client is removed and was controlling a tuner, this tuner will be allocated to a
+ // previously preempted client. This client will be notified by a callback with
+ // RADIO_EVENT_CONTROL indicating gain of control.
+ // - If a preempted client is waiting for an audio source and one becomes available:
+ // Allocate the tuner to the most recently added client waiting for an audio source
+ // - If not:
+ // Allocate the tuner to the most recently added client.
+ // Each time a tuner using the audio source is opened or closed, the audio policy manager is
+ // notified of the connection or disconnection of AUDIO_DEVICE_IN_FM_TUNER.
+
+ sp<ModuleClient> youngestClient;
+ sp<ModuleClient> youngestClientAudio;
+ size_t allocatedTuners = 0;
+ size_t allocatedAudio = 0;
+ for (ssize_t i = mModuleClients.size() - 1; i >= 0; i--) {
+ if (mModuleClients[i]->getTuner() == NULL) {
+ if (mModuleClients[i]->audio()) {
+ if (youngestClientAudio == 0) {
+ youngestClientAudio = mModuleClients[i];
+ }
+ } else {
+ if (youngestClient == 0) {
+ youngestClient = mModuleClients[i];
+ }
+ }
+ } else {
+ if (mModuleClients[i]->audio()) {
+ allocatedAudio++;
+ } else {
+ allocatedTuners++;
+ }
+ }
+ }
+
+ ALOG_ASSERT(allocatedTuners + allocatedAudio < mProperties.num_tuners,
+ "removeClient() removed client but no tuner available");
+
+ ALOG_ASSERT(!moduleClient->audio() || allocatedAudio < mProperties.num_audio_sources,
+ "removeClient() removed audio client but no tuner with audio available");
+
+ if (allocatedAudio < mProperties.num_audio_sources && youngestClientAudio != 0) {
+ youngestClient = youngestClientAudio;
+ }
+
+ ALOG_ASSERT(youngestClient != 0, "removeClient() removed client no candidate found for tuner");
+
+ struct radio_hal_band_config halConfig = youngestClient->halConfig();
+ ret = mHwDevice->open_tuner(mHwDevice, &halConfig, youngestClient->audio(),
+ RadioService::callback, moduleClient->callbackThread().get(),
+ &halTuner);
+
+ if (ret == 0) {
+ youngestClient->setTuner(halTuner);
+ if (youngestClient->audio()) {
+ notifyDeviceConnection(true, "");
+ }
+ }
+}
+
+status_t RadioService::Module::setMute(bool mute)
+{
+ Mutex::Autolock _l(mLock);
+ if (mute != mMute) {
+ mMute = mute;
+ //TODO notifify audio policy manager of media activity on radio audio device
+ }
+ return NO_ERROR;
+}
+
+status_t RadioService::Module::getMute(bool *mute)
+{
+ Mutex::Autolock _l(mLock);
+ *mute = mMute;
+ return NO_ERROR;
+}
+
+
+const struct radio_band_config *RadioService::Module::getDefaultConfig() const
+{
+ if (mProperties.num_bands == 0) {
+ return NULL;
+ }
+ return &mProperties.bands[0];
+}
+
+void RadioService::Module::notifyDeviceConnection(bool connected,
+ const char *address) {
+ int64_t token = IPCThreadState::self()->clearCallingIdentity();
+ AudioSystem::setDeviceConnectionState(AUDIO_DEVICE_IN_FM_TUNER,
+ connected ? AUDIO_POLICY_DEVICE_STATE_AVAILABLE :
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address, kRadioTunerAudioDeviceName);
+ IPCThreadState::self()->restoreCallingIdentity(token);
+}
+
+#undef LOG_TAG
+#define LOG_TAG "RadioService::ModuleClient"
+
+RadioService::ModuleClient::ModuleClient(const sp<Module>& module,
+ const sp<IRadioClient>& client,
+ const struct radio_band_config *config,
+ bool audio)
+ : mModule(module), mClient(client), mConfig(*config), mAudio(audio), mTuner(NULL)
+{
+}
+
+void RadioService::ModuleClient::onFirstRef()
+{
+ mCallbackThread = new CallbackThread(this);
+ IInterface::asBinder(mClient)->linkToDeath(this);
+}
+
+RadioService::ModuleClient::~ModuleClient() {
+ if (mClient != 0) {
+ IInterface::asBinder(mClient)->unlinkToDeath(this);
+ mClient.clear();
+ }
+ if (mCallbackThread != 0) {
+ mCallbackThread->exit();
+ }
+}
+
+status_t RadioService::ModuleClient::dump(int fd __unused,
+ const Vector<String16>& args __unused) {
+ String8 result;
+ return NO_ERROR;
+}
+
+void RadioService::ModuleClient::detach() {
+ ALOGV("%s", __FUNCTION__);
+ sp<ModuleClient> strongMe = this;
+ {
+ AutoMutex lock(mLock);
+ if (mClient != 0) {
+ IInterface::asBinder(mClient)->unlinkToDeath(this);
+ mClient.clear();
+ }
+ }
+ sp<Module> module = mModule.promote();
+ if (module == 0) {
+ return;
+ }
+ module->removeClient(this);
+}
+
+radio_hal_band_config_t RadioService::ModuleClient::halConfig() const
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s locked", __FUNCTION__);
+ return mConfig.band;
+}
+
+const struct radio_tuner *RadioService::ModuleClient::getTuner() const
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s locked", __FUNCTION__);
+ return mTuner;
+}
+
+void RadioService::ModuleClient::setTuner(const struct radio_tuner *tuner)
+{
+ ALOGV("%s %p", __FUNCTION__, this);
+
+ AutoMutex lock(mLock);
+ mTuner = tuner;
+ ALOGV("%s locked", __FUNCTION__);
+
+ radio_hal_event_t event;
+ event.type = RADIO_EVENT_CONTROL;
+ event.status = 0;
+ event.on = mTuner != NULL;
+ mCallbackThread->sendEvent(&event);
+ ALOGV("%s DONE", __FUNCTION__);
+
+}
+
+status_t RadioService::ModuleClient::setConfiguration(const struct radio_band_config *config)
+{
+ AutoMutex lock(mLock);
+ status_t status = NO_ERROR;
+ ALOGV("%s locked", __FUNCTION__);
+
+ if (mTuner != NULL) {
+ struct radio_hal_band_config halConfig;
+ halConfig = config->band;
+ status = (status_t)mTuner->set_configuration(mTuner, &halConfig);
+ if (status == NO_ERROR) {
+ mConfig = *config;
+ }
+ } else {
+ mConfig = *config;
+ status == INVALID_OPERATION;
+ }
+
+ return status;
+}
+
+status_t RadioService::ModuleClient::getConfiguration(struct radio_band_config *config)
+{
+ AutoMutex lock(mLock);
+ status_t status = NO_ERROR;
+ ALOGV("%s locked", __FUNCTION__);
+
+ if (mTuner != NULL) {
+ struct radio_hal_band_config halConfig;
+ status = (status_t)mTuner->get_configuration(mTuner, &halConfig);
+ if (status == NO_ERROR) {
+ mConfig.band = halConfig;
+ }
+ }
+ *config = mConfig;
+
+ return status;
+}
+
+status_t RadioService::ModuleClient::setMute(bool mute)
+{
+ sp<Module> module;
+ {
+ Mutex::Autolock _l(mLock);
+ ALOGV("%s locked", __FUNCTION__);
+ if (mTuner == NULL || !mAudio) {
+ return INVALID_OPERATION;
+ }
+ module = mModule.promote();
+ if (module == 0) {
+ return NO_INIT;
+ }
+ }
+ module->setMute(mute);
+ return NO_ERROR;
+}
+
+status_t RadioService::ModuleClient::getMute(bool *mute)
+{
+ sp<Module> module;
+ {
+ Mutex::Autolock _l(mLock);
+ ALOGV("%s locked", __FUNCTION__);
+ module = mModule.promote();
+ if (module == 0) {
+ return NO_INIT;
+ }
+ }
+ return module->getMute(mute);
+}
+
+status_t RadioService::ModuleClient::scan(radio_direction_t direction, bool skipSubChannel)
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s locked", __FUNCTION__);
+ status_t status;
+ if (mTuner != NULL) {
+ status = (status_t)mTuner->scan(mTuner, direction, skipSubChannel);
+ } else {
+ status = INVALID_OPERATION;
+ }
+ return status;
+}
+
+status_t RadioService::ModuleClient::step(radio_direction_t direction, bool skipSubChannel)
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s locked", __FUNCTION__);
+ status_t status;
+ if (mTuner != NULL) {
+ status = (status_t)mTuner->step(mTuner, direction, skipSubChannel);
+ } else {
+ status = INVALID_OPERATION;
+ }
+ return status;
+}
+
+status_t RadioService::ModuleClient::tune(unsigned int channel, unsigned int subChannel)
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s locked", __FUNCTION__);
+ status_t status;
+ if (mTuner != NULL) {
+ status = (status_t)mTuner->tune(mTuner, channel, subChannel);
+ } else {
+ status = INVALID_OPERATION;
+ }
+ return status;
+}
+
+status_t RadioService::ModuleClient::cancel()
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s locked", __FUNCTION__);
+ status_t status;
+ if (mTuner != NULL) {
+ status = (status_t)mTuner->cancel(mTuner);
+ } else {
+ status = INVALID_OPERATION;
+ }
+ return status;
+}
+
+status_t RadioService::ModuleClient::getProgramInformation(struct radio_program_info *info)
+{
+ AutoMutex lock(mLock);
+ ALOGV("%s locked", __FUNCTION__);
+ status_t status;
+ if (mTuner != NULL) {
+ status = (status_t)mTuner->get_program_information(mTuner, info);
+ } else {
+ status = INVALID_OPERATION;
+ }
+ return status;
+}
+
+status_t RadioService::ModuleClient::hasControl(bool *hasControl)
+{
+ Mutex::Autolock lock(mLock);
+ ALOGV("%s locked", __FUNCTION__);
+ *hasControl = mTuner != NULL;
+ return NO_ERROR;
+}
+
+void RadioService::ModuleClient::onCallbackEvent(const sp<IMemory>& eventMemory)
+{
+ if (eventMemory == 0 || eventMemory->pointer() == NULL) {
+ return;
+ }
+
+ sp<IRadioClient> client;
+ {
+ AutoMutex lock(mLock);
+ ALOGV("%s locked", __FUNCTION__);
+ radio_event_t *event = (radio_event_t *)eventMemory->pointer();
+ switch (event->type) {
+ case RADIO_EVENT_CONFIG:
+ mConfig.band = event->config.band;
+ event->config.region = mConfig.region;
+ break;
+ default:
+ break;
+ }
+
+ client = mClient;
+ }
+ if (client != 0) {
+ client->onEvent(eventMemory);
+ }
+}
+
+
+void RadioService::ModuleClient::binderDied(
+ const wp<IBinder> &who __unused) {
+ ALOGW("client binder died for client %p", this);
+ detach();
+}
+
+}; // namespace android
diff --git a/services/radio/RadioService.h b/services/radio/RadioService.h
new file mode 100644
index 0000000..49feda6
--- /dev/null
+++ b/services/radio/RadioService.h
@@ -0,0 +1,211 @@
+/*
+ * Copyright (C) 2015 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_RADIO_SERVICE_H
+#define ANDROID_HARDWARE_RADIO_SERVICE_H
+
+#include <utils/Vector.h>
+//#include <binder/AppOpsManager.h>
+#include <binder/MemoryDealer.h>
+#include <binder/BinderService.h>
+#include <binder/IAppOpsCallback.h>
+#include <radio/IRadioService.h>
+#include <radio/IRadio.h>
+#include <radio/IRadioClient.h>
+#include <system/radio.h>
+#include <hardware/radio.h>
+
+namespace android {
+
+class MemoryHeapBase;
+
+class RadioService :
+ public BinderService<RadioService>,
+ public BnRadioService
+{
+ friend class BinderService<RadioService>;
+
+public:
+ class ModuleClient;
+ class Module;
+
+ static char const* getServiceName() { return "media.radio"; }
+
+ RadioService();
+ virtual ~RadioService();
+
+ // IRadioService
+ virtual status_t listModules(struct radio_properties *properties,
+ uint32_t *numModules);
+
+ virtual status_t attach(radio_handle_t handle,
+ const sp<IRadioClient>& client,
+ const struct radio_band_config *config,
+ bool withAudio,
+ sp<IRadio>& radio);
+
+ virtual status_t onTransact(uint32_t code, const Parcel& data,
+ Parcel* reply, uint32_t flags);
+
+ virtual status_t dump(int fd, const Vector<String16>& args);
+
+
+ class Module : public virtual RefBase {
+ public:
+
+ Module(radio_hw_device* hwDevice,
+ struct radio_properties properties);
+
+ virtual ~Module();
+
+ sp<ModuleClient> addClient(const sp<IRadioClient>& client,
+ const struct radio_band_config *config,
+ bool audio);
+
+ void removeClient(const sp<ModuleClient>& moduleClient);
+
+ status_t setMute(bool mute);
+
+ status_t getMute(bool *mute);
+
+ virtual status_t dump(int fd, const Vector<String16>& args);
+
+ const struct radio_hw_device *hwDevice() const { return mHwDevice; }
+ const struct radio_properties properties() const { return mProperties; }
+ const struct radio_band_config *getDefaultConfig() const ;
+
+ private:
+
+ void notifyDeviceConnection(bool connected, const char *address);
+
+ Mutex mLock; // protects mModuleClients
+ const struct radio_hw_device *mHwDevice; // HAL hardware device
+ const struct radio_properties mProperties; // cached hardware module properties
+ Vector< sp<ModuleClient> > mModuleClients; // list of attached clients
+ bool mMute; // radio audio source state
+ // when unmuted, audio is routed to the
+ // output device selected for media use case.
+ }; // class Module
+
+ class CallbackThread : public Thread {
+ public:
+
+ CallbackThread(const wp<ModuleClient>& moduleClient);
+
+ virtual ~CallbackThread();
+
+
+ // Thread virtuals
+ virtual bool threadLoop();
+
+ // RefBase
+ virtual void onFirstRef();
+
+ void exit();
+
+ void sendEvent(radio_hal_event_t *halEvent);
+ sp<IMemory> prepareEvent(radio_hal_event_t *halEvent);
+
+ private:
+ wp<ModuleClient> mModuleClient; // client module the thread belongs to
+ Condition mCallbackCond; // condition signaled when a new event is posted
+ Mutex mCallbackLock; // protects mEventQueue
+ Vector< sp<IMemory> > mEventQueue; // pending callback events
+ sp<MemoryDealer> mMemoryDealer; // shared memory for callback event
+ }; // class CallbackThread
+
+ class ModuleClient : public BnRadio,
+ public IBinder::DeathRecipient {
+ public:
+
+ ModuleClient(const sp<Module>& module,
+ const sp<IRadioClient>& client,
+ const struct radio_band_config *config,
+ bool audio);
+
+ virtual ~ModuleClient();
+
+ // IRadio
+ virtual void detach();
+
+ virtual status_t setConfiguration(const struct radio_band_config *config);
+
+ virtual status_t getConfiguration(struct radio_band_config *config);
+
+ virtual status_t setMute(bool mute);
+
+ virtual status_t getMute(bool *mute);
+
+ virtual status_t scan(radio_direction_t direction, bool skipSubChannel);
+
+ virtual status_t step(radio_direction_t direction, bool skipSubChannel);
+
+ virtual status_t tune(unsigned int channel, unsigned int subChannel);
+
+ virtual status_t cancel();
+
+ virtual status_t getProgramInformation(struct radio_program_info *info);
+
+ virtual status_t hasControl(bool *hasControl);
+
+ virtual status_t dump(int fd, const Vector<String16>& args);
+
+ sp<IRadioClient> client() const { return mClient; }
+ wp<Module> module() const { return mModule; }
+ radio_hal_band_config_t halConfig() const;
+ sp<CallbackThread> callbackThread() const { return mCallbackThread; }
+ void setTuner(const struct radio_tuner *tuner);
+ const struct radio_tuner *getTuner() const;
+ bool audio() const { return mAudio; }
+
+ void onCallbackEvent(const sp<IMemory>& event);
+
+ virtual void onFirstRef();
+
+
+ // IBinder::DeathRecipient implementation
+ virtual void binderDied(const wp<IBinder> &who);
+
+ private:
+
+ mutable Mutex mLock; // protects mClient, mConfig and mTuner
+ wp<Module> mModule; // The module this client is attached to
+ sp<IRadioClient> mClient; // event callback binder interface
+ radio_band_config_t mConfig; // current band configuration
+ sp<CallbackThread> mCallbackThread; // event callback thread
+ const bool mAudio;
+ const struct radio_tuner *mTuner; // HAL tuner interface. NULL indicates that
+ // this client does not have control on any
+ // tuner
+ }; // class ModuleClient
+
+
+ static void callback(radio_hal_event_t *halEvent, void *cookie);
+
+private:
+
+ virtual void onFirstRef();
+
+ static void convertProperties(radio_properties_t *properties,
+ const radio_hal_properties_t *halProperties);
+ Mutex mServiceLock; // protects mModules
+ volatile int32_t mNextUniqueId; // for module ID allocation
+ DefaultKeyedVector< radio_handle_t, sp<Module> > mModules;
+};
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_RADIO_SERVICE_H
diff --git a/tools/resampler_tools/Android.mk b/tools/resampler_tools/Android.mk
index e8cbe39..b58e4cd 100644
--- a/tools/resampler_tools/Android.mk
+++ b/tools/resampler_tools/Android.mk
@@ -1,6 +1,6 @@
# Copyright 2005 The Android Open Source Project
#
-# Android.mk for resampler_tools
+# Android.mk for resampler_tools
#
diff --git a/tools/resampler_tools/fir.cpp b/tools/resampler_tools/fir.cpp
index 62eddca..fe4d212 100644
--- a/tools/resampler_tools/fir.cpp
+++ b/tools/resampler_tools/fir.cpp
@@ -66,19 +66,20 @@
static void usage(char* name) {
fprintf(stderr,
- "usage: %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]"
+ "usage: %s [-h] [-d] [-D] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]"
" [-f {float|fixed|fixed16}] [-b beta] [-v dBFS] [-l lerp]\n"
- " %s [-h] [-d] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]"
+ " %s [-h] [-d] [-D] [-s sample_rate] [-c cut-off_frequency] [-n half_zero_crossings]"
" [-f {float|fixed|fixed16}] [-b beta] [-v dBFS] -p M/N\n"
" -h this help message\n"
" -d debug, print comma-separated coefficient table\n"
+ " -D generate extra declarations\n"
" -p generate poly-phase filter coefficients, with sample increment M/N\n"
" -s sample rate (48000)\n"
" -c cut-off frequency (20478)\n"
" -n number of zero-crossings on one side (8)\n"
" -l number of lerping bits (4)\n"
" -m number of polyphases (related to -l, default 16)\n"
- " -f output format, can be fixed-point or floating-point (fixed)\n"
+ " -f output format, can be fixed, fixed16, or float (fixed)\n"
" -b kaiser window parameter beta (7.865 [-80dB])\n"
" -v attenuation in dBFS (0)\n",
name, name
@@ -97,7 +98,8 @@
double Fs = 48000;
double Fc = 20478;
double atten = 1;
- int format = 0;
+ int format = 0; // 0=fixed, 1=float
+ bool declarations = false;
// in order to keep the errors associated with the linear
// interpolation of the coefficients below the quantization error
@@ -158,11 +160,14 @@
int M = 1 << 4; // number of phases for interpolation
int ch;
- while ((ch = getopt(argc, argv, ":hds:c:n:f:l:m:b:p:v:z:")) != -1) {
+ while ((ch = getopt(argc, argv, ":hds:c:n:f:l:m:b:p:v:z:D")) != -1) {
switch (ch) {
case 'd':
debug = true;
break;
+ case 'D':
+ declarations = true;
+ break;
case 'p':
if (sscanf(optarg, "%u/%u", &polyM, &polyN) != 2) {
usage(argv[0]);
@@ -225,24 +230,26 @@
for (int i = M-1 ; i; i>>=1, nz++);
// generate the right half of the filter
if (!debug) {
- printf("// cmd-line: ");
- for (int i=1 ; i<argc ; i++) {
- printf("%s ", argv[i]);
+ printf("// cmd-line:");
+ for (int i=0 ; i<argc ; i++) {
+ printf(" %s", argv[i]);
}
printf("\n");
- if (!polyphase) {
- printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", N);
- printf("const int32_t RESAMPLE_FIR_INT_PHASES = %d;\n", M);
- printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", nzc);
- } else {
- printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", 2*nzc*polyN);
- printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", 2*nzc);
+ if (declarations) {
+ if (!polyphase) {
+ printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", N);
+ printf("const int32_t RESAMPLE_FIR_INT_PHASES = %d;\n", M);
+ printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", nzc);
+ } else {
+ printf("const int32_t RESAMPLE_FIR_SIZE = %d;\n", 2*nzc*polyN);
+ printf("const int32_t RESAMPLE_FIR_NUM_COEF = %d;\n", 2*nzc);
+ }
+ if (!format) {
+ printf("const int32_t RESAMPLE_FIR_COEF_BITS = %d;\n", nc);
+ }
+ printf("\n");
+ printf("static %s resampleFIR[] = {", !format ? "int32_t" : "float");
}
- if (!format) {
- printf("const int32_t RESAMPLE_FIR_COEF_BITS = %d;\n", nc);
- }
- printf("\n");
- printf("static %s resampleFIR[] = {", !format ? "int32_t" : "float");
}
if (!polyphase) {
@@ -260,12 +267,15 @@
if (!format) {
int64_t yi = toint(y, 1ULL<<(nc-1));
if (nc > 16) {
- printf("0x%08x, ", int32_t(yi));
+ printf("0x%08x,", int32_t(yi));
} else {
- printf("0x%04x, ", int32_t(yi)&0xffff);
+ printf("0x%04x,", int32_t(yi)&0xffff);
}
} else {
- printf("%.9g%s ", y, debug ? "," : "f,");
+ printf("%.9g%s", y, debug ? "," : "f,");
+ }
+ if (j != nzc-1) {
+ printf(" ");
}
}
}
@@ -283,23 +293,22 @@
if (!format) {
int64_t yi = toint(y, 1ULL<<(nc-1));
if (nc > 16) {
- printf("0x%08x, ", int32_t(yi));
+ printf("0x%08x,", int32_t(yi));
} else {
- printf("0x%04x, ", int32_t(yi)&0xffff);
+ printf("0x%04x,", int32_t(yi)&0xffff);
}
} else {
- printf("%.9g%s", y, debug ? "" : "f");
+ printf("%.9g%s", y, debug ? "," : "f,");
}
- if (debug && (i==nzc-1)) {
- } else {
- printf(", ");
+ if (i != nzc-1) {
+ printf(" ");
}
}
}
}
- if (!debug) {
+ if (!debug && declarations) {
printf("\n};");
}
printf("\n");
