Merge changes Ic355174a,I8b04ba9e,I3fefb13c

* changes:
  libhardware_legacy: Android.mk -> Android.bp
  remove legacy audio policy
  libhardware_legacy doesn't need libmedia
diff --git a/Android.bp b/Android.bp
index 087342a..d3a3a9c 100644
--- a/Android.bp
+++ b/Android.bp
@@ -1,10 +1,51 @@
 // Copyright 2006 The Android Open Source Project
 
+subdirs = [
+    "audio",
+]
+
+cc_library_headers {
+    name: "libhardware_legacy_headers",
+    export_include_dirs: ["include"],
+
+    header_libs: ["libcutils_headers"],
+    export_header_lib_headers: ["libcutils_headers"],
+}
+
 cc_library {
     name: "libpower",
 
-    srcs: ["power/power.c"],
+    srcs: ["power.c"],
     export_include_dirs: ["include"],
     shared_libs: ["libcutils", "liblog"],
     vendor_available: true,
 }
+
+cc_library_shared {
+    name: "libhardware_legacy",
+
+    shared_libs: [
+        "libbase",
+        "libdl",
+        "libcutils",
+        "liblog",
+    ],
+
+    header_libs: [
+        "libhardware_legacy_headers",
+    ],
+    export_header_lib_headers: ["libhardware_legacy_headers"],
+
+    export_include_dirs: ["include"],
+
+    cflags: [
+        "-DQEMU_HARDWARE",
+        "-Wno-unused-parameter",
+        "-Wno-gnu-designator",
+    ],
+
+    srcs: [
+        "power.c",
+        "uevent.c",
+    ],
+}
diff --git a/Android.mk b/Android.mk
deleted file mode 100644
index d17ba64..0000000
--- a/Android.mk
+++ /dev/null
@@ -1,41 +0,0 @@
-# Copyright 2006 The Android Open Source Project
-
-# Setting LOCAL_PATH will mess up all-subdir-makefiles, so do it beforehand.
-legacy_modules := power uevent
-
-SAVE_MAKEFILES := $(call all-named-subdir-makefiles,$(legacy_modules))
-LEGACY_AUDIO_MAKEFILES := $(call all-named-subdir-makefiles,audio)
-
-LOCAL_PATH:= $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_SHARED_LIBRARIES := libbase libcutils liblog libmedia
-LOCAL_EXPORT_SHARED_LIBRARY_HEADERS := libmedia
-
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/include
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/include
-
-LOCAL_CFLAGS  += -DQEMU_HARDWARE -Wno-unused-parameter -Wno-gnu-designator
-QEMU_HARDWARE := true
-
-LOCAL_SHARED_LIBRARIES += libdl
-
-include $(SAVE_MAKEFILES)
-
-# TODO: Remove this line b/29915755
-ifndef BRILLO
-LOCAL_WHOLE_STATIC_LIBRARIES := libwifi-hal-common
-endif
-
-LOCAL_MODULE:= libhardware_legacy
-
-include $(BUILD_SHARED_LIBRARY)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := libhardware_legacy_headers
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/include
-include $(BUILD_HEADER_LIBRARY)
-
-# legacy_audio builds it's own set of libraries that aren't linked into
-# hardware_legacy
-include $(LEGACY_AUDIO_MAKEFILES)
diff --git a/audio/Android.bp b/audio/Android.bp
new file mode 100644
index 0000000..5141dee
--- /dev/null
+++ b/audio/Android.bp
@@ -0,0 +1,25 @@
+// Copyright 2011 The Android Open Source Project
+
+//AUDIO_POLICY_TEST := true
+//ENABLE_AUDIO_DUMP := true
+
+cc_library_static {
+
+    srcs: [
+        "AudioHardwareInterface.cpp",
+        "audio_hw_hal.cpp",
+    ],
+
+    name: "libaudiohw_legacy",
+    static_libs: ["libmedia_helper"],
+    cflags: [
+        "-Wno-unused-parameter",
+        "-Wno-gnu-designator",
+    ],
+
+    header_libs: [
+        "libbase_headers",
+        "libhardware_legacy_headers",
+    ],
+    export_header_lib_headers: ["libhardware_legacy_headers"],
+}
diff --git a/audio/Android.mk b/audio/Android.mk
deleted file mode 100644
index a64c6b8..0000000
--- a/audio/Android.mk
+++ /dev/null
@@ -1,67 +0,0 @@
-# Copyright 2011 The Android Open Source Project
-
-#AUDIO_POLICY_TEST := true
-#ENABLE_AUDIO_DUMP := true
-
-LOCAL_PATH := $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
-    AudioHardwareInterface.cpp \
-    audio_hw_hal.cpp
-
-LOCAL_MODULE := libaudiohw_legacy
-LOCAL_SHARED_LIBRARIES := libmedia
-LOCAL_STATIC_LIBRARIES := libmedia_helper
-LOCAL_CFLAGS := -Wno-unused-parameter -Wno-gnu-designator
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../include
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/../include
-
-include $(BUILD_STATIC_LIBRARY)
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
-    AudioPolicyManagerBase.cpp \
-    AudioPolicyCompatClient.cpp \
-    audio_policy_hal.cpp
-
-ifeq ($(AUDIO_POLICY_TEST),true)
-  LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
-endif
-
-LOCAL_SHARED_LIBRARIES := libmedia
-LOCAL_STATIC_LIBRARIES := libmedia_helper
-LOCAL_MODULE := libaudiopolicy_legacy
-LOCAL_CFLAGS += -Wno-unused-parameter
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../include
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)/../include
-
-include $(BUILD_STATIC_LIBRARY)
-
-# The default audio policy, for now still implemented on top of legacy
-# policy code
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
-    AudioPolicyManagerDefault.cpp
-
-LOCAL_SHARED_LIBRARIES := \
-    libcutils \
-    libmedia \
-    libutils \
-    liblog
-
-LOCAL_STATIC_LIBRARIES := \
-    libmedia_helper
-
-LOCAL_WHOLE_STATIC_LIBRARIES := \
-    libaudiopolicy_legacy
-
-LOCAL_MODULE := audio_policy.default
-LOCAL_MODULE_RELATIVE_PATH := hw
-LOCAL_CFLAGS := -Wno-unused-parameter
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../include
-
-include $(BUILD_SHARED_LIBRARY)
-
diff --git a/audio/AudioPolicyCompatClient.cpp b/audio/AudioPolicyCompatClient.cpp
deleted file mode 100644
index e2ee222..0000000
--- a/audio/AudioPolicyCompatClient.cpp
+++ /dev/null
@@ -1,147 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "AudioPolicyCompatClient"
-//#define LOG_NDEBUG 0
-
-#include <stdint.h>
-
-#include <hardware/hardware.h>
-#include <system/audio.h>
-#include <system/audio_policy.h>
-#include <hardware/audio_policy.h>
-
-#include <hardware_legacy/AudioSystemLegacy.h>
-
-#include "AudioPolicyCompatClient.h"
-
-namespace android_audio_legacy {
-
-audio_module_handle_t AudioPolicyCompatClient::loadHwModule(const char *moduleName)
-{
-    return mServiceOps->load_hw_module(mService, moduleName);
-}
-
-audio_io_handle_t AudioPolicyCompatClient::openOutput(audio_module_handle_t module,
-                                                      audio_devices_t *pDevices,
-                                                      uint32_t *pSamplingRate,
-                                                      audio_format_t *pFormat,
-                                                      audio_channel_mask_t *pChannelMask,
-                                                      uint32_t *pLatencyMs,
-                                                      audio_output_flags_t flags,
-                                                      const audio_offload_info_t *offloadInfo)
-{
-    return mServiceOps->open_output_on_module(mService, module, pDevices, pSamplingRate,
-                                              pFormat, pChannelMask, pLatencyMs,
-                                              flags, offloadInfo);
-}
-
-audio_io_handle_t AudioPolicyCompatClient::openDuplicateOutput(audio_io_handle_t output1,
-                                                          audio_io_handle_t output2)
-{
-    return mServiceOps->open_duplicate_output(mService, output1, output2);
-}
-
-status_t AudioPolicyCompatClient::closeOutput(audio_io_handle_t output)
-{
-    return mServiceOps->close_output(mService, output);
-}
-
-status_t AudioPolicyCompatClient::suspendOutput(audio_io_handle_t output)
-{
-    return mServiceOps->suspend_output(mService, output);
-}
-
-status_t AudioPolicyCompatClient::restoreOutput(audio_io_handle_t output)
-{
-    return mServiceOps->restore_output(mService, output);
-}
-
-audio_io_handle_t AudioPolicyCompatClient::openInput(audio_module_handle_t module,
-                                                     audio_devices_t *pDevices,
-                                                     uint32_t *pSamplingRate,
-                                                     audio_format_t *pFormat,
-                                                     audio_channel_mask_t *pChannelMask)
-{
-    return mServiceOps->open_input_on_module(mService, module, pDevices,
-                                             pSamplingRate, pFormat, pChannelMask);
-}
-
-status_t AudioPolicyCompatClient::closeInput(audio_io_handle_t input)
-{
-    return mServiceOps->close_input(mService, input);
-}
-
-status_t AudioPolicyCompatClient::invalidateStream(AudioSystem::stream_type stream)
-{
-    return mServiceOps->invalidate_stream(mService, (audio_stream_type_t)stream);
-}
-
-status_t AudioPolicyCompatClient::moveEffects(audio_session_t session, audio_io_handle_t srcOutput,
-                                               audio_io_handle_t dstOutput)
-{
-    return mServiceOps->move_effects(mService, session, srcOutput, dstOutput);
-}
-
-String8 AudioPolicyCompatClient::getParameters(audio_io_handle_t ioHandle, const String8& keys)
-{
-    char *str;
-    String8 out_str8;
-
-    str = mServiceOps->get_parameters(mService, ioHandle, keys.string());
-    out_str8 = String8(str);
-    free(str);
-
-    return out_str8;
-}
-
-void AudioPolicyCompatClient::setParameters(audio_io_handle_t ioHandle,
-                                            const String8& keyValuePairs,
-                                            int delayMs)
-{
-    mServiceOps->set_parameters(mService, ioHandle, keyValuePairs.string(),
-                           delayMs);
-}
-
-status_t AudioPolicyCompatClient::setStreamVolume(
-                                             AudioSystem::stream_type stream,
-                                             float volume,
-                                             audio_io_handle_t output,
-                                             int delayMs)
-{
-    return mServiceOps->set_stream_volume(mService, (audio_stream_type_t)stream,
-                                          volume, output, delayMs);
-}
-
-status_t AudioPolicyCompatClient::startTone(ToneGenerator::tone_type tone,
-                                       AudioSystem::stream_type stream)
-{
-    return mServiceOps->start_tone(mService,
-                                   AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
-                                   (audio_stream_type_t)stream);
-}
-
-status_t AudioPolicyCompatClient::stopTone()
-{
-    return mServiceOps->stop_tone(mService);
-}
-
-status_t AudioPolicyCompatClient::setVoiceVolume(float volume, int delayMs)
-{
-    return mServiceOps->set_voice_volume(mService, volume, delayMs);
-}
-
-}; // namespace android_audio_legacy
diff --git a/audio/AudioPolicyCompatClient.h b/audio/AudioPolicyCompatClient.h
deleted file mode 100644
index 32804be..0000000
--- a/audio/AudioPolicyCompatClient.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_AUDIOPOLICYCLIENTLEGACY_H
-#define ANDROID_AUDIOPOLICYCLIENTLEGACY_H
-
-#include <system/audio.h>
-#include <system/audio_policy.h>
-#include <hardware/audio_policy.h>
-
-#include <hardware_legacy/AudioSystemLegacy.h>
-#include <hardware_legacy/AudioPolicyInterface.h>
-
-/************************************/
-/* FOR BACKWARDS COMPATIBILITY ONLY */
-/************************************/
-namespace android_audio_legacy {
-
-class AudioPolicyCompatClient : public AudioPolicyClientInterface {
-public:
-    AudioPolicyCompatClient(struct audio_policy_service_ops *serviceOps,
-                            void *service) :
-            mServiceOps(serviceOps) , mService(service) {}
-
-    virtual audio_module_handle_t loadHwModule(const char *moduleName);
-
-    virtual audio_io_handle_t openOutput(audio_module_handle_t module,
-                                         audio_devices_t *pDevices,
-                                         uint32_t *pSamplingRate,
-                                         audio_format_t *pFormat,
-                                         audio_channel_mask_t *pChannelMask,
-                                         uint32_t *pLatencyMs,
-                                         audio_output_flags_t flags,
-                                         const audio_offload_info_t *offloadInfo);
-    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
-                                                  audio_io_handle_t output2);
-    virtual status_t closeOutput(audio_io_handle_t output);
-    virtual status_t suspendOutput(audio_io_handle_t output);
-    virtual status_t restoreOutput(audio_io_handle_t output);
-    virtual audio_io_handle_t openInput(audio_module_handle_t module,
-                                        audio_devices_t *pDevices,
-                                        uint32_t *pSamplingRate,
-                                        audio_format_t *pFormat,
-                                        audio_channel_mask_t *pChannelMask);
-    virtual status_t closeInput(audio_io_handle_t input);
-    virtual status_t invalidateStream(AudioSystem::stream_type stream);
-    virtual status_t moveEffects(audio_session_t session,
-                                 audio_io_handle_t srcOutput,
-                                 audio_io_handle_t dstOutput);
-
-    virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
-    virtual void setParameters(audio_io_handle_t ioHandle,
-                               const String8& keyValuePairs,
-                               int delayMs = 0);
-    virtual status_t setStreamVolume(AudioSystem::stream_type stream,
-                                     float volume,
-                                     audio_io_handle_t output,
-                                     int delayMs = 0);
-    virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream);
-    virtual status_t stopTone();
-    virtual status_t setVoiceVolume(float volume, int delayMs = 0);
-
-private:
-    struct audio_policy_service_ops* mServiceOps;
-    void*                            mService;
-};
-
-}; // namespace android_audio_legacy
-
-#endif // ANDROID_AUDIOPOLICYCLIENTLEGACY_H
diff --git a/audio/AudioPolicyManagerBase.cpp b/audio/AudioPolicyManagerBase.cpp
deleted file mode 100644
index 74ee22a..0000000
--- a/audio/AudioPolicyManagerBase.cpp
+++ /dev/null
@@ -1,4368 +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 "AudioPolicyManagerBase"
-//#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
-// 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
-
-#include <inttypes.h>
-#include <math.h>
-
-#include <cutils/properties.h>
-#include <utils/Log.h>
-#include <utils/Timers.h>
-
-#include <hardware/audio.h>
-#include <hardware/audio_effect.h>
-#include <hardware_legacy/audio_policy_conf.h>
-#include <hardware_legacy/AudioPolicyManagerBase.h>
-
-namespace android_audio_legacy {
-
-// ----------------------------------------------------------------------------
-// AudioPolicyInterface implementation
-// ----------------------------------------------------------------------------
-
-
-status_t AudioPolicyManagerBase::setDeviceConnectionState(audio_devices_t device,
-                                                  AudioSystem::device_connection_state state,
-                                                  const char *device_address)
-{
-    // device_address can be NULL and should be handled as an empty string in this case,
-    // and it is not checked by AudioPolicyInterfaceImpl.cpp
-    if (device_address == NULL) {
-        device_address = "";
-    }
-    ALOGV("setDeviceConnectionState() device: 0x%X, state %d, address %s", device, state, device_address);
-
-    // connect/disconnect only 1 device at a time
-    if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
-
-    if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
-        ALOGE("setDeviceConnectionState() invalid address: %s", device_address);
-        return BAD_VALUE;
-    }
-
-    // handle output devices
-    if (audio_is_output_device(device)) {
-        SortedVector <audio_io_handle_t> outputs;
-
-        if (!mHasA2dp && audio_is_a2dp_out_device(device)) {
-            ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device);
-            return BAD_VALUE;
-        }
-        if (!mHasUsb && audio_is_usb_out_device(device)) {
-            ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device);
-            return BAD_VALUE;
-        }
-        if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) {
-            ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device);
-            return BAD_VALUE;
-        }
-
-        // 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;
-        String8 paramStr;
-        switch (state)
-        {
-        // handle output device connection
-        case AudioSystem::DEVICE_STATE_AVAILABLE:
-            if (mAvailableOutputDevices & device) {
-                ALOGW("setDeviceConnectionState() device already connected: %x", device);
-                return INVALID_OPERATION;
-            }
-            ALOGV("setDeviceConnectionState() connecting device %x", device);
-
-            if (mHasA2dp && audio_is_a2dp_out_device(device)) {
-                // handle A2DP device connection
-                AudioParameter param;
-                param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address));
-                paramStr = param.toString();
-            } else if (mHasUsb && audio_is_usb_out_device(device)) {
-                // handle USB device connection
-                paramStr = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
-            }
-
-            if (checkOutputsForDevice(device, state, outputs, paramStr) != NO_ERROR) {
-                return INVALID_OPERATION;
-            }
-            ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
-                  outputs.size());
-            // register new device as available
-            mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device);
-
-            if (mHasA2dp && audio_is_a2dp_out_device(device)) {
-                // handle A2DP device connection
-                mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
-                mA2dpSuspended = false;
-            } else if (audio_is_bluetooth_sco_device(device)) {
-                // handle SCO device connection
-                mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
-            } else if (mHasUsb && audio_is_usb_out_device(device)) {
-                // handle USB device connection
-                mUsbOutCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
-            }
-
-            break;
-        // handle output device disconnection
-        case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
-            if (!(mAvailableOutputDevices & device)) {
-                ALOGW("setDeviceConnectionState() device not connected: %x", device);
-                return INVALID_OPERATION;
-            }
-
-            ALOGV("setDeviceConnectionState() disconnecting device %x", device);
-            // remove device from available output devices
-            mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device);
-            checkOutputsForDevice(device, state, outputs, paramStr);
-
-            if (mHasA2dp && audio_is_a2dp_out_device(device)) {
-                // handle A2DP device disconnection
-                mA2dpDeviceAddress = "";
-                mA2dpSuspended = false;
-            } else if (audio_is_bluetooth_sco_device(device)) {
-                // handle SCO device disconnection
-                mScoDeviceAddress = "";
-            } else if (mHasUsb && audio_is_usb_out_device(device)) {
-                // handle USB device disconnection
-                mUsbOutCardAndDevice = "";
-            }
-            // not currently handling multiple simultaneous submixes: ignoring remote submix
-            //   case and address
-            } break;
-
-        default:
-            ALOGE("setDeviceConnectionState() invalid state: %x", state);
-            return BAD_VALUE;
-        }
-
-        checkA2dpSuspend();
-        checkOutputForAllStrategies();
-        // outputs must be closed after checkOutputForAllStrategies() is executed
-        if (!outputs.isEmpty()) {
-            for (size_t i = 0; i < outputs.size(); i++) {
-                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 == AudioSystem::DEVICE_STATE_UNAVAILABLE) ||
-                        (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
-                         (desc->mDirectOpenCount == 0))) {
-                    closeOutput(outputs[i]);
-                }
-            }
-        }
-
-        updateDevicesAndOutputs();
-        for (size_t i = 0; i < mOutputs.size(); i++) {
-            // 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.
-            setOutputDevice(mOutputs.keyAt(i),
-                            getNewDevice(mOutputs.keyAt(i), true /*fromCache*/),
-                            !mOutputs.valueAt(i)->isDuplicated(),
-                            0);
-        }
-
-        return NO_ERROR;
-    }  // end if is output device
-
-    // handle input devices
-    if (audio_is_input_device(device)) {
-        SortedVector <audio_io_handle_t> inputs;
-
-        String8 paramStr;
-        switch (state)
-        {
-        // handle input device connection
-        case AudioSystem::DEVICE_STATE_AVAILABLE: {
-            if (mAvailableInputDevices & device) {
-                ALOGW("setDeviceConnectionState() device already connected: %d", device);
-                return INVALID_OPERATION;
-            }
-
-            if (mHasUsb && audio_is_usb_in_device(device)) {
-                // handle USB device connection
-                paramStr = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
-            } else if (mHasA2dp && audio_is_a2dp_in_device(device)) {
-                // handle A2DP device connection
-                AudioParameter param;
-                param.add(String8(AUDIO_PARAMETER_A2DP_SOURCE_ADDRESS), String8(device_address));
-                paramStr = param.toString();
-            }
-
-            if (checkInputsForDevice(device, state, inputs, paramStr) != NO_ERROR) {
-                return INVALID_OPERATION;
-            }
-            mAvailableInputDevices = mAvailableInputDevices | (device & ~AUDIO_DEVICE_BIT_IN);
-            }
-            break;
-
-        // handle input device disconnection
-        case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
-            if (!(mAvailableInputDevices & device)) {
-                ALOGW("setDeviceConnectionState() device not connected: %d", device);
-                return INVALID_OPERATION;
-            }
-            checkInputsForDevice(device, state, inputs, paramStr);
-            mAvailableInputDevices = (audio_devices_t) (mAvailableInputDevices & ~device);
-        } break;
-
-        default:
-            ALOGE("setDeviceConnectionState() invalid state: %x", state);
-            return BAD_VALUE;
-        }
-
-        closeAllInputs();
-
-        return NO_ERROR;
-    } // end if is input device
-
-    ALOGW("setDeviceConnectionState() invalid device: %x", device);
-    return BAD_VALUE;
-}
-
-AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(audio_devices_t device,
-                                                  const char *device_address)
-{
-    // similar to setDeviceConnectionState
-    if (device_address == NULL) {
-        device_address = "";
-    }
-    AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
-    String8 address = String8(device_address);
-    if (audio_is_output_device(device)) {
-        if (device & mAvailableOutputDevices) {
-            if (audio_is_a2dp_out_device(device) &&
-                (!mHasA2dp || (address != "" && mA2dpDeviceAddress != address))) {
-                return state;
-            }
-            if (audio_is_bluetooth_sco_device(device) &&
-                address != "" && mScoDeviceAddress != address) {
-                return state;
-            }
-            if (audio_is_usb_out_device(device) &&
-                (!mHasUsb || (address != "" && mUsbOutCardAndDevice != address))) {
-                ALOGE("getDeviceConnectionState() invalid device: %x", device);
-                return state;
-            }
-            if (audio_is_remote_submix_device((audio_devices_t)device) && !mHasRemoteSubmix) {
-                return state;
-            }
-            state = AudioSystem::DEVICE_STATE_AVAILABLE;
-        }
-    } else if (audio_is_input_device(device)) {
-        if (device & mAvailableInputDevices) {
-            state = AudioSystem::DEVICE_STATE_AVAILABLE;
-        }
-    }
-
-    return state;
-}
-
-void AudioPolicyManagerBase::setPhoneState(int state)
-{
-    ALOGV("setPhoneState() state %d", state);
-    audio_devices_t newDevice = AUDIO_DEVICE_NONE;
-    if (state < 0 || state >= AudioSystem::NUM_MODES) {
-        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 < AudioSystem::NUM_STREAM_TYPES; stream++) {
-            handleIncallSonification(stream, false, true);
-        }
-    }
-
-    // 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
-    newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
-    checkA2dpSuspend();
-    checkOutputForAllStrategies();
-    updateDevicesAndOutputs();
-
-    AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
-
-    // force routing command to audio hardware when ending call
-    // even if no device change is needed
-    if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
-        newDevice = hwOutputDesc->device();
-    }
-
-    int delayMs = 0;
-    if (isStateInCall(state)) {
-        nsecs_t sysTime = systemTime();
-        for (size_t i = 0; i < mOutputs.size(); i++) {
-            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*/));
-        }
-    }
-
-    // change routing is necessary
-    setOutputDevice(mPrimaryOutput, newDevice, force, delayMs);
-
-    // 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 < AudioSystem::NUM_STREAM_TYPES; stream++) {
-            handleIncallSonification(stream, true, true);
-        }
-    }
-
-    // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
-    if (state == AudioSystem::MODE_RINGTONE &&
-        isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
-        mLimitRingtoneVolume = true;
-    } else {
-        mLimitRingtoneVolume = false;
-    }
-}
-
-void AudioPolicyManagerBase::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
-{
-    ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
-
-    bool forceVolumeReeval = false;
-    switch(usage) {
-    case AudioSystem::FOR_COMMUNICATION:
-        if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO &&
-            config != AudioSystem::FORCE_NONE) {
-            ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
-            return;
-        }
-        forceVolumeReeval = true;
-        mForceUse[usage] = config;
-        break;
-    case AudioSystem::FOR_MEDIA:
-        if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP &&
-            config != AudioSystem::FORCE_WIRED_ACCESSORY &&
-            config != AudioSystem::FORCE_ANALOG_DOCK &&
-            config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE &&
-            config != AudioSystem::FORCE_NO_BT_A2DP) {
-            ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
-            return;
-        }
-        mForceUse[usage] = config;
-        break;
-    case AudioSystem::FOR_RECORD:
-        if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY &&
-            config != AudioSystem::FORCE_NONE) {
-            ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
-            return;
-        }
-        mForceUse[usage] = config;
-        break;
-    case AudioSystem::FOR_DOCK:
-        if (config != AudioSystem::FORCE_NONE && config != AudioSystem::FORCE_BT_CAR_DOCK &&
-            config != AudioSystem::FORCE_BT_DESK_DOCK &&
-            config != AudioSystem::FORCE_WIRED_ACCESSORY &&
-            config != AudioSystem::FORCE_ANALOG_DOCK &&
-            config != AudioSystem::FORCE_DIGITAL_DOCK) {
-            ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
-        }
-        forceVolumeReeval = true;
-        mForceUse[usage] = config;
-        break;
-    case AudioSystem::FOR_SYSTEM:
-        if (config != AudioSystem::FORCE_NONE &&
-            config != AudioSystem::FORCE_SYSTEM_ENFORCED) {
-            ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
-        }
-        forceVolumeReeval = true;
-        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();
-    for (size_t i = 0; i < mOutputs.size(); i++) {
-        audio_io_handle_t output = mOutputs.keyAt(i);
-        audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/);
-        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) {
-        AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
-        audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
-        if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
-            ALOGV("setForceUse() changing device from %x to %x for input %d",
-                    inputDesc->mDevice, newDevice, activeInput);
-            inputDesc->mDevice = newDevice;
-            AudioParameter param = AudioParameter();
-            param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
-            mpClientInterface->setParameters(activeInput, param.toString());
-        }
-    }
-
-}
-
-AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
-{
-    return mForceUse[usage];
-}
-
-void AudioPolicyManagerBase::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
-AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::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++) {
-            IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
-            if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
-                if (profile->isCompatibleProfile(device, samplingRate, format,
-                                           channelMask,
-                                           AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
-                    if (mAvailableOutputDevices & profile->mSupportedDevices) {
-                        return mHwModules[i]->mOutputProfiles[j];
-                    }
-                }
-            } else {
-                if (profile->isCompatibleProfile(device, samplingRate, format,
-                                           channelMask,
-                                           AUDIO_OUTPUT_FLAG_DIRECT)) {
-                    if (mAvailableOutputDevices & profile->mSupportedDevices) {
-                        return mHwModules[i]->mOutputProfiles[j];
-                    }
-                }
-            }
-        }
-    }
-    return 0;
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
-                                    uint32_t samplingRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    AudioSystem::output_flags flags,
-                                    const audio_offload_info_t *offloadInfo)
-{
-    audio_io_handle_t output = 0;
-    uint32_t latency = 0;
-    routing_strategy strategy = getStrategy((AudioSystem::stream_type)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);
-
-#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");
-            AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
-            outputDesc->mDevice = mTestDevice;
-            outputDesc->mSamplingRate = mTestSamplingRate;
-            outputDesc->mFormat = mTestFormat;
-            outputDesc->mChannelMask = mTestChannels;
-            outputDesc->mLatency = mTestLatencyMs;
-            outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
-            outputDesc->mRefCount[stream] = 0;
-            mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
-                                            &outputDesc->mSamplingRate,
-                                            &outputDesc->mFormat,
-                                            &outputDesc->mChannelMask,
-                                            &outputDesc->mLatency,
-                                            outputDesc->mFlags,
-                                            offloadInfo);
-            if (mTestOutputs[mCurOutput]) {
-                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 = (AudioSystem::output_flags)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
-    }
-
-    // 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.
-    IOProfile *profile = NULL;
-    if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
-            !isNonOffloadableEffectEnabled()) {
-        profile = getProfileForDirectOutput(device,
-                                           samplingRate,
-                                           format,
-                                           channelMask,
-                                           (audio_output_flags_t)flags);
-    }
-
-    if (profile != NULL) {
-        AudioOutputDescriptor *outputDesc = NULL;
-
-        for (size_t i = 0; i < mOutputs.size(); i++) {
-            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->mId);
-        }
-        outputDesc = new AudioOutputDescriptor(profile);
-        outputDesc->mDevice = device;
-        outputDesc->mSamplingRate = samplingRate;
-        outputDesc->mFormat = format;
-        outputDesc->mChannelMask = channelMask;
-        outputDesc->mLatency = 0;
-        outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
-        outputDesc->mRefCount[stream] = 0;
-        outputDesc->mStopTime[stream] = 0;
-        outputDesc->mDirectOpenCount = 1;
-        output = mpClientInterface->openOutput(profile->mModule->mHandle,
-                                        &outputDesc->mDevice,
-                                        &outputDesc->mSamplingRate,
-                                        &outputDesc->mFormat,
-                                        &outputDesc->mChannelMask,
-                                        &outputDesc->mLatency,
-                                        outputDesc->mFlags,
-                                        offloadInfo);
-
-        // only accept an output with the requested parameters
-        if (output == 0 ||
-            (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
-            (format != AUDIO_FORMAT_DEFAULT && format != outputDesc->mFormat) ||
-            (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
-            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 != 0) {
-                mpClientInterface->closeOutput(output);
-            }
-            delete outputDesc;
-            return 0;
-        }
-        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);
-        return 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);
-
-        output = selectOutput(outputs, flags);
-    }
-    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 AudioPolicyManagerBase::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
-                                                       AudioSystem::output_flags flags)
-{
-    // 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++) {
-        AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]);
-        if (!outputDesc->isDuplicated()) {
-            int commonFlags = (int)AudioSystem::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 AudioPolicyManagerBase::startOutput(audio_io_handle_t output,
-                                             AudioSystem::stream_type 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;
-    }
-
-    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) {
-        audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
-        routing_strategy strategy = getStrategy(stream);
-        bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
-                            (strategy == STRATEGY_SONIFICATION_RESPECTFUL);
-        uint32_t waitMs = 0;
-        bool force = false;
-        for (size_t i = 0; i < mOutputs.size(); i++) {
-            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.
-                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);
-        if (waitMs > muteWaitMs) {
-            usleep((waitMs - muteWaitMs) * 2 * 1000);
-        }
-    }
-    return NO_ERROR;
-}
-
-
-status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output,
-                                            AudioSystem::stream_type 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;
-    }
-
-    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-
-    // 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) {
-            outputDesc->mStopTime[stream] = systemTime();
-            audio_devices_t newDevice = getNewDevice(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);
-                AudioOutputDescriptor *desc = mOutputs.valueAt(i);
-                if (curOutput != output &&
-                        desc->isActive() &&
-                        outputDesc->sharesHwModuleWith(desc) &&
-                        (newDevice != desc->device())) {
-                    setOutputDevice(curOutput,
-                                    getNewDevice(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 AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output)
-{
-    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) {
-        AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-        if (outputDesc->isActive()) {
-            mpClientInterface->closeOutput(output);
-            delete mOutputs.valueAt(index);
-            mOutputs.removeItem(output);
-            mTestOutputs[testIndex] = 0;
-        }
-        return;
-    }
-#endif //AUDIO_POLICY_TEST
-
-    AudioOutputDescriptor *desc = mOutputs.valueAt(index);
-    if (desc->mFlags & AudioSystem::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);
-            }
-        }
-    }
-}
-
-
-audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
-                                    uint32_t samplingRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    AudioSystem::audio_in_acoustics acoustics)
-{
-    audio_io_handle_t input = 0;
-    audio_devices_t device = getDeviceForInputSource(inputSource);
-
-    ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x",
-          inputSource, samplingRate, format, channelMask, acoustics);
-
-    if (device == AUDIO_DEVICE_NONE) {
-        ALOGW("getInput() could not find device for inputSource %d", inputSource);
-        return 0;
-    }
-
-    // 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;
-    }
-
-    IOProfile *profile = getInputProfile(device,
-                                         samplingRate,
-                                         format,
-                                         channelMask);
-    if (profile == NULL) {
-        ALOGW("getInput() could not find profile for device 0x%X, samplingRate %d, format %d, "
-                "channelMask 0x%X",
-                device, samplingRate, format, channelMask);
-        return 0;
-    }
-
-    if (profile->mModule->mHandle == 0) {
-        ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
-        return 0;
-    }
-
-    AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);
-
-    inputDesc->mInputSource = inputSource;
-    inputDesc->mDevice = device;
-    inputDesc->mSamplingRate = samplingRate;
-    inputDesc->mFormat = format;
-    inputDesc->mChannelMask = channelMask;
-    inputDesc->mRefCount = 0;
-
-    input = mpClientInterface->openInput(profile->mModule->mHandle,
-                                    &inputDesc->mDevice,
-                                    &inputDesc->mSamplingRate,
-                                    &inputDesc->mFormat,
-                                    &inputDesc->mChannelMask);
-
-    // only accept input with the exact requested set of parameters
-    if (input == 0 ||
-        (samplingRate != inputDesc->mSamplingRate) ||
-        (format != inputDesc->mFormat) ||
-        (channelMask != inputDesc->mChannelMask)) {
-        ALOGI("getInput() failed opening input: samplingRate %d, format %d, channelMask 0x%X",
-                samplingRate, format, channelMask);
-        if (input != 0) {
-            mpClientInterface->closeInput(input);
-        }
-        delete inputDesc;
-        return 0;
-    }
-    addInput(input, inputDesc);
-
-    return input;
-}
-
-status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input)
-{
-    ALOGV("startInput() input %d", input);
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        ALOGW("startInput() unknown input %d", input);
-        return BAD_VALUE;
-    }
-    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
-#ifdef AUDIO_POLICY_TEST
-    if (mTestInput == 0)
-#endif //AUDIO_POLICY_TEST
-    {
-        // refuse 2 active AudioRecord clients at the same time except if the active input
-        // uses AUDIO_SOURCE_HOTWORD in which case it is closed.
-        audio_io_handle_t activeInput = getActiveInput();
-        if (!isVirtualInputDevice(inputDesc->mDevice) && activeInput != 0) {
-            AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput);
-            if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
-                ALOGW("startInput() preempting already started low-priority input %d", activeInput);
-                stopInput(activeInput);
-                releaseInput(activeInput);
-            } else {
-                ALOGW("startInput() input %d failed: other input already started", input);
-                return INVALID_OPERATION;
-            }
-        }
-    }
-
-    audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
-    if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
-        inputDesc->mDevice = newDevice;
-    }
-
-    // automatically enable the remote submix output when input is started
-    if (audio_is_remote_submix_device(inputDesc->mDevice)) {
-        setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
-                AudioSystem::DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
-    }
-
-    AudioParameter param = AudioParameter();
-    param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
-
-    int aliasSource = (inputDesc->mInputSource == AUDIO_SOURCE_HOTWORD) ?
-                                        AUDIO_SOURCE_VOICE_RECOGNITION : inputDesc->mInputSource;
-
-    param.addInt(String8(AudioParameter::keyInputSource), aliasSource);
-    ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
-
-    mpClientInterface->setParameters(input, param.toString());
-
-    inputDesc->mRefCount = 1;
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input)
-{
-    ALOGV("stopInput() input %d", input);
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        ALOGW("stopInput() unknown input %d", input);
-        return BAD_VALUE;
-    }
-    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
-    if (inputDesc->mRefCount == 0) {
-        ALOGW("stopInput() input %d already stopped", input);
-        return INVALID_OPERATION;
-    } else {
-        // automatically disable the remote submix output when input is stopped
-        if (audio_is_remote_submix_device(inputDesc->mDevice)) {
-            setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
-                    AudioSystem::DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
-        }
-
-        AudioParameter param = AudioParameter();
-        param.addInt(String8(AudioParameter::keyRouting), 0);
-        mpClientInterface->setParameters(input, param.toString());
-        inputDesc->mRefCount = 0;
-        return NO_ERROR;
-    }
-}
-
-void AudioPolicyManagerBase::releaseInput(audio_io_handle_t input)
-{
-    ALOGV("releaseInput() %d", input);
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        ALOGW("releaseInput() releasing unknown input %d", input);
-        return;
-    }
-    mpClientInterface->closeInput(input);
-    delete mInputs.valueAt(index);
-    mInputs.removeItem(input);
-
-    ALOGV("releaseInput() exit");
-}
-
-void AudioPolicyManagerBase::closeAllInputs() {
-    for(size_t input_index = 0; input_index < mInputs.size(); input_index++) {
-        mpClientInterface->closeInput(mInputs.keyAt(input_index));
-    }
-    mInputs.clear();
-}
-
-void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type 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;
-}
-
-status_t AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type 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);
-
-    // compute and apply stream volume on all outputs according to connected device
-    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) || (device == curDevice)) {
-            status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
-            if (volStatus != NO_ERROR) {
-                status = volStatus;
-            }
-        }
-    }
-    return status;
-}
-
-status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type 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 AudioPolicyManagerBase::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++) {
-        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 AudioPolicyManagerBase::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(AudioSystem::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 AudioPolicyManagerBase::registerEffect(const effect_descriptor_t *desc,
-                                audio_io_handle_t io,
-                                uint32_t strategy,
-                                audio_session_t 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);
-
-    EffectDescriptor *pDesc = new EffectDescriptor();
-    memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t));
-    pDesc->mIo = io;
-    pDesc->mStrategy = (routing_strategy)strategy;
-    pDesc->mSession = session;
-    pDesc->mEnabled = false;
-
-    mEffects.add(id, pDesc);
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::unregisterEffect(int id)
-{
-    ssize_t index = mEffects.indexOfKey(id);
-    if (index < 0) {
-        ALOGW("unregisterEffect() unknown effect ID %d", id);
-        return INVALID_OPERATION;
-    }
-
-    EffectDescriptor *pDesc = mEffects.valueAt(index);
-
-    setEffectEnabled(pDesc, false);
-
-    if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) {
-        ALOGW("unregisterEffect() memory %d too big for total %d",
-                pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
-        pDesc->mDesc.memoryUsage = mTotalEffectsMemory;
-    }
-    mTotalEffectsMemory -= pDesc->mDesc.memoryUsage;
-    ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
-            pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
-
-    mEffects.removeItem(id);
-    delete pDesc;
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::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 AudioPolicyManagerBase::setEffectEnabled(EffectDescriptor *pDesc, bool enabled)
-{
-    if (enabled == pDesc->mEnabled) {
-        ALOGV("setEffectEnabled(%s) effect already %s",
-             enabled?"true":"false", enabled?"enabled":"disabled");
-        return INVALID_OPERATION;
-    }
-
-    if (enabled) {
-        if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
-            ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
-                 pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10);
-            return INVALID_OPERATION;
-        }
-        mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad;
-        ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
-    } else {
-        if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) {
-            ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
-                    pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
-            pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
-        }
-        mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad;
-        ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
-    }
-    pDesc->mEnabled = enabled;
-    return NO_ERROR;
-}
-
-bool AudioPolicyManagerBase::isNonOffloadableEffectEnabled()
-{
-    for (size_t i = 0; i < mEffects.size(); i++) {
-        const EffectDescriptor * const pDesc = mEffects.valueAt(i);
-        if (pDesc->mEnabled && (pDesc->mStrategy == STRATEGY_MEDIA) &&
-                ((pDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
-            ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
-                  pDesc->mDesc.name, pDesc->mSession);
-            return true;
-        }
-    }
-    return false;
-}
-
-bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const
-{
-    nsecs_t sysTime = systemTime();
-    for (size_t i = 0; i < mOutputs.size(); i++) {
-        const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
-        if (outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) {
-            return true;
-        }
-    }
-    return false;
-}
-
-bool AudioPolicyManagerBase::isStreamActiveRemotely(int stream, uint32_t inPastMs) const
-{
-    nsecs_t sysTime = systemTime();
-    for (size_t i = 0; i < mOutputs.size(); i++) {
-        const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
-        if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
-                outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) {
-            return true;
-        }
-    }
-    return false;
-}
-
-bool AudioPolicyManagerBase::isSourceActive(audio_source_t source) const
-{
-    for (size_t i = 0; i < mInputs.size(); i++) {
-        const AudioInputDescriptor * inputDescriptor = mInputs.valueAt(i);
-        if ((inputDescriptor->mInputSource == (int)source ||
-                (source == (audio_source_t)AUDIO_SOURCE_VOICE_RECOGNITION &&
-                 inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD))
-             && (inputDescriptor->mRefCount > 0)) {
-            return true;
-        }
-    }
-    return false;
-}
-
-
-status_t AudioPolicyManagerBase::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, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
-    result.append(buffer);
-    snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string());
-    result.append(buffer);
-    snprintf(buffer, SIZE, " USB audio ALSA %s\n", mUsbOutCardAndDevice.string());
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AudioSystem::FOR_DOCK]);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AudioSystem::FOR_SYSTEM]);
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-
-
-    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 < AudioSystem::NUM_STREAM_TYPES; 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);
-    }
-
-
-    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 AudioPolicyManagerBase::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
-    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 != NULL ? "" : "NOT ");
-    return (profile != NULL);
-}
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManagerBase
-// ----------------------------------------------------------------------------
-
-AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
-    :
-#ifdef AUDIO_POLICY_TEST
-    Thread(false),
-#endif //AUDIO_POLICY_TEST
-    mPrimaryOutput((audio_io_handle_t)0),
-    mAvailableOutputDevices(AUDIO_DEVICE_NONE),
-    mPhoneState(AudioSystem::MODE_NORMAL),
-    mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
-    mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
-    mA2dpSuspended(false), mHasA2dp(false), mHasUsb(false), mHasRemoteSubmix(false),
-    mSpeakerDrcEnabled(false)
-{
-    mpClientInterface = clientInterface;
-
-    for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
-        mForceUse[i] = AudioSystem::FORCE_NONE;
-    }
-
-    mA2dpDeviceAddress = String8("");
-    mScoDeviceAddress = String8("");
-    mUsbOutCardAndDevice = String8("");
-
-    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();
-        }
-    }
-
-    // must be done after reading the policy
-    initializeVolumeCurves();
-
-    // open all output streams needed to access attached devices
-    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.
-        for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
-        {
-            const IOProfile *outProfile = mHwModules[i]->mOutputProfiles[j];
-
-            if ((outProfile->mSupportedDevices & mAttachedOutputDevices) &&
-                    ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
-                AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile);
-                outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice &
-                                                            outProfile->mSupportedDevices);
-                audio_io_handle_t output = mpClientInterface->openOutput(
-                                                outProfile->mModule->mHandle,
-                                                &outputDesc->mDevice,
-                                                &outputDesc->mSamplingRate,
-                                                &outputDesc->mFormat,
-                                                &outputDesc->mChannelMask,
-                                                &outputDesc->mLatency,
-                                                outputDesc->mFlags);
-                if (output == 0) {
-                    delete outputDesc;
-                } else {
-                    mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices |
-                                            (outProfile->mSupportedDevices & mAttachedOutputDevices));
-                    if (mPrimaryOutput == 0 &&
-                            outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
-                        mPrimaryOutput = output;
-                    }
-                    addOutput(output, outputDesc);
-                    setOutputDevice(output,
-                                    (audio_devices_t)(mDefaultOutputDevice &
-                                                        outProfile->mSupportedDevices),
-                                    true);
-                }
-            }
-        }
-    }
-
-    ALOGE_IF((mAttachedOutputDevices & ~mAvailableOutputDevices),
-             "Not output found for attached devices %08x",
-             (mAttachedOutputDevices & ~mAvailableOutputDevices));
-
-    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 = AudioSystem::PCM_16_BIT;
-        mTestChannels =  AudioSystem::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
-}
-
-AudioPolicyManagerBase::~AudioPolicyManagerBase()
-{
-#ifdef AUDIO_POLICY_TEST
-    exit();
-#endif //AUDIO_POLICY_TEST
-   for (size_t i = 0; i < mOutputs.size(); i++) {
-        mpClientInterface->closeOutput(mOutputs.keyAt(i));
-        delete mOutputs.valueAt(i);
-   }
-   for (size_t i = 0; i < mInputs.size(); i++) {
-        mpClientInterface->closeInput(mInputs.keyAt(i));
-        delete mInputs.valueAt(i);
-   }
-   for (size_t i = 0; i < mHwModules.size(); i++) {
-        delete mHwModules[i];
-   }
-}
-
-status_t AudioPolicyManagerBase::initCheck()
-{
-    return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
-}
-
-#ifdef AUDIO_POLICY_TEST
-bool AudioPolicyManagerBase::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 = AudioSystem::INVALID_FORMAT;
-                if (value == "PCM 16 bits") {
-                    format = AudioSystem::PCM_16_BIT;
-                } else if (value == "PCM 8 bits") {
-                    format = AudioSystem::PCM_8_BIT;
-                } else if (value == "Compressed MP3") {
-                    format = AudioSystem::MP3;
-                }
-                if (format != AudioSystem::INVALID_FORMAT) {
-                    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 =  AudioSystem::CHANNEL_OUT_STEREO;
-                } else if (value == "Channels Mono") {
-                    channels =  AudioSystem::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"));
-
-                AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
-                mpClientInterface->closeOutput(mPrimaryOutput);
-
-                audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
-
-                delete mOutputs.valueFor(mPrimaryOutput);
-                mOutputs.removeItem(mPrimaryOutput);
-
-                AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
-                outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
-                mPrimaryOutput = mpClientInterface->openOutput(moduleHandle,
-                                                &outputDesc->mDevice,
-                                                &outputDesc->mSamplingRate,
-                                                &outputDesc->mFormat,
-                                                &outputDesc->mChannelMask,
-                                                &outputDesc->mLatency,
-                                                outputDesc->mFlags);
-                if (mPrimaryOutput == 0) {
-                    ALOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
-                            outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
-                } else {
-                    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 AudioPolicyManagerBase::exit()
-{
-    {
-        AutoMutex _l(mLock);
-        requestExit();
-        mWaitWorkCV.signal();
-    }
-    requestExitAndWait();
-}
-
-int AudioPolicyManagerBase::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 AudioPolicyManagerBase::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc)
-{
-    outputDesc->mId = id;
-    mOutputs.add(id, outputDesc);
-}
-
-void AudioPolicyManagerBase::addInput(audio_io_handle_t id, AudioInputDescriptor *inputDesc)
-{
-    inputDesc->mId = id;
-    mInputs.add(id, inputDesc);
-}
-
-status_t AudioPolicyManagerBase::checkOutputsForDevice(audio_devices_t device,
-                                                       AudioSystem::device_connection_state state,
-                                                       SortedVector<audio_io_handle_t>& outputs,
-                                                       const String8 paramStr)
-{
-    AudioOutputDescriptor *desc;
-
-    if (state == AudioSystem::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 & device)) {
-                ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
-                outputs.add(mOutputs.keyAt(i));
-            }
-        }
-        // then look for output profiles that can be routed to this device
-        SortedVector<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++)
-            {
-                if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices & device) {
-                    ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
-                    profiles.add(mHwModules[i]->mOutputProfiles[j]);
-                }
-            }
-        }
-
-        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++) {
-            IOProfile *profile = profiles[profile_index];
-
-            // nothing to do if one output is already opened for this profile
-            size_t j;
-            for (j = 0; j < mOutputs.size(); j++) {
-                desc = mOutputs.valueAt(j);
-                if (!desc->isDuplicated() && desc->mProfile == profile) {
-                    break;
-                }
-            }
-            if (j != mOutputs.size()) {
-                continue;
-            }
-
-            ALOGV("opening output for device %08x with params %s", device, paramStr.string());
-            desc = new AudioOutputDescriptor(profile);
-            desc->mDevice = device;
-            audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
-            offloadInfo.sample_rate = desc->mSamplingRate;
-            offloadInfo.format = desc->mFormat;
-            offloadInfo.channel_mask = desc->mChannelMask;
-
-            audio_io_handle_t output = mpClientInterface->openOutput(profile->mModule->mHandle,
-                                                                       &desc->mDevice,
-                                                                       &desc->mSamplingRate,
-                                                                       &desc->mFormat,
-                                                                       &desc->mChannelMask,
-                                                                       &desc->mLatency,
-                                                                       desc->mFlags,
-                                                                       &offloadInfo);
-            if (output != 0) {
-                if (!paramStr.isEmpty()) {
-                    // Here is where the out_set_parameters() for card & device gets called
-                    mpClientInterface->setParameters(output, paramStr);
-                }
-
-                // 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() direct output sup sampling rates %s",
-                              reply.string());
-                    value = strpbrk((char *)reply.string(), "=");
-                    if (value != NULL) {
-                        loadSamplingRates(value + 1, profile);
-                    }
-                }
-                if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
-                    reply = mpClientInterface->getParameters(output,
-                                                   String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
-                    ALOGV("checkOutputsForDevice() direct output sup formats %s",
-                              reply.string());
-                    value = strpbrk((char *)reply.string(), "=");
-                    if (value != NULL) {
-                        loadFormats(value + 1, profile);
-                    }
-                }
-                if (profile->mChannelMasks[0] == 0) {
-                    reply = mpClientInterface->getParameters(output,
-                                                  String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
-                    ALOGV("checkOutputsForDevice() direct output sup channel masks %s",
-                              reply.string());
-                    value = strpbrk((char *)reply.string(), "=");
-                    if (value != NULL) {
-                        loadOutChannels(value + 1, profile);
-                    }
-                }
-                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("checkOutputsForDevice() direct output missing param");
-                    mpClientInterface->closeOutput(output);
-                    output = 0;
-                } else if (profile->mSamplingRates[0] == 0) {
-                    mpClientInterface->closeOutput(output);
-                    desc->mSamplingRate = profile->mSamplingRates[1];
-                    offloadInfo.sample_rate = desc->mSamplingRate;
-                    output = mpClientInterface->openOutput(
-                                                    profile->mModule->mHandle,
-                                                    &desc->mDevice,
-                                                    &desc->mSamplingRate,
-                                                    &desc->mFormat,
-                                                    &desc->mChannelMask,
-                                                    &desc->mLatency,
-                                                    desc->mFlags,
-                                                    &offloadInfo);
-                }
-
-                if (output != 0) {
-                    addOutput(output, desc);
-                    if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) {
-                        audio_io_handle_t duplicatedOutput = 0;
-
-                        // 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 != 0) {
-                            // add duplicated output descriptor
-                            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);
-                            output = 0;
-                        }
-                    }
-                }
-            }
-            if (output == 0) {
-                ALOGW("checkOutputsForDevice() could not open output for device %x", device);
-                delete desc;
-                profiles.removeAt(profile_index);
-                profile_index--;
-            } else {
-                outputs.add(output);
-                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() &&
-                    !(desc->mProfile->mSupportedDevices & mAvailableOutputDevices)) {
-                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++)
-            {
-                IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
-                if (profile->mSupportedDevices & 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 AudioPolicyManagerBase::checkInputsForDevice(audio_devices_t device,
-                                                      AudioSystem::device_connection_state state,
-                                                      SortedVector<audio_io_handle_t>& inputs,
-                                                      const String8 paramStr)
-{
-    AudioInputDescriptor *desc;
-    if (state == AudioSystem::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 & (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<IOProfile *> profiles;
-        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++)
-            {
-                if (mHwModules[module_index]->mInputProfiles[profile_index]->mSupportedDevices
-                        & (device & ~AUDIO_DEVICE_BIT_IN)) {
-                    ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
-                          profile_index, module_index);
-                    profiles.add(mHwModules[module_index]->mInputProfiles[profile_index]);
-                }
-            }
-        }
-
-        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++) {
-
-            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, paramStr.string());
-            desc = new AudioInputDescriptor(profile);
-            desc->mDevice = device;
-
-            audio_io_handle_t input = mpClientInterface->openInput(profile->mModule->mHandle,
-                                            &desc->mDevice,
-                                            &desc->mSamplingRate,
-                                            &desc->mFormat,
-                                            &desc->mChannelMask);
-
-            if (input != 0) {
-                if (!paramStr.isEmpty()) {
-                    mpClientInterface->setParameters(input, paramStr);
-                }
-
-                // 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) {
-                        loadSamplingRates(value + 1, profile);
-                    }
-                }
-                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) {
-                        loadFormats(value + 1, profile);
-                    }
-                }
-                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) {
-                        loadInChannels(value + 1, profile);
-                    }
-                }
-                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 = 0;
-                }
-
-                if (input != 0) {
-                    addInput(input, desc);
-                }
-            } // endif input != 0
-
-            if (input == 0) {
-                ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
-                delete desc;
-                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 & mAvailableInputDevices)) {
-                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++)
-            {
-                IOProfile *profile = mHwModules[module_index]->mInputProfiles[profile_index];
-                if (profile->mSupportedDevices & device) {
-                    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 AudioPolicyManagerBase::closeOutput(audio_io_handle_t output)
-{
-    ALOGV("closeOutput(%d)", output);
-
-    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
-    if (outputDesc == NULL) {
-        ALOGW("closeOutput() unknown output %d", output);
-        return;
-    }
-
-    // look for duplicated outputs connected to the output being removed.
-    for (size_t i = 0; i < mOutputs.size(); i++) {
-        AudioOutputDescriptor *dupOutputDesc = mOutputs.valueAt(i);
-        if (dupOutputDesc->isDuplicated() &&
-                (dupOutputDesc->mOutput1 == outputDesc ||
-                dupOutputDesc->mOutput2 == outputDesc)) {
-            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 < (int)AudioSystem::NUM_STREAM_TYPES; j++) {
-                int refCount = dupOutputDesc->mRefCount[j];
-                outputDesc2->changeRefCount((AudioSystem::stream_type)j,-refCount);
-            }
-            audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
-            ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
-
-            mpClientInterface->closeOutput(duplicatedOutput);
-            delete mOutputs.valueFor(duplicatedOutput);
-            mOutputs.removeItem(duplicatedOutput);
-        }
-    }
-
-    AudioParameter param;
-    param.add(String8("closing"), String8("true"));
-    mpClientInterface->setParameters(output, param.toString());
-
-    mpClientInterface->closeOutput(output);
-    delete outputDesc;
-    mOutputs.removeItem(output);
-    mPreviousOutputs = mOutputs;
-}
-
-SortedVector<audio_io_handle_t> AudioPolicyManagerBase::getOutputsForDevice(audio_devices_t device,
-                        DefaultKeyedVector<audio_io_handle_t, 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 AudioPolicyManagerBase::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 AudioPolicyManagerBase::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);
-
-    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++) {
-            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++) {
-                EffectDescriptor *desc = mEffects.valueAt(i);
-                if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
-                        desc->mIo != fxOutput) {
-                    if (moved.indexOf(desc->mIo) < 0) {
-                        ALOGV("checkOutputForStrategy() moving effect %d to output %d",
-                              mEffects.keyAt(i), fxOutput);
-                        mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, desc->mIo,
-                                                       fxOutput);
-                        moved.add(desc->mIo);
-                    }
-                    desc->mIo = fxOutput;
-                }
-            }
-        }
-        // Move tracks associated to this strategy from previous output to new output
-        for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
-            if (getStrategy((AudioSystem::stream_type)i) == strategy) {
-                mpClientInterface->invalidateStream((AudioSystem::stream_type)i);
-            }
-        }
-    }
-}
-
-void AudioPolicyManagerBase::checkOutputForAllStrategies()
-{
-    checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
-    checkOutputForStrategy(STRATEGY_PHONE);
-    checkOutputForStrategy(STRATEGY_SONIFICATION);
-    checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
-    checkOutputForStrategy(STRATEGY_MEDIA);
-    checkOutputForStrategy(STRATEGY_DTMF);
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getA2dpOutput()
-{
-    if (!mHasA2dp) {
-        return 0;
-    }
-
-    for (size_t i = 0; i < mOutputs.size(); i++) {
-        AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
-        if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
-            return mOutputs.keyAt(i);
-        }
-    }
-
-    return 0;
-}
-
-void AudioPolicyManagerBase::checkA2dpSuspend()
-{
-    if (!mHasA2dp) {
-        return;
-    }
-    audio_io_handle_t a2dpOutput = getA2dpOutput();
-    if (a2dpOutput == 0) {
-        return;
-    }
-
-    // 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 (((mScoDeviceAddress == "") ||
-             ((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) &&
-              (mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) &&
-             ((mPhoneState != AudioSystem::MODE_IN_CALL) &&
-              (mPhoneState != AudioSystem::MODE_RINGTONE))) {
-
-            mpClientInterface->restoreOutput(a2dpOutput);
-            mA2dpSuspended = false;
-        }
-    } else {
-        if (((mScoDeviceAddress != "") &&
-             ((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
-              (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) ||
-             ((mPhoneState == AudioSystem::MODE_IN_CALL) ||
-              (mPhoneState == AudioSystem::MODE_RINGTONE))) {
-
-            mpClientInterface->suspendOutput(a2dpOutput);
-            mA2dpSuspended = true;
-        }
-    }
-}
-
-audio_devices_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache)
-{
-    audio_devices_t device = AUDIO_DEVICE_NONE;
-
-    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
-    // check the following by order of priority to request a routing change if necessary:
-    // 1: the strategy enforced audible is active 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 sonification is active on the output:
-    //      use device for strategy sonification
-    // 4: the strategy "respectful" sonification is active on the output:
-    //      use device for strategy "respectful" sonification
-    // 5: the strategy media is active on the output:
-    //      use device for strategy media
-    // 6: the strategy DTMF is active on the output:
-    //      use device for strategy DTMF
-    if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
-        device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
-    } else if (isInCall() ||
-                    outputDesc->isStrategyActive(STRATEGY_PHONE)) {
-        device = getDeviceForStrategy(STRATEGY_PHONE, 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_MEDIA)) {
-        device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
-    } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
-        device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
-    }
-
-    ALOGV("getNewDevice() selected device %x", device);
-    return device;
-}
-
-uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) {
-    return (uint32_t)getStrategy(stream);
-}
-
-audio_devices_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) {
-    audio_devices_t devices;
-    // 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 < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
-        devices = AUDIO_DEVICE_NONE;
-    } else {
-        AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream);
-        devices = getDeviceForStrategy(strategy, true /*fromCache*/);
-    }
-    return devices;
-}
-
-AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy(
-        AudioSystem::stream_type stream) {
-    // stream to strategy mapping
-    switch (stream) {
-    case AudioSystem::VOICE_CALL:
-    case AudioSystem::BLUETOOTH_SCO:
-        return STRATEGY_PHONE;
-    case AudioSystem::RING:
-    case AudioSystem::ALARM:
-        return STRATEGY_SONIFICATION;
-    case AudioSystem::NOTIFICATION:
-        return STRATEGY_SONIFICATION_RESPECTFUL;
-    case AudioSystem::DTMF:
-        return STRATEGY_DTMF;
-    default:
-        ALOGE("unknown stream type");
-    case AudioSystem::SYSTEM:
-        // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
-        // while key clicks are played produces a poor result
-    case AudioSystem::TTS:
-    case AudioSystem::MUSIC:
-        return STRATEGY_MEDIA;
-    case AudioSystem::ENFORCED_AUDIBLE:
-        return STRATEGY_ENFORCED_AUDIBLE;
-    }
-}
-
-void AudioPolicyManagerBase::handleNotificationRoutingForStream(AudioSystem::stream_type stream) {
-    switch(stream) {
-    case AudioSystem::MUSIC:
-        checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
-        updateDevicesAndOutputs();
-        break;
-    default:
-        break;
-    }
-}
-
-audio_devices_t AudioPolicyManagerBase::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];
-    }
-
-    switch (strategy) {
-
-    case STRATEGY_SONIFICATION_RESPECTFUL:
-        if (isInCall()) {
-            device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
-        } else if (isStreamActiveRemotely(AudioSystem::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*/);
-        } else if (isStreamActive(AudioSystem::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*/);
-        }
-
-        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:
-        // for phone strategy, we first consider the forced use and then the available devices by order
-        // of priority
-        switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
-        case AudioSystem::FORCE_BT_SCO:
-            if (!isInCall() || strategy != STRATEGY_DTMF) {
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
-                if (device) break;
-            }
-            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
-            if (device) break;
-            device = mAvailableOutputDevices & 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 (mHasA2dp && !isInCall() &&
-                    (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
-                    (getA2dpOutput() != 0) && !mA2dpSuspended) {
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
-                if (device) break;
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
-                if (device) break;
-            }
-            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
-            if (device) break;
-            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
-            if (device) break;
-            if (mPhoneState != AudioSystem::MODE_IN_CALL) {
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
-                if (device) break;
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
-                if (device) break;
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
-                if (device) break;
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
-                if (device) break;
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
-                if (device) break;
-            }
-            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE;
-            if (device) break;
-            device = mDefaultOutputDevice;
-            if (device == AUDIO_DEVICE_NONE) {
-                ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
-            }
-            break;
-
-        case AudioSystem::FORCE_SPEAKER:
-            // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
-            // A2DP speaker when forcing to speaker output
-            if (mHasA2dp && !isInCall() &&
-                    (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
-                    (getA2dpOutput() != 0) && !mA2dpSuspended) {
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
-                if (device) break;
-            }
-            if (mPhoneState != AudioSystem::MODE_IN_CALL) {
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
-                if (device) break;
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
-                if (device) break;
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
-                if (device) break;
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
-                if (device) break;
-                device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
-                if (device) break;
-            }
-            device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
-            if (device) break;
-            device = mDefaultOutputDevice;
-            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[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_SYSTEM_ENFORCED)) {
-            device = mAvailableOutputDevices & 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
-
-    case STRATEGY_MEDIA: {
-        uint32_t device2 = AUDIO_DEVICE_NONE;
-        if (strategy != STRATEGY_SONIFICATION) {
-            // no sonification on remote submix (e.g. WFD)
-            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
-        }
-        if ((device2 == AUDIO_DEVICE_NONE) &&
-                mHasA2dp && (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
-                (getA2dpOutput() != 0) && !mA2dpSuspended) {
-            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
-            if (device2 == AUDIO_DEVICE_NONE) {
-                device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
-            }
-            if (device2 == AUDIO_DEVICE_NONE) {
-                device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
-            }
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
-        }
-        if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
-            // no sonification on aux digital (e.g. HDMI)
-            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
-        }
-        if ((device2 == AUDIO_DEVICE_NONE) &&
-                (mForceUse[AudioSystem::FOR_DOCK] == AudioSystem::FORCE_ANALOG_DOCK)) {
-            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
-        }
-
-        // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
-        // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
-        device |= device2;
-        if (device) break;
-        device = mDefaultOutputDevice;
-        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 AudioPolicyManagerBase::updateDevicesAndOutputs()
-{
-    for (int i = 0; i < NUM_STRATEGIES; i++) {
-        mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
-    }
-    mPreviousOutputs = mOutputs;
-}
-
-uint32_t AudioPolicyManagerBase::checkDeviceMuteStrategies(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() && (AudioSystem::popCount(device) >= 2);
-    // temporary mute output if device selection changes to avoid volume bursts due to
-    // different per device volumes
-    bool tempMute = outputDesc->isActive() && (device != prevDevice);
-
-    for (size_t i = 0; i < NUM_STRATEGIES; i++) {
-        audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
-        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 || tempMute) {
-            for (size_t j = 0; j < mOutputs.size(); j++) {
-                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)) {
-                    // do tempMute only for current output
-                    if (tempMute && (desc == outputDesc)) {
-                        setStrategyMute((routing_strategy)i, true, curOutput);
-                        setStrategyMute((routing_strategy)i, false, curOutput,
-                                            desc->latency() * 2, device);
-                    }
-                    if ((tempMute && (desc == outputDesc)) || mute) {
-                        if (muteWaitMs < desc->latency()) {
-                            muteWaitMs = desc->latency();
-                        }
-                    }
-                }
-            }
-        }
-    }
-
-    // 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).
-    muteWaitMs *= 2;
-    // 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 AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output,
-                                             audio_devices_t device,
-                                             bool force,
-                                             int delayMs)
-{
-    ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
-    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
-    AudioParameter param;
-    uint32_t muteWaitMs;
-
-    if (outputDesc->isDuplicated()) {
-        muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
-        muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, 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) == 0)) {
-        return 0;
-    }
-
-    // filter devices according to output selected
-    device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices);
-
-    audio_devices_t prevDevice = outputDesc->mDevice;
-
-    ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
-
-    if (device != AUDIO_DEVICE_NONE) {
-        outputDesc->mDevice = device;
-
-        // Force routing if previously asked for this output
-        if (outputDesc->mForceRouting) {
-            ALOGV("Force routing to current device as previous device was null for this output");
-            force = true;
-
-            // Request consumed. Reset mForceRouting to false
-            outputDesc->mForceRouting = false;
-        }
-    }
-    else {
-        // Device is null and does not reflect the routing. Save the necessity to force
-        // re-routing upon next attempt to select a non-null device for this output
-        outputDesc->mForceRouting = true;
-    }
-
-    muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
-
-    // Do not change the routing if:
-    //  - the requested device is AUDIO_DEVICE_NONE
-    //  - the requested device is the same as current device and force is not specified.
-    // Doing this check here allows the caller to call setOutputDevice() without conditions
-    if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force) {
-        ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output);
-        return muteWaitMs;
-    }
-
-    ALOGV("setOutputDevice() changing device");
-    // do the routing
-    param.addInt(String8(AudioParameter::keyRouting), (int)device);
-    mpClientInterface->setParameters(output, param.toString(), delayMs);
-
-    // update stream volumes according to new device
-    applyStreamVolumes(output, device, delayMs);
-
-    return muteWaitMs;
-}
-
-AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getInputProfile(audio_devices_t device,
-                                                   uint32_t samplingRate,
-                                                   audio_format_t format,
-                                                   audio_channel_mask_t channelMask)
-{
-    // 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++)
-        {
-            IOProfile *profile = mHwModules[i]->mInputProfiles[j];
-            // profile->log();
-            if (profile->isCompatibleProfile(device, samplingRate, format,
-                                             channelMask, AUDIO_OUTPUT_FLAG_NONE)) {
-                return profile;
-            }
-        }
-    }
-    return NULL;
-}
-
-audio_devices_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
-{
-    uint32_t device = AUDIO_DEVICE_NONE;
-
-    switch (inputSource) {
-    case AUDIO_SOURCE_VOICE_UPLINK:
-      if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
-          device = AUDIO_DEVICE_IN_VOICE_CALL;
-          break;
-      }
-      // FALL THROUGH
-
-    case AUDIO_SOURCE_DEFAULT:
-    case AUDIO_SOURCE_MIC:
-    if (mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
-        device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
-        break;
-    }
-    // FALL THROUGH
-
-    case AUDIO_SOURCE_VOICE_RECOGNITION:
-    case AUDIO_SOURCE_HOTWORD:
-    case AUDIO_SOURCE_VOICE_COMMUNICATION:
-        if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
-            mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
-            device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
-        } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) {
-            device = AUDIO_DEVICE_IN_WIRED_HEADSET;
-        } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_USB_DEVICE) {
-            device = AUDIO_DEVICE_IN_USB_DEVICE;
-        } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-        }
-        break;
-    case AUDIO_SOURCE_CAMCORDER:
-        if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) {
-            device = AUDIO_DEVICE_IN_BACK_MIC;
-        } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-        }
-        break;
-    case AUDIO_SOURCE_VOICE_DOWNLINK:
-    case AUDIO_SOURCE_VOICE_CALL:
-        if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
-            device = AUDIO_DEVICE_IN_VOICE_CALL;
-        }
-        break;
-    case AUDIO_SOURCE_REMOTE_SUBMIX:
-        if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
-            device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
-        }
-        break;
-    default:
-        ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
-        break;
-    }
-    ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
-    return device;
-}
-
-bool AudioPolicyManagerBase::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;
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getActiveInput(bool ignoreVirtualInputs)
-{
-    for (size_t i = 0; i < mInputs.size(); i++) {
-        const AudioInputDescriptor * input_descriptor = mInputs.valueAt(i);
-        if ((input_descriptor->mRefCount > 0)
-                && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
-            return mInputs.keyAt(i);
-        }
-    }
-    return 0;
-}
-
-
-audio_devices_t AudioPolicyManagerBase::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 (AudioSystem::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.
-        if (device & AUDIO_DEVICE_OUT_SPEAKER) {
-            device = AUDIO_DEVICE_OUT_SPEAKER;
-        } else {
-            device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
-        }
-    }
-
-    ALOGW_IF(AudioSystem::popCount(device) != 1,
-            "getDeviceForVolume() invalid device combination: %08x",
-            device);
-
-    return device;
-}
-
-AudioPolicyManagerBase::device_category AudioPolicyManagerBase::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_SPEAKER:
-        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
-        case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
-        case AUDIO_DEVICE_OUT_AUX_DIGITAL:
-        case AUDIO_DEVICE_OUT_USB_ACCESSORY:
-        case AUDIO_DEVICE_OUT_USB_DEVICE:
-        case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
-        default:
-            return DEVICE_CATEGORY_SPEAKER;
-    }
-}
-
-float AudioPolicyManagerBase::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 AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManagerBase::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 AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sDefaultSystemVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT] = {
-    {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-    AudioPolicyManagerBase::sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
-    {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManagerBase::VolumeCurvePoint
-            *AudioPolicyManagerBase::sVolumeProfiles[AudioSystem::NUM_STREAM_TYPES]
-                                                   [AudioPolicyManagerBase::DEVICE_CATEGORY_CNT] = {
-    { // AUDIO_STREAM_VOICE_CALL
-        sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVoiceVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    { // AUDIO_STREAM_SYSTEM
-        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultSystemVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    { // AUDIO_STREAM_RING
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    { // AUDIO_STREAM_MUSIC
-        sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    { // AUDIO_STREAM_ALARM
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    { // AUDIO_STREAM_NOTIFICATION
-        sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    { // AUDIO_STREAM_BLUETOOTH_SCO
-        sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultVoiceVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    { // AUDIO_STREAM_ENFORCED_AUDIBLE
-        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultSystemVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    {  // AUDIO_STREAM_DTMF
-        sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultSystemVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-    { // AUDIO_STREAM_TTS
-        sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
-        sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
-        sDefaultMediaVolumeCurve  // DEVICE_CATEGORY_EARPIECE
-    },
-};
-
-void AudioPolicyManagerBase::initializeVolumeCurves()
-{
-    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; 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;
-    }
-}
-
-float AudioPolicyManagerBase::computeVolume(int stream,
-                                            int index,
-                                            audio_io_handle_t output,
-                                            audio_devices_t device)
-{
-    float volume = 1.0;
-    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
-    StreamDescriptor &streamDesc = mStreams[stream];
-
-    if (device == AUDIO_DEVICE_NONE) {
-        device = outputDesc->device();
-    }
-
-    // if volume is not 0 (not muted), force media volume to max on digital output
-    if (stream == AudioSystem::MUSIC &&
-        index != mStreams[stream].mIndexMin &&
-        (device == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
-         device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) {
-        return 1.0;
-    }
-
-    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((AudioSystem::stream_type)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 == AudioSystem::SYSTEM)
-                || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
-                    (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::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(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
-                mLimitRingtoneVolume) {
-            audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
-            float musicVol = computeVolume(AudioSystem::MUSIC,
-                               mStreams[AudioSystem::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 AudioPolicyManagerBase::checkAndSetVolume(int 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 == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
-        (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
-        ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
-             stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
-        return INVALID_OPERATION;
-    }
-
-    float volume = computeVolume(stream, index, output, device);
-    // 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 == AudioSystem::BLUETOOTH_SCO) {
-            mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs);
-        }
-        mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
-    }
-
-    if (stream == AudioSystem::VOICE_CALL ||
-        stream == AudioSystem::BLUETOOTH_SCO) {
-        float voiceVolume;
-        // Force voice volume to max for bluetooth SCO as volume is managed by the headset
-        if (stream == AudioSystem::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 AudioPolicyManagerBase::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 < AudioSystem::NUM_STREAM_TYPES; stream++) {
-        checkAndSetVolume(stream,
-                          mStreams[stream].getVolumeIndex(device),
-                          output,
-                          device,
-                          delayMs,
-                          force);
-    }
-}
-
-void AudioPolicyManagerBase::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 < AudioSystem::NUM_STREAM_TYPES; stream++) {
-        if (getStrategy((AudioSystem::stream_type)stream) == strategy) {
-            setStreamMute(stream, on, output, delayMs, device);
-        }
-    }
-}
-
-void AudioPolicyManagerBase::setStreamMute(int stream,
-                                           bool on,
-                                           audio_io_handle_t output,
-                                           int delayMs,
-                                           audio_devices_t device)
-{
-    StreamDescriptor &streamDesc = mStreams[stream];
-    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 != AudioSystem::ENFORCED_AUDIBLE) ||
-                     (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::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 AudioPolicyManagerBase::handleIncallSonification(int 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((AudioSystem::stream_type)stream);
-    if ((stream_strategy == STRATEGY_SONIFICATION) ||
-            ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
-        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 (AudioSystem::isLowVisibility((AudioSystem::stream_type)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(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
-                } else {
-                    mpClientInterface->stopTone();
-                }
-            }
-        }
-    }
-}
-
-bool AudioPolicyManagerBase::isInCall()
-{
-    return isStateInCall(mPhoneState);
-}
-
-bool AudioPolicyManagerBase::isStateInCall(int state) {
-    return ((state == AudioSystem::MODE_IN_CALL) ||
-            (state == AudioSystem::MODE_IN_COMMUNICATION));
-}
-
-uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad()
-{
-    return MAX_EFFECTS_CPU_LOAD;
-}
-
-uint32_t AudioPolicyManagerBase::getMaxEffectsMemory()
-{
-    return MAX_EFFECTS_MEMORY;
-}
-
-// --- AudioOutputDescriptor class implementation
-
-AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor(
-        const IOProfile *profile)
-    : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT),
-      mChannelMask(0), mLatency(0),
-    mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE),
-    mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0),
-    mForceRouting(false)
-{
-    // clear usage count for all stream types
-    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; 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) {
-        mSamplingRate = profile->mSamplingRates[0];
-        mFormat = profile->mFormats[0];
-        mChannelMask = profile->mChannelMasks[0];
-        mFlags = profile->mFlags;
-    }
-}
-
-audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::device() const
-{
-    if (isDuplicated()) {
-        return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
-    } else {
-        return mDevice;
-    }
-}
-
-uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::latency()
-{
-    if (isDuplicated()) {
-        return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
-    } else {
-        return mLatency;
-    }
-}
-
-bool AudioPolicyManagerBase::AudioOutputDescriptor::sharesHwModuleWith(
-        const 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 AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type 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 AudioPolicyManagerBase::AudioOutputDescriptor::supportedDevices()
-{
-    if (isDuplicated()) {
-        return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
-    } else {
-        return mProfile->mSupportedDevices ;
-    }
-}
-
-bool AudioPolicyManagerBase::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
-{
-    return isStrategyActive(NUM_STRATEGIES, inPastMs);
-}
-
-bool AudioPolicyManagerBase::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
-                                                                       uint32_t inPastMs,
-                                                                       nsecs_t sysTime) const
-{
-    if ((sysTime == 0) && (inPastMs != 0)) {
-        sysTime = systemTime();
-    }
-    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
-        if (((getStrategy((AudioSystem::stream_type)i) == strategy) ||
-                (NUM_STRATEGIES == strategy)) &&
-                isStreamActive((AudioSystem::stream_type)i, inPastMs, sysTime)) {
-            return true;
-        }
-    }
-    return false;
-}
-
-bool AudioPolicyManagerBase::AudioOutputDescriptor::isStreamActive(AudioSystem::stream_type 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;
-}
-
-
-status_t AudioPolicyManagerBase::AudioOutputDescriptor::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    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 < AudioSystem::NUM_STREAM_TYPES; 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
-
-AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor(const IOProfile *profile)
-    :  mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0),
-      mDevice(AUDIO_DEVICE_NONE), mRefCount(0),
-      mInputSource(0), mProfile(profile)
-{
-    if (profile != NULL) {
-         mSamplingRate = profile->mSamplingRates[0];
-         mFormat = profile->mFormats[0];
-         mChannelMask = profile->mChannelMasks[0];
-     }
-}
-
-status_t AudioPolicyManagerBase::AudioInputDescriptor::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    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);
-    write(fd, result.string(), result.size());
-
-    return NO_ERROR;
-}
-
-// --- StreamDescriptor class implementation
-
-AudioPolicyManagerBase::StreamDescriptor::StreamDescriptor()
-    :   mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
-{
-    mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
-}
-
-int AudioPolicyManagerBase::StreamDescriptor::getVolumeIndex(audio_devices_t device)
-{
-    device = AudioPolicyManagerBase::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 AudioPolicyManagerBase::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 AudioPolicyManagerBase::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;
-}
-
-// --- IOProfile class implementation
-
-AudioPolicyManagerBase::HwModule::HwModule(const char *name)
-    : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(AUDIO_MODULE_HANDLE_NONE)
-{
-}
-
-AudioPolicyManagerBase::HwModule::~HwModule()
-{
-    for (size_t i = 0; i < mOutputProfiles.size(); i++) {
-         delete mOutputProfiles[i];
-    }
-    for (size_t i = 0; i < mInputProfiles.size(); i++) {
-         delete mInputProfiles[i];
-    }
-    free((void *)mName);
-}
-
-void AudioPolicyManagerBase::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);
-    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);
-        }
-    }
-}
-
-AudioPolicyManagerBase::IOProfile::IOProfile(HwModule *module)
-    : mFlags((audio_output_flags_t)0), mModule(module)
-{
-}
-
-AudioPolicyManagerBase::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 AudioPolicyManagerBase::IOProfile::isCompatibleProfile(audio_devices_t device,
-                                                            uint32_t samplingRate,
-                                                            audio_format_t format,
-                                                            audio_channel_mask_t channelMask,
-                                                            audio_output_flags_t flags) const
-{
-    if (samplingRate == 0 || !audio_is_valid_format(format) || channelMask == 0) {
-         return false;
-     }
-
-     if ((mSupportedDevices & device) != device) {
-         return false;
-     }
-     if ((mFlags & flags) != flags) {
-         return false;
-     }
-     size_t i;
-     for (i = 0; i < mSamplingRates.size(); i++)
-     {
-         if (mSamplingRates[i] == samplingRate) {
-             break;
-         }
-     }
-     if (i == mSamplingRates.size()) {
-         return false;
-     }
-     for (i = 0; i < mFormats.size(); i++)
-     {
-         if (mFormats[i] == format) {
-             break;
-         }
-     }
-     if (i == mFormats.size()) {
-         return false;
-     }
-     for (i = 0; i < mChannelMasks.size(); i++)
-     {
-         if (mChannelMasks[i] == channelMask) {
-             break;
-         }
-     }
-     if (i == mChannelMasks.size()) {
-         return false;
-     }
-     return true;
-}
-
-void AudioPolicyManagerBase::IOProfile::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "    - sampling rates: ");
-    result.append(buffer);
-    for (size_t i = 0; i < mSamplingRates.size(); i++) {
-        snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
-        result.append(buffer);
-        result.append(i == (mSamplingRates.size() - 1) ? "\n" : ", ");
-    }
-
-    snprintf(buffer, SIZE, "    - channel masks: ");
-    result.append(buffer);
-    for (size_t i = 0; i < mChannelMasks.size(); i++) {
-        snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
-        result.append(buffer);
-        result.append(i == (mChannelMasks.size() - 1) ? "\n" : ", ");
-    }
-
-    snprintf(buffer, SIZE, "    - formats: ");
-    result.append(buffer);
-    for (size_t i = 0; i < mFormats.size(); i++) {
-        snprintf(buffer, SIZE, "0x%08x", mFormats[i]);
-        result.append(buffer);
-        result.append(i == (mFormats.size() - 1) ? "\n" : ", ");
-    }
-
-    snprintf(buffer, SIZE, "    - devices: 0x%04x\n", mSupportedDevices);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "    - flags: 0x%04x\n", mFlags);
-    result.append(buffer);
-
-    write(fd, result.string(), result.size());
-}
-
-void AudioPolicyManagerBase::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);
-    ALOGV("    - flags: 0x%04x\n", mFlags);
-}
-
-// --- 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 struct StringToEnum sDeviceNameToEnumTable[] = {
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
-    STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
-    STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
-    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_BLUETOOTH_A2DP),
-};
-
-const struct StringToEnum sFlagNameToEnumTable[] = {
-    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),
-};
-
-const struct 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_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 struct StringToEnum sOutChannelsNameToEnumTable[] = {
-    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
-    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
-    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
-    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
-};
-
-const struct StringToEnum sInChannelsNameToEnumTable[] = {
-    STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
-    STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
-    STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
-};
-
-
-uint32_t AudioPolicyManagerBase::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;
-}
-
-bool AudioPolicyManagerBase::stringToBool(const char *value)
-{
-    return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
-}
-
-audio_output_flags_t AudioPolicyManagerBase::parseFlagNames(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(sFlagNameToEnumTable,
-                               ARRAY_SIZE(sFlagNameToEnumTable),
-                               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 (audio_output_flags_t)flag;
-}
-
-audio_devices_t AudioPolicyManagerBase::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 AudioPolicyManagerBase::loadSamplingRates(char *name, IOProfile *profile)
-{
-    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) {
-        profile->mSamplingRates.add(0);
-        return;
-    }
-
-    while (str != NULL) {
-        uint32_t rate = atoi(str);
-        if (rate != 0) {
-            ALOGV("loadSamplingRates() adding rate %d", rate);
-            profile->mSamplingRates.add(rate);
-        }
-        str = strtok(NULL, "|");
-    }
-    return;
-}
-
-void AudioPolicyManagerBase::loadFormats(char *name, IOProfile *profile)
-{
-    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) {
-        profile->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) {
-            profile->mFormats.add(format);
-        }
-        str = strtok(NULL, "|");
-    }
-    return;
-}
-
-void AudioPolicyManagerBase::loadInChannels(char *name, IOProfile *profile)
-{
-    const char *str = strtok(name, "|");
-
-    ALOGV("loadInChannels() %s", name);
-
-    if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
-        profile->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);
-            profile->mChannelMasks.add(channelMask);
-        }
-        str = strtok(NULL, "|");
-    }
-    return;
-}
-
-void AudioPolicyManagerBase::loadOutChannels(char *name, IOProfile *profile)
-{
-    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) {
-        profile->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) {
-            profile->mChannelMasks.add(channelMask);
-        }
-        str = strtok(NULL, "|");
-    }
-    return;
-}
-
-status_t AudioPolicyManagerBase::loadInput(cnode *root, HwModule *module)
-{
-    cnode *node = root->first_child;
-
-    IOProfile *profile = new IOProfile(module);
-
-    while (node) {
-        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
-            loadSamplingRates((char *)node->value, profile);
-        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
-            loadFormats((char *)node->value, profile);
-        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
-            loadInChannels((char *)node->value, profile);
-        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
-            profile->mSupportedDevices = parseDeviceNames((char *)node->value);
-        }
-        node = node->next;
-    }
-    ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
-            "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 != AUDIO_DEVICE_NONE) &&
-            (profile->mChannelMasks.size() != 0) &&
-            (profile->mSamplingRates.size() != 0) &&
-            (profile->mFormats.size() != 0)) {
-
-        ALOGV("loadInput() adding input mSupportedDevices 0x%X", profile->mSupportedDevices);
-
-        module->mInputProfiles.add(profile);
-        return NO_ERROR;
-    } else {
-        delete profile;
-        return BAD_VALUE;
-    }
-}
-
-status_t AudioPolicyManagerBase::loadOutput(cnode *root, HwModule *module)
-{
-    cnode *node = root->first_child;
-
-    IOProfile *profile = new IOProfile(module);
-
-    while (node) {
-        if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
-            loadSamplingRates((char *)node->value, profile);
-        } else if (strcmp(node->name, FORMATS_TAG) == 0) {
-            loadFormats((char *)node->value, profile);
-        } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
-            loadOutChannels((char *)node->value, profile);
-        } else if (strcmp(node->name, DEVICES_TAG) == 0) {
-            profile->mSupportedDevices = parseDeviceNames((char *)node->value);
-        } else if (strcmp(node->name, FLAGS_TAG) == 0) {
-            profile->mFlags = parseFlagNames((char *)node->value);
-        }
-        node = node->next;
-    }
-    ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
-            "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 != AUDIO_DEVICE_NONE) &&
-            (profile->mChannelMasks.size() != 0) &&
-            (profile->mSamplingRates.size() != 0) &&
-            (profile->mFormats.size() != 0)) {
-
-        ALOGV("loadOutput() adding output mSupportedDevices %04x, mFlags %04x",
-              profile->mSupportedDevices, profile->mFlags);
-
-        module->mOutputProfiles.add(profile);
-        return NO_ERROR;
-    } else {
-        delete profile;
-        return BAD_VALUE;
-    }
-}
-
-void AudioPolicyManagerBase::loadHwModule(cnode *root)
-{
-    cnode *node = config_find(root, OUTPUTS_TAG);
-    status_t status = NAME_NOT_FOUND;
-
-    HwModule *module = new HwModule(root->name);
-
-    if (node != NULL) {
-        if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) {
-            mHasA2dp = true;
-        } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) {
-            mHasUsb = true;
-        } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) {
-            mHasRemoteSubmix = true;
-        }
-
-        node = node->first_child;
-        while (node) {
-            ALOGV("loadHwModule() loading output %s", node->name);
-            status_t tmpStatus = loadOutput(node, module);
-            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 = loadInput(node, module);
-            if (status == NAME_NOT_FOUND || status == NO_ERROR) {
-                status = tmpStatus;
-            }
-            node = node->next;
-        }
-    }
-    if (status == NO_ERROR) {
-        mHwModules.add(module);
-    } else {
-        delete module;
-    }
-}
-
-void AudioPolicyManagerBase::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 AudioPolicyManagerBase::loadGlobalConfig(cnode *root)
-{
-    cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
-    if (node == NULL) {
-        return;
-    }
-    node = node->first_child;
-    while (node) {
-        if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
-            mAttachedOutputDevices = parseDeviceNames((char *)node->value);
-            ALOGW_IF(mAttachedOutputDevices == AUDIO_DEVICE_NONE,
-                    "loadGlobalConfig() no attached output devices");
-            ALOGV("loadGlobalConfig() mAttachedOutputDevices %04x", mAttachedOutputDevices);
-        } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
-            mDefaultOutputDevice = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
-                                              ARRAY_SIZE(sDeviceNameToEnumTable),
-                                              (char *)node->value);
-            ALOGW_IF(mDefaultOutputDevice == AUDIO_DEVICE_NONE,
-                    "loadGlobalConfig() default device not specified");
-            ALOGV("loadGlobalConfig() mDefaultOutputDevice %04x", mDefaultOutputDevice);
-        } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
-            mAvailableInputDevices = parseDeviceNames((char *)node->value) & ~AUDIO_DEVICE_BIT_IN;
-            ALOGV("loadGlobalConfig() mAvailableInputDevices %04x", mAvailableInputDevices);
-        } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
-            mSpeakerDrcEnabled = stringToBool((char *)node->value);
-            ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
-        }
-        node = node->next;
-    }
-}
-
-status_t AudioPolicyManagerBase::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);
-
-    loadGlobalConfig(root);
-    loadHwModules(root);
-
-    config_free(root);
-    free(root);
-    free(data);
-
-    ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
-
-    return NO_ERROR;
-}
-
-void AudioPolicyManagerBase::defaultAudioPolicyConfig(void)
-{
-    HwModule *module;
-    IOProfile *profile;
-
-    mDefaultOutputDevice = AUDIO_DEVICE_OUT_SPEAKER;
-    mAttachedOutputDevices = AUDIO_DEVICE_OUT_SPEAKER;
-    mAvailableInputDevices = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN;
-
-    module = new HwModule("primary");
-
-    profile = new IOProfile(module);
-    profile->mSamplingRates.add(44100);
-    profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
-    profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
-    profile->mSupportedDevices = AUDIO_DEVICE_OUT_SPEAKER;
-    profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
-    module->mOutputProfiles.add(profile);
-
-    profile = new IOProfile(module);
-    profile->mSamplingRates.add(8000);
-    profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
-    profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
-    profile->mSupportedDevices = AUDIO_DEVICE_IN_BUILTIN_MIC;
-    module->mInputProfiles.add(profile);
-
-    mHwModules.add(module);
-}
-
-}; // namespace android
diff --git a/audio/AudioPolicyManagerDefault.cpp b/audio/AudioPolicyManagerDefault.cpp
deleted file mode 100644
index 9083638..0000000
--- a/audio/AudioPolicyManagerDefault.cpp
+++ /dev/null
@@ -1,34 +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 "AudioPolicyManagerDefault"
-//#define LOG_NDEBUG 0
-
-#include "AudioPolicyManagerDefault.h"
-
-namespace android_audio_legacy {
-
-extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
-{
-    return new AudioPolicyManagerDefault(clientInterface);
-}
-
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
-{
-    delete interface;
-}
-
-}; // namespace android
diff --git a/audio/AudioPolicyManagerDefault.h b/audio/AudioPolicyManagerDefault.h
deleted file mode 100644
index 987fdf0..0000000
--- a/audio/AudioPolicyManagerDefault.h
+++ /dev/null
@@ -1,35 +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 <stdbool.h>
-
-#include <hardware_legacy/AudioPolicyManagerBase.h>
-
-namespace android_audio_legacy {
-
-class AudioPolicyManagerDefault: public AudioPolicyManagerBase
-{
-
-public:
-                explicit AudioPolicyManagerDefault(AudioPolicyClientInterface *clientInterface)
-                : AudioPolicyManagerBase(clientInterface) {}
-
-        virtual ~AudioPolicyManagerDefault() {}
-
-};
-};
diff --git a/audio/audio_policy_hal.cpp b/audio/audio_policy_hal.cpp
deleted file mode 100644
index b7fe245..0000000
--- a/audio/audio_policy_hal.cpp
+++ /dev/null
@@ -1,477 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "legacy_audio_policy_hal"
-//#define LOG_NDEBUG 0
-
-#include <stdint.h>
-
-#include <hardware/hardware.h>
-#include <system/audio.h>
-#include <system/audio_policy.h>
-#include <hardware/audio_policy.h>
-
-#include <hardware_legacy/AudioPolicyInterface.h>
-#include <hardware_legacy/AudioSystemLegacy.h>
-
-#include "AudioPolicyCompatClient.h"
-
-namespace android_audio_legacy {
-
-extern "C" {
-
-struct legacy_ap_module {
-    struct audio_policy_module module;
-};
-
-struct legacy_ap_device {
-    struct audio_policy_device device;
-};
-
-struct legacy_audio_policy {
-    struct audio_policy policy;
-
-    void *service;
-    struct audio_policy_service_ops *aps_ops;
-    AudioPolicyCompatClient *service_client;
-    AudioPolicyInterface *apm;
-};
-
-static inline struct legacy_audio_policy * to_lap(struct audio_policy *pol)
-{
-    return reinterpret_cast<struct legacy_audio_policy *>(pol);
-}
-
-static inline const struct legacy_audio_policy * to_clap(const struct audio_policy *pol)
-{
-    return reinterpret_cast<const struct legacy_audio_policy *>(pol);
-}
-
-
-static int ap_set_device_connection_state(struct audio_policy *pol,
-                                          audio_devices_t device,
-                                          audio_policy_dev_state_t state,
-                                          const char *device_address)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->setDeviceConnectionState(
-                    (AudioSystem::audio_devices)device,
-                    (AudioSystem::device_connection_state)state,
-                    device_address);
-}
-
-static audio_policy_dev_state_t ap_get_device_connection_state(
-                                            const struct audio_policy *pol,
-                                            audio_devices_t device,
-                                            const char *device_address)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return (audio_policy_dev_state_t)lap->apm->getDeviceConnectionState(
-                    (AudioSystem::audio_devices)device,
-                    device_address);
-}
-
-static void ap_set_phone_state(struct audio_policy *pol, audio_mode_t state)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    // as this is the legacy API, don't change it to use audio_mode_t instead of int
-    lap->apm->setPhoneState((int) state);
-}
-
-    /* indicate a change in ringer mode */
-static void ap_set_ringer_mode(struct audio_policy *pol, uint32_t mode,
-                               uint32_t mask)
-{
-    // deprecated, never called
-}
-
-    /* force using a specific device category for the specified usage */
-static void ap_set_force_use(struct audio_policy *pol,
-                          audio_policy_force_use_t usage,
-                          audio_policy_forced_cfg_t config)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->setForceUse((AudioSystem::force_use)usage,
-                          (AudioSystem::forced_config)config);
-}
-
-    /* retreive current device category forced for a given usage */
-static audio_policy_forced_cfg_t ap_get_force_use(
-                                               const struct audio_policy *pol,
-                                               audio_policy_force_use_t usage)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return (audio_policy_forced_cfg_t)lap->apm->getForceUse(
-                          (AudioSystem::force_use)usage);
-}
-
-/* if can_mute is true, then audio streams that are marked ENFORCED_AUDIBLE
- * can still be muted. */
-static void ap_set_can_mute_enforced_audible(struct audio_policy *pol,
-                                             bool can_mute)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->setSystemProperty("ro.camera.sound.forced", can_mute ? "0" : "1");
-}
-
-static int ap_init_check(const struct audio_policy *pol)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->initCheck();
-}
-
-static audio_io_handle_t ap_get_output(struct audio_policy *pol,
-                                       audio_stream_type_t stream,
-                                       uint32_t sampling_rate,
-                                       audio_format_t format,
-                                       audio_channel_mask_t channelMask,
-                                       audio_output_flags_t flags,
-                                       const audio_offload_info_t *offloadInfo)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-
-    ALOGV("%s: tid %d", __func__, gettid());
-    return lap->apm->getOutput((AudioSystem::stream_type)stream,
-                               sampling_rate, format, channelMask,
-                               (AudioSystem::output_flags)flags,
-                               offloadInfo);
-}
-
-static int ap_start_output(struct audio_policy *pol, audio_io_handle_t output,
-                           audio_stream_type_t stream, audio_session_t session)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->startOutput(output, (AudioSystem::stream_type)stream,
-                                 session);
-}
-
-static int ap_stop_output(struct audio_policy *pol, audio_io_handle_t output,
-                          audio_stream_type_t stream, audio_session_t session)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->stopOutput(output, (AudioSystem::stream_type)stream,
-                                session);
-}
-
-static void ap_release_output(struct audio_policy *pol,
-                              audio_io_handle_t output)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->releaseOutput(output);
-}
-
-static audio_io_handle_t ap_get_input(struct audio_policy *pol, audio_source_t inputSource,
-                                      uint32_t sampling_rate,
-                                      audio_format_t format,
-                                      audio_channel_mask_t channelMask,
-                                      audio_in_acoustics_t acoustics)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->getInput((int) inputSource, sampling_rate, format, channelMask,
-                              (AudioSystem::audio_in_acoustics)acoustics);
-}
-
-static int ap_start_input(struct audio_policy *pol, audio_io_handle_t input)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->startInput(input);
-}
-
-static int ap_stop_input(struct audio_policy *pol, audio_io_handle_t input)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->stopInput(input);
-}
-
-static void ap_release_input(struct audio_policy *pol, audio_io_handle_t input)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->releaseInput(input);
-}
-
-static void ap_init_stream_volume(struct audio_policy *pol,
-                                  audio_stream_type_t stream, int index_min,
-                                  int index_max)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    lap->apm->initStreamVolume((AudioSystem::stream_type)stream, index_min,
-                               index_max);
-}
-
-static int ap_set_stream_volume_index(struct audio_policy *pol,
-                                      audio_stream_type_t stream,
-                                      int index)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->setStreamVolumeIndex((AudioSystem::stream_type)stream,
-                                          index,
-                                          AUDIO_DEVICE_OUT_DEFAULT);
-}
-
-static int ap_get_stream_volume_index(const struct audio_policy *pol,
-                                      audio_stream_type_t stream,
-                                      int *index)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->getStreamVolumeIndex((AudioSystem::stream_type)stream,
-                                          index,
-                                          AUDIO_DEVICE_OUT_DEFAULT);
-}
-
-static int ap_set_stream_volume_index_for_device(struct audio_policy *pol,
-                                      audio_stream_type_t stream,
-                                      int index,
-                                      audio_devices_t device)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->setStreamVolumeIndex((AudioSystem::stream_type)stream,
-                                          index,
-                                          device);
-}
-
-static int ap_get_stream_volume_index_for_device(const struct audio_policy *pol,
-                                      audio_stream_type_t stream,
-                                      int *index,
-                                      audio_devices_t device)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->getStreamVolumeIndex((AudioSystem::stream_type)stream,
-                                          index,
-                                          device);
-}
-
-static uint32_t ap_get_strategy_for_stream(const struct audio_policy *pol,
-                                           audio_stream_type_t stream)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->getStrategyForStream((AudioSystem::stream_type)stream);
-}
-
-static audio_devices_t ap_get_devices_for_stream(const struct audio_policy *pol,
-                                       audio_stream_type_t stream)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->getDevicesForStream((AudioSystem::stream_type)stream);
-}
-
-static audio_io_handle_t ap_get_output_for_effect(struct audio_policy *pol,
-                                            const struct effect_descriptor_s *desc)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->getOutputForEffect(desc);
-}
-
-static int ap_register_effect(struct audio_policy *pol,
-                              const struct effect_descriptor_s *desc,
-                              audio_io_handle_t io,
-                              uint32_t strategy,
-                              audio_session_t session,
-                              int id)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->registerEffect(desc, io, strategy, session, id);
-}
-
-static int ap_unregister_effect(struct audio_policy *pol, int id)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->unregisterEffect(id);
-}
-
-static int ap_set_effect_enabled(struct audio_policy *pol, int id, bool enabled)
-{
-    struct legacy_audio_policy *lap = to_lap(pol);
-    return lap->apm->setEffectEnabled(id, enabled);
-}
-
-static bool ap_is_stream_active(const struct audio_policy *pol, audio_stream_type_t stream,
-                                uint32_t in_past_ms)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->isStreamActive((int) stream, in_past_ms);
-}
-
-static bool ap_is_stream_active_remotely(const struct audio_policy *pol, audio_stream_type_t stream,
-                                uint32_t in_past_ms)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->isStreamActiveRemotely((int) stream, in_past_ms);
-}
-
-static bool ap_is_source_active(const struct audio_policy *pol, audio_source_t source)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->isSourceActive(source);
-}
-
-static int ap_dump(const struct audio_policy *pol, int fd)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->dump(fd);
-}
-
-static bool ap_is_offload_supported(const struct audio_policy *pol,
-                                    const audio_offload_info_t *info)
-{
-    const struct legacy_audio_policy *lap = to_clap(pol);
-    return lap->apm->isOffloadSupported(*info);
-}
-
-static int create_legacy_ap(const struct audio_policy_device *device,
-                            struct audio_policy_service_ops *aps_ops,
-                            void *service,
-                            struct audio_policy **ap)
-{
-    struct legacy_audio_policy *lap;
-    int ret;
-
-    if (!service || !aps_ops)
-        return -EINVAL;
-
-    lap = (struct legacy_audio_policy *)calloc(1, sizeof(*lap));
-    if (!lap)
-        return -ENOMEM;
-
-    lap->policy.set_device_connection_state = ap_set_device_connection_state;
-    lap->policy.get_device_connection_state = ap_get_device_connection_state;
-    lap->policy.set_phone_state = ap_set_phone_state;
-    lap->policy.set_ringer_mode = ap_set_ringer_mode;
-    lap->policy.set_force_use = ap_set_force_use;
-    lap->policy.get_force_use = ap_get_force_use;
-    lap->policy.set_can_mute_enforced_audible =
-        ap_set_can_mute_enforced_audible;
-    lap->policy.init_check = ap_init_check;
-    lap->policy.get_output = ap_get_output;
-    lap->policy.start_output = ap_start_output;
-    lap->policy.stop_output = ap_stop_output;
-    lap->policy.release_output = ap_release_output;
-    lap->policy.get_input = ap_get_input;
-    lap->policy.start_input = ap_start_input;
-    lap->policy.stop_input = ap_stop_input;
-    lap->policy.release_input = ap_release_input;
-    lap->policy.init_stream_volume = ap_init_stream_volume;
-    lap->policy.set_stream_volume_index = ap_set_stream_volume_index;
-    lap->policy.get_stream_volume_index = ap_get_stream_volume_index;
-    lap->policy.set_stream_volume_index_for_device = ap_set_stream_volume_index_for_device;
-    lap->policy.get_stream_volume_index_for_device = ap_get_stream_volume_index_for_device;
-    lap->policy.get_strategy_for_stream = ap_get_strategy_for_stream;
-    lap->policy.get_devices_for_stream = ap_get_devices_for_stream;
-    lap->policy.get_output_for_effect = ap_get_output_for_effect;
-    lap->policy.register_effect = ap_register_effect;
-    lap->policy.unregister_effect = ap_unregister_effect;
-    lap->policy.set_effect_enabled = ap_set_effect_enabled;
-    lap->policy.is_stream_active = ap_is_stream_active;
-    lap->policy.is_stream_active_remotely = ap_is_stream_active_remotely;
-    lap->policy.is_source_active = ap_is_source_active;
-    lap->policy.dump = ap_dump;
-    lap->policy.is_offload_supported = ap_is_offload_supported;
-
-    lap->service = service;
-    lap->aps_ops = aps_ops;
-    lap->service_client =
-        new AudioPolicyCompatClient(aps_ops, service);
-    if (!lap->service_client) {
-        ret = -ENOMEM;
-        goto err_new_compat_client;
-    }
-
-    lap->apm = createAudioPolicyManager(lap->service_client);
-    if (!lap->apm) {
-        ret = -ENOMEM;
-        goto err_create_apm;
-    }
-
-    *ap = &lap->policy;
-    return 0;
-
-err_create_apm:
-    delete lap->service_client;
-err_new_compat_client:
-    free(lap);
-    *ap = NULL;
-    return ret;
-}
-
-static int destroy_legacy_ap(const struct audio_policy_device *ap_dev,
-                             struct audio_policy *ap)
-{
-    struct legacy_audio_policy *lap = to_lap(ap);
-
-    if (!lap)
-        return 0;
-
-    if (lap->apm)
-        destroyAudioPolicyManager(lap->apm);
-    if (lap->service_client)
-        delete lap->service_client;
-    free(lap);
-    return 0;
-}
-
-static int legacy_ap_dev_close(hw_device_t* device)
-{
-    if (device)
-        free(device);
-    return 0;
-}
-
-static int legacy_ap_dev_open(const hw_module_t* module, const char* name,
-                                    hw_device_t** device)
-{
-    struct legacy_ap_device *dev;
-
-    if (strcmp(name, AUDIO_POLICY_INTERFACE) != 0)
-        return -EINVAL;
-
-    dev = (struct legacy_ap_device *)calloc(1, sizeof(*dev));
-    if (!dev)
-        return -ENOMEM;
-
-    dev->device.common.tag = HARDWARE_DEVICE_TAG;
-    dev->device.common.version = 0;
-    dev->device.common.module = const_cast<hw_module_t*>(module);
-    dev->device.common.close = legacy_ap_dev_close;
-    dev->device.create_audio_policy = create_legacy_ap;
-    dev->device.destroy_audio_policy = destroy_legacy_ap;
-
-    *device = &dev->device.common;
-
-    return 0;
-}
-
-static struct hw_module_methods_t legacy_ap_module_methods = {
-        .open = legacy_ap_dev_open
-};
-
-struct legacy_ap_module HAL_MODULE_INFO_SYM = {
-    .module = {
-        .common = {
-            .tag = HARDWARE_MODULE_TAG,
-            .version_major = 1,
-            .version_minor = 0,
-            .id = AUDIO_POLICY_HARDWARE_MODULE_ID,
-            .name = "LEGACY Audio Policy HAL",
-            .author = "The Android Open Source Project",
-            .methods = &legacy_ap_module_methods,
-            .dso = NULL,
-            .reserved = {0},
-        },
-    },
-};
-
-}; // extern "C"
-
-}; // namespace android_audio_legacy
diff --git a/power/power.c b/power.c
similarity index 100%
rename from power/power.c
rename to power.c
diff --git a/power/Android.mk b/power/Android.mk
deleted file mode 100644
index 3e3ff5d..0000000
--- a/power/Android.mk
+++ /dev/null
@@ -1,3 +0,0 @@
-# Copyright 2006 The Android Open Source Project
-
-LOCAL_SRC_FILES += power/power.c
diff --git a/uevent/uevent.c b/uevent.c
similarity index 100%
rename from uevent/uevent.c
rename to uevent.c
diff --git a/uevent/Android.mk b/uevent/Android.mk
deleted file mode 100644
index 2d8b524..0000000
--- a/uevent/Android.mk
+++ /dev/null
@@ -1,3 +0,0 @@
-# Copyright 2008 The Android Open Source Project
-
-LOCAL_SRC_FILES += uevent/uevent.c