diff --git a/libs/audioflinger/Android.mk b/libs/audioflinger/Android.mk
index f5c03bb..b68bfc1 100644
--- a/libs/audioflinger/Android.mk
+++ b/libs/audioflinger/Android.mk
@@ -47,7 +47,7 @@
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:=               \
-    AudioPolicyManagerGeneric.cpp
+    AudioPolicyManagerBase.cpp
 
 LOCAL_SHARED_LIBRARIES := \
     libcutils \
@@ -60,7 +60,7 @@
  LOCAL_SHARED_LIBRARIES += libdl
 endif
 
-LOCAL_MODULE:= libaudiopolicygeneric
+LOCAL_MODULE:= libaudiopolicybase
 
 ifeq ($(BOARD_HAVE_BLUETOOTH),true)
   LOCAL_CFLAGS += -DWITH_A2DP
@@ -70,7 +70,7 @@
   LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
 endif
 
-include $(BUILD_SHARED_LIBRARY)
+include $(BUILD_STATIC_LIBRARY)
 
 include $(CLEAR_VARS)
 
@@ -87,11 +87,10 @@
     libutils \
 	libbinder \
     libmedia \
-    libhardware_legacy \
-    libaudiopolicygeneric
+    libhardware_legacy
 
 ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true)
-  LOCAL_STATIC_LIBRARIES += libaudiointerface
+  LOCAL_STATIC_LIBRARIES += libaudiointerface libaudiopolicybase
   LOCAL_CFLAGS += -DGENERIC_AUDIO
 else
   LOCAL_SHARED_LIBRARIES += libaudio libaudiopolicy
diff --git a/libs/audioflinger/AudioPolicyManagerBase.cpp b/libs/audioflinger/AudioPolicyManagerBase.cpp
new file mode 100644
index 0000000..055dbca
--- /dev/null
+++ b/libs/audioflinger/AudioPolicyManagerBase.cpp
@@ -0,0 +1,1925 @@
+/*
+ * 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
+#include <utils/Log.h>
+#include <hardware_legacy/AudioPolicyManagerBase.h>
+#include <media/mediarecorder.h>
+
+namespace android {
+
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+
+
+status_t AudioPolicyManagerBase::setDeviceConnectionState(AudioSystem::audio_devices device,
+                                                  AudioSystem::device_connection_state state,
+                                                  const char *device_address)
+{
+
+    LOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
+
+    // connect/disconnect only 1 device at a time
+    if (AudioSystem::popCount(device) != 1) return BAD_VALUE;
+
+    if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
+        LOGE("setDeviceConnectionState() invalid address: %s", device_address);
+        return BAD_VALUE;
+    }
+
+    // handle output devices
+    if (AudioSystem::isOutputDevice(device)) {
+
+#ifndef WITH_A2DP
+        if (AudioSystem::isA2dpDevice(device)) {
+            LOGE("setDeviceConnectionState() invalid device: %x", device);
+            return BAD_VALUE;
+        }
+#endif
+
+        switch (state)
+        {
+        // handle output device connection
+        case AudioSystem::DEVICE_STATE_AVAILABLE:
+            if (mAvailableOutputDevices & device) {
+                LOGW("setDeviceConnectionState() device already connected: %x", device);
+                return INVALID_OPERATION;
+            }
+            LOGV("setDeviceConnectionState() connecting device %x", device);
+
+            // register new device as available
+            mAvailableOutputDevices |= device;
+
+#ifdef WITH_A2DP
+            // handle A2DP device connection
+            if (AudioSystem::isA2dpDevice(device)) {
+                status_t status = handleA2dpConnection(device, device_address);
+                if (status != NO_ERROR) {
+                    mAvailableOutputDevices &= ~device;
+                    return status;
+                }
+            } else
+#endif
+            {
+                if (AudioSystem::isBluetoothScoDevice(device)) {
+                    LOGV("setDeviceConnectionState() BT SCO  device, address %s", device_address);
+                    // keep track of SCO device address
+                    mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+#ifdef WITH_A2DP
+                    if ((mA2dpDeviceAddress == mScoDeviceAddress) &&
+                        (mPhoneState != AudioSystem::MODE_NORMAL)) {
+                        mpClientInterface->suspendOutput(mA2dpOutput);
+                    }
+#endif
+                }
+            }
+            break;
+        // handle output device disconnection
+        case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
+            if (!(mAvailableOutputDevices & device)) {
+                LOGW("setDeviceConnectionState() device not connected: %x", device);
+                return INVALID_OPERATION;
+            }
+
+
+            LOGV("setDeviceConnectionState() disconnecting device %x", device);
+            // remove device from available output devices
+            mAvailableOutputDevices &= ~device;
+
+#ifdef WITH_A2DP
+            // handle A2DP device disconnection
+            if (AudioSystem::isA2dpDevice(device)) {
+                status_t status = handleA2dpDisconnection(device, device_address);
+                if (status != NO_ERROR) {
+                    mAvailableOutputDevices |= device;
+                    return status;
+                }
+            } else
+#endif
+            {
+                if (AudioSystem::isBluetoothScoDevice(device)) {
+                    mScoDeviceAddress = "";
+#ifdef WITH_A2DP
+                    if ((mA2dpDeviceAddress == mScoDeviceAddress) &&
+                        (mPhoneState != AudioSystem::MODE_NORMAL)) {
+                        mpClientInterface->restoreOutput(mA2dpOutput);
+                    }
+#endif
+                }
+            }
+            } break;
+
+        default:
+            LOGE("setDeviceConnectionState() invalid state: %x", state);
+            return BAD_VALUE;
+        }
+
+        // request routing change if necessary
+        uint32_t newDevice = getNewDevice(mHardwareOutput, false);
+#ifdef WITH_A2DP
+        checkOutputForAllStrategies(newDevice);
+        // A2DP outputs must be closed after checkOutputForAllStrategies() is executed
+        if (state == AudioSystem::DEVICE_STATE_UNAVAILABLE && AudioSystem::isA2dpDevice(device)) {
+            closeA2dpOutputs();
+        }
+#endif
+        updateDeviceForStrategy();
+        setOutputDevice(mHardwareOutput, newDevice);
+
+        if (device == AudioSystem::DEVICE_OUT_WIRED_HEADSET) {
+            device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
+        } else if (device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO ||
+                   device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
+                   device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
+            device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+        } else {
+            return NO_ERROR;
+        }
+    }
+    // handle input devices
+    if (AudioSystem::isInputDevice(device)) {
+
+        switch (state)
+        {
+        // handle input device connection
+        case AudioSystem::DEVICE_STATE_AVAILABLE: {
+            if (mAvailableInputDevices & device) {
+                LOGW("setDeviceConnectionState() device already connected: %d", device);
+                return INVALID_OPERATION;
+            }
+            mAvailableInputDevices |= device;
+            }
+            break;
+
+        // handle input device disconnection
+        case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
+            if (!(mAvailableInputDevices & device)) {
+                LOGW("setDeviceConnectionState() device not connected: %d", device);
+                return INVALID_OPERATION;
+            }
+            mAvailableInputDevices &= ~device;
+            } break;
+
+        default:
+            LOGE("setDeviceConnectionState() invalid state: %x", state);
+            return BAD_VALUE;
+        }
+
+        audio_io_handle_t activeInput = getActiveInput();
+        if (activeInput != 0) {
+            AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
+            uint32_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+            if (newDevice != inputDesc->mDevice) {
+                LOGV("setDeviceConnectionState() 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());
+            }
+        }
+
+        return NO_ERROR;
+    }
+
+    LOGW("setDeviceConnectionState() invalid device: %x", device);
+    return BAD_VALUE;
+}
+
+AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(AudioSystem::audio_devices device,
+                                                  const char *device_address)
+{
+    AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
+    String8 address = String8(device_address);
+    if (AudioSystem::isOutputDevice(device)) {
+        if (device & mAvailableOutputDevices) {
+#ifdef WITH_A2DP
+            if (AudioSystem::isA2dpDevice(device) &&
+                address != "" && mA2dpDeviceAddress != address) {
+                return state;
+            }
+#endif
+            if (AudioSystem::isBluetoothScoDevice(device) &&
+                address != "" && mScoDeviceAddress != address) {
+                return state;
+            }
+            state = AudioSystem::DEVICE_STATE_AVAILABLE;
+        }
+    } else if (AudioSystem::isInputDevice(device)) {
+        if (device & mAvailableInputDevices) {
+            state = AudioSystem::DEVICE_STATE_AVAILABLE;
+        }
+    }
+
+    return state;
+}
+
+void AudioPolicyManagerBase::setPhoneState(int state)
+{
+    LOGV("setPhoneState() state %d", state);
+    uint32_t newDevice = 0;
+    if (state < 0 || state >= AudioSystem::NUM_MODES) {
+        LOGW("setPhoneState() invalid state %d", state);
+        return;
+    }
+
+    if (state == mPhoneState ) {
+        LOGW("setPhoneState() setting same state %d", state);
+        return;
+    }
+
+    // if leaving call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+        LOGV("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 ((oldState != AudioSystem::MODE_IN_CALL) && (state == AudioSystem::MODE_IN_CALL)) {
+        LOGV("  Entering call in setPhoneState()");
+        // force routing command to audio hardware when starting a call
+        // even if no device change is needed
+        force = true;
+    } else if ((oldState == AudioSystem::MODE_IN_CALL) && (state != AudioSystem::MODE_IN_CALL)) {
+        LOGV("  Exiting call in setPhoneState()");
+        // force routing command to audio hardware when exiting a call
+        // even if no device change is needed
+        force = true;
+    }
+
+    // check for device and output changes triggered by new phone state
+    newDevice = getNewDevice(mHardwareOutput, false);
+#ifdef WITH_A2DP
+    checkOutputForAllStrategies(newDevice);
+    // suspend A2DP output if SCO device address is the same as A2DP device address.
+    // no need to check that a SCO device is actually connected as mScoDeviceAddress == ""
+    // if none is connected and the test below will fail.
+    if (mA2dpDeviceAddress == mScoDeviceAddress) {
+        if (oldState == AudioSystem::MODE_NORMAL) {
+            mpClientInterface->suspendOutput(mA2dpOutput);
+        } else if (state == AudioSystem::MODE_NORMAL) {
+            mpClientInterface->restoreOutput(mA2dpOutput);
+        }
+    }
+#endif
+    updateDeviceForStrategy();
+
+    AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
+
+    // force routing command to audio hardware when ending call
+    // even if no device change is needed
+    if (oldState == AudioSystem::MODE_IN_CALL && newDevice == 0) {
+        newDevice = hwOutputDesc->device();
+    }
+    // change routing is necessary
+    setOutputDevice(mHardwareOutput, newDevice, force);
+
+    // if entering in call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (state == AudioSystem::MODE_IN_CALL) {
+        LOGV("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 &&
+        (hwOutputDesc->mRefCount[AudioSystem::MUSIC] ||
+        (systemTime() - mMusicStopTime) < seconds(SONIFICATION_HEADSET_MUSIC_DELAY))) {
+        mLimitRingtoneVolume = true;
+    } else {
+        mLimitRingtoneVolume = false;
+    }
+}
+
+void AudioPolicyManagerBase::setRingerMode(uint32_t mode, uint32_t mask)
+{
+    LOGV("setRingerMode() mode %x, mask %x", mode, mask);
+
+    mRingerMode = mode;
+}
+
+void AudioPolicyManagerBase::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
+{
+    LOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
+
+    switch(usage) {
+    case AudioSystem::FOR_COMMUNICATION:
+        if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO &&
+            config != AudioSystem::FORCE_NONE) {
+            LOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
+            return;
+        }
+        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_NONE) {
+            LOGW("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) {
+            LOGW("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) {
+            LOGW("setForceUse() invalid config %d for FOR_DOCK", config);
+        }
+        mForceUse[usage] = config;
+        break;
+    default:
+        LOGW("setForceUse() invalid usage %d", usage);
+        break;
+    }
+
+    // check for device and output changes triggered by new phone state
+    uint32_t newDevice = getNewDevice(mHardwareOutput, false);
+#ifdef WITH_A2DP
+    checkOutputForAllStrategies(newDevice);
+#endif
+    updateDeviceForStrategy();
+    setOutputDevice(mHardwareOutput, newDevice);
+}
+
+AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
+{
+    return mForceUse[usage];
+}
+
+void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value)
+{
+    LOGV("setSystemProperty() property %s, value %s", property, value);
+    if (strcmp(property, "ro.camera.sound.forced") == 0) {
+        if (atoi(value)) {
+            LOGV("ENFORCED_AUDIBLE cannot be muted");
+            mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false;
+        } else {
+            LOGV("ENFORCED_AUDIBLE can be muted");
+            mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true;
+        }
+    }
+}
+
+audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channels,
+                                    AudioSystem::output_flags flags)
+{
+    audio_io_handle_t output = 0;
+    uint32_t latency = 0;
+    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+    uint32_t device = getDeviceForStrategy(strategy);
+    LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags);
+
+#ifdef AUDIO_POLICY_TEST
+    if (mCurOutput != 0) {
+        LOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channels %x, mDirectOutput %d",
+                mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+        if (mTestOutputs[mCurOutput] == 0) {
+            LOGV("getOutput() opening test output");
+            AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+            outputDesc->mDevice = mTestDevice;
+            outputDesc->mSamplingRate = mTestSamplingRate;
+            outputDesc->mFormat = mTestFormat;
+            outputDesc->mChannels = mTestChannels;
+            outputDesc->mLatency = mTestLatencyMs;
+            outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
+            outputDesc->mRefCount[stream] = 0;
+            mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice,
+                                            &outputDesc->mSamplingRate,
+                                            &outputDesc->mFormat,
+                                            &outputDesc->mChannels,
+                                            &outputDesc->mLatency,
+                                            outputDesc->mFlags);
+            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:
+    // 1 a direct output is explicitely requested
+    // 2 the audio format is compressed
+    if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+         (format !=0 && !AudioSystem::isLinearPCM(format))) {
+
+        LOGV("getOutput() opening direct output device %x", device);
+        AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+        outputDesc->mDevice = device;
+        outputDesc->mSamplingRate = samplingRate;
+        outputDesc->mFormat = format;
+        outputDesc->mChannels = channels;
+        outputDesc->mLatency = 0;
+        outputDesc->mFlags = (AudioSystem::output_flags)(flags | AudioSystem::OUTPUT_FLAG_DIRECT);
+        outputDesc->mRefCount[stream] = 1;
+        output = mpClientInterface->openOutput(&outputDesc->mDevice,
+                                        &outputDesc->mSamplingRate,
+                                        &outputDesc->mFormat,
+                                        &outputDesc->mChannels,
+                                        &outputDesc->mLatency,
+                                        outputDesc->mFlags);
+
+        // only accept an output with the requeted parameters
+        if (output == 0 ||
+            (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
+            (format != 0 && format != outputDesc->mFormat) ||
+            (channels != 0 && channels != outputDesc->mChannels)) {
+            LOGV("getOutput() failed opening direct output: samplingRate %d, format %d, channels %d",
+                    samplingRate, format, channels);
+            if (output != 0) {
+                mpClientInterface->closeOutput(output);
+            }
+            delete outputDesc;
+            return 0;
+        }
+        addOutput(output, outputDesc);
+        return output;
+    }
+
+    if (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO &&
+        channels != AudioSystem::CHANNEL_OUT_STEREO) {
+        return 0;
+    }
+    // open a non direct output
+
+    // get which output is suitable for the specified stream. The actual routing change will happen
+    // when startOutput() will be called
+    uint32_t a2dpDevice = device & AudioSystem::DEVICE_OUT_ALL_A2DP;
+    if (AudioSystem::popCount((AudioSystem::audio_devices)device) == 2) {
+#ifdef WITH_A2DP
+        if (a2dpUsedForSonification() && a2dpDevice != 0) {
+            // if playing on 2 devices among which one is A2DP, use duplicated output
+            LOGV("getOutput() using duplicated output");
+            LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device in multiple %x selected but A2DP output not opened", device);
+            output = mDuplicatedOutput;
+        } else
+#endif
+        {
+            // if playing on 2 devices among which none is A2DP, use hardware output
+            output = mHardwareOutput;
+        }
+        LOGV("getOutput() using output %d for 2 devices %x", output, device);
+    } else {
+#ifdef WITH_A2DP
+        if (a2dpDevice != 0) {
+            // if playing on A2DP device, use a2dp output
+            LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device %x selected but A2DP output not opened", device);
+            output = mA2dpOutput;
+        } else
+#endif
+        {
+            // if playing on not A2DP device, use hardware output
+            output = mHardwareOutput;
+        }
+    }
+
+
+    LOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
+                stream, samplingRate, format, channels, flags);
+
+    return output;
+}
+
+status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+    LOGV("startOutput() output %d, stream %d", output, stream);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        LOGW("startOutput() unknow output %d", output);
+        return BAD_VALUE;
+    }
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+
+#ifdef WITH_A2DP
+    if (mA2dpOutput != 0  && !a2dpUsedForSonification() && strategy == STRATEGY_SONIFICATION) {
+        setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
+    }
+#endif
+
+    // incremenent 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
+    // necassary for a correct control of hardware output routing by startOutput() and stopOutput()
+    outputDesc->changeRefCount(stream, 1);
+
+    setOutputDevice(output, getNewDevice(output));
+
+    // handle special case for sonification while in call
+    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+        handleIncallSonification(stream, true, false);
+    }
+
+    // apply volume rules for current stream and device if necessary
+    checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, outputDesc->device());
+
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+    LOGV("stopOutput() output %d, stream %d", output, stream);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        LOGW("stopOutput() unknow output %d", output);
+        return BAD_VALUE;
+    }
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+
+    // handle special case for sonification while in call
+    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+        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 last music track was stopped - see computeVolume()
+        if (stream == AudioSystem::MUSIC) {
+            mMusicStopTime = systemTime();
+        }
+
+        setOutputDevice(output, getNewDevice(output));
+
+#ifdef WITH_A2DP
+        if (mA2dpOutput != 0 && !a2dpUsedForSonification() && strategy == STRATEGY_SONIFICATION) {
+            setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput, mOutputs.valueFor(mHardwareOutput)->mLatency*2);
+        }
+#endif
+        return NO_ERROR;
+    } else {
+        LOGW("stopOutput() refcount is already 0 for output %d", output);
+        return INVALID_OPERATION;
+    }
+}
+
+void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output)
+{
+    LOGV("releaseOutput() %d", output);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        LOGW("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->refCount() == 0) {
+            mpClientInterface->closeOutput(output);
+            delete mOutputs.valueAt(index);
+            mOutputs.removeItem(output);
+            mTestOutputs[testIndex] = 0;
+        }
+        return;
+    }
+#endif //AUDIO_POLICY_TEST
+
+    if (mOutputs.valueAt(index)->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) {
+        mpClientInterface->closeOutput(output);
+        delete mOutputs.valueAt(index);
+        mOutputs.removeItem(output);
+    }
+}
+
+audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channels,
+                                    AudioSystem::audio_in_acoustics acoustics)
+{
+    audio_io_handle_t input = 0;
+    uint32_t device = getDeviceForInputSource(inputSource);
+
+    LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics);
+
+    if (device == 0) {
+        return 0;
+    }
+
+    // adapt channel selection to input source
+    switch(inputSource) {
+    case AUDIO_SOURCE_VOICE_UPLINK:
+        channels = AudioSystem::CHANNEL_IN_VOICE_UPLINK;
+        break;
+    case AUDIO_SOURCE_VOICE_DOWNLINK:
+        channels = AudioSystem::CHANNEL_IN_VOICE_DNLINK;
+        break;
+    case AUDIO_SOURCE_VOICE_CALL:
+        channels = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK);
+        break;
+    default:
+        break;
+    }
+
+    AudioInputDescriptor *inputDesc = new AudioInputDescriptor();
+
+    inputDesc->mInputSource = inputSource;
+    inputDesc->mDevice = device;
+    inputDesc->mSamplingRate = samplingRate;
+    inputDesc->mFormat = format;
+    inputDesc->mChannels = channels;
+    inputDesc->mAcoustics = acoustics;
+    inputDesc->mRefCount = 0;
+    input = mpClientInterface->openInput(&inputDesc->mDevice,
+                                    &inputDesc->mSamplingRate,
+                                    &inputDesc->mFormat,
+                                    &inputDesc->mChannels,
+                                    inputDesc->mAcoustics);
+
+    // only accept input with the exact requested set of parameters
+    if (input == 0 ||
+        (samplingRate != inputDesc->mSamplingRate) ||
+        (format != inputDesc->mFormat) ||
+        (channels != inputDesc->mChannels)) {
+        LOGV("getInput() failed opening input: samplingRate %d, format %d, channels %d",
+                samplingRate, format, channels);
+        if (input != 0) {
+            mpClientInterface->closeInput(input);
+        }
+        delete inputDesc;
+        return 0;
+    }
+    mInputs.add(input, inputDesc);
+    return input;
+}
+
+status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input)
+{
+    LOGV("startInput() input %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        LOGW("startInput() unknow 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
+        if (getActiveInput() != 0) {
+            LOGW("startInput() input %d failed: other input already started", input);
+            return INVALID_OPERATION;
+        }
+    }
+
+    AudioParameter param = AudioParameter();
+    param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
+
+    // use Voice Recognition mode or not for this input based on input source
+    int vr_enabled = inputDesc->mInputSource == AUDIO_SOURCE_VOICE_RECOGNITION ? 1 : 0;
+    param.addInt(String8("vr_mode"), vr_enabled);
+    LOGV("AudioPolicyManager::startInput(%d), setting vr_mode to %d", inputDesc->mInputSource, vr_enabled);
+
+    mpClientInterface->setParameters(input, param.toString());
+
+    inputDesc->mRefCount = 1;
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input)
+{
+    LOGV("stopInput() input %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        LOGW("stopInput() unknow input %d", input);
+        return BAD_VALUE;
+    }
+    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+    if (inputDesc->mRefCount == 0) {
+        LOGW("stopInput() input %d already stopped", input);
+        return INVALID_OPERATION;
+    } else {
+        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)
+{
+    LOGV("releaseInput() %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        LOGW("releaseInput() releasing unknown input %d", input);
+        return;
+    }
+    mpClientInterface->closeInput(input);
+    delete mInputs.valueAt(index);
+    mInputs.removeItem(input);
+    LOGV("releaseInput() exit");
+}
+
+void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream,
+                                            int indexMin,
+                                            int indexMax)
+{
+    LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+    if (indexMin < 0 || indexMin >= indexMax) {
+        LOGW("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)
+{
+
+    if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
+        return BAD_VALUE;
+    }
+
+    // Force max volume if stream cannot be muted
+    if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
+
+    LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index);
+    mStreams[stream].mIndexCur = 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++) {
+        status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), mOutputs.valueAt(i)->device());
+        if (volStatus != NO_ERROR) {
+            status = volStatus;
+        }
+    }
+    return status;
+}
+
+status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
+{
+    if (index == 0) {
+        return BAD_VALUE;
+    }
+    LOGV("getStreamVolumeIndex() stream %d", stream);
+    *index =  mStreams[stream].mIndexCur;
+    return NO_ERROR;
+}
+
+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, " Hardware Output: %d\n", mHardwareOutput);
+    result.append(buffer);
+#ifdef WITH_A2DP
+    snprintf(buffer, SIZE, " A2DP Output: %d\n", mA2dpOutput);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Duplicated Output: %d\n", mDuplicatedOutput);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
+    result.append(buffer);
+#endif
+    snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.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, " Ringer mode: %d\n", mRingerMode);
+    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);
+    write(fd, result.string(), result.size());
+
+    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  Index Min  Index Max  Index Cur  Can be muted\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+        snprintf(buffer, SIZE, " %02d", i);
+        mStreams[i].dump(buffer + 3, SIZE);
+        write(fd, buffer, strlen(buffer));
+    }
+
+    return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManagerBase
+// ----------------------------------------------------------------------------
+
+AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
+    :
+#ifdef AUDIO_POLICY_TEST
+    Thread(false),
+#endif //AUDIO_POLICY_TEST
+    mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0), mMusicStopTime(0), mLimitRingtoneVolume(false)
+{
+    mpClientInterface = clientInterface;
+
+    for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
+        mForceUse[i] = AudioSystem::FORCE_NONE;
+    }
+
+    // devices available by default are speaker, ear piece and microphone
+    mAvailableOutputDevices = AudioSystem::DEVICE_OUT_EARPIECE |
+                        AudioSystem::DEVICE_OUT_SPEAKER;
+    mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+
+#ifdef WITH_A2DP
+    mA2dpOutput = 0;
+    mDuplicatedOutput = 0;
+    mA2dpDeviceAddress = String8("");
+#endif
+    mScoDeviceAddress = String8("");
+
+    // open hardware output
+    AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+    outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
+    mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
+                                    &outputDesc->mSamplingRate,
+                                    &outputDesc->mFormat,
+                                    &outputDesc->mChannels,
+                                    &outputDesc->mLatency,
+                                    outputDesc->mFlags);
+
+    if (mHardwareOutput == 0) {
+        LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d",
+                outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
+    } else {
+        addOutput(mHardwareOutput, outputDesc);
+        setOutputDevice(mHardwareOutput, (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER, true);
+    }
+
+    updateDeviceForStrategy();
+#ifdef AUDIO_POLICY_TEST
+    AudioParameter outputCmd = AudioParameter();
+    outputCmd.addInt(String8("set_id"), 0);
+    mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
+
+    mTestDevice = AudioSystem::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);
+   }
+   mOutputs.clear();
+   for (size_t i = 0; i < mInputs.size(); i++) {
+        mpClientInterface->closeInput(mInputs.keyAt(i));
+        delete mInputs.valueAt(i);
+   }
+   mInputs.clear();
+}
+
+#ifdef AUDIO_POLICY_TEST
+bool AudioPolicyManagerBase::threadLoop()
+{
+    LOGV("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) {
+            LOGV("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"));
+
+                mpClientInterface->closeOutput(mHardwareOutput);
+                delete mOutputs.valueFor(mHardwareOutput);
+                mOutputs.removeItem(mHardwareOutput);
+
+                AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+                outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
+                mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
+                                                &outputDesc->mSamplingRate,
+                                                &outputDesc->mFormat,
+                                                &outputDesc->mChannels,
+                                                &outputDesc->mLatency,
+                                                outputDesc->mFlags);
+                if (mHardwareOutput == 0) {
+                    LOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
+                            outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
+                } else {
+                    AudioParameter outputCmd = AudioParameter();
+                    outputCmd.addInt(String8("set_id"), 0);
+                    mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
+                    addOutput(mHardwareOutput, 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);
+}
+
+
+#ifdef WITH_A2DP
+status_t AudioPolicyManagerBase::handleA2dpConnection(AudioSystem::audio_devices device,
+                                                 const char *device_address)
+{
+    // when an A2DP device is connected, open an A2DP and a duplicated output
+    LOGV("opening A2DP output for device %s", device_address);
+    AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+    outputDesc->mDevice = device;
+    mA2dpOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
+                                            &outputDesc->mSamplingRate,
+                                            &outputDesc->mFormat,
+                                            &outputDesc->mChannels,
+                                            &outputDesc->mLatency,
+                                            outputDesc->mFlags);
+    if (mA2dpOutput) {
+        // add A2DP output descriptor
+        addOutput(mA2dpOutput, outputDesc);
+        // set initial stream volume for A2DP device
+        applyStreamVolumes(mA2dpOutput, device);
+        if (a2dpUsedForSonification()) {
+            mDuplicatedOutput = mpClientInterface->openDuplicateOutput(mA2dpOutput, mHardwareOutput);
+        }
+        if (mDuplicatedOutput != 0 ||
+            !a2dpUsedForSonification()) {
+            // If both A2DP and duplicated outputs are open, send device address to A2DP hardware
+            // interface
+            AudioParameter param;
+            param.add(String8("a2dp_sink_address"), String8(device_address));
+            mpClientInterface->setParameters(mA2dpOutput, param.toString());
+            mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+
+            if (a2dpUsedForSonification()) {
+                // add duplicated output descriptor
+                AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor();
+                dupOutputDesc->mOutput1 = mOutputs.valueFor(mHardwareOutput);
+                dupOutputDesc->mOutput2 = mOutputs.valueFor(mA2dpOutput);
+                dupOutputDesc->mSamplingRate = outputDesc->mSamplingRate;
+                dupOutputDesc->mFormat = outputDesc->mFormat;
+                dupOutputDesc->mChannels = outputDesc->mChannels;
+                dupOutputDesc->mLatency = outputDesc->mLatency;
+                addOutput(mDuplicatedOutput, dupOutputDesc);
+                applyStreamVolumes(mDuplicatedOutput, device);
+            }
+        } else {
+            LOGW("getOutput() could not open duplicated output for %d and %d",
+                    mHardwareOutput, mA2dpOutput);
+            mpClientInterface->closeOutput(mA2dpOutput);
+            mOutputs.removeItem(mA2dpOutput);
+            mA2dpOutput = 0;
+            delete outputDesc;
+            return NO_INIT;
+        }
+    } else {
+        LOGW("setDeviceConnectionState() could not open A2DP output for device %x", device);
+        delete outputDesc;
+        return NO_INIT;
+    }
+    AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
+
+    if (mA2dpDeviceAddress == mScoDeviceAddress) {
+        // It is normal to suspend twice if we are both in call,
+        // and have the hardware audio output routed to BT SCO
+        if (mPhoneState != AudioSystem::MODE_NORMAL) {
+            mpClientInterface->suspendOutput(mA2dpOutput);
+        }
+        if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)hwOutputDesc->device())) {
+            mpClientInterface->suspendOutput(mA2dpOutput);
+        }
+    }
+
+    if (!a2dpUsedForSonification()) {
+        // mute music on A2DP output if a notification or ringtone is playing
+        uint32_t refCount = hwOutputDesc->strategyRefCount(STRATEGY_SONIFICATION);
+        for (uint32_t i = 0; i < refCount; i++) {
+            setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
+        }
+    }
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerBase::handleA2dpDisconnection(AudioSystem::audio_devices device,
+                                                    const char *device_address)
+{
+    if (mA2dpOutput == 0) {
+        LOGW("setDeviceConnectionState() disconnecting A2DP and no A2DP output!");
+        return INVALID_OPERATION;
+    }
+
+    if (mA2dpDeviceAddress != device_address) {
+        LOGW("setDeviceConnectionState() disconnecting unknow A2DP sink address %s", device_address);
+        return INVALID_OPERATION;
+    }
+
+    // mute media during 2 seconds to avoid outputing sound on hardware output while music stream
+    // is switched from A2DP output and before music is paused by music application
+    setStrategyMute(STRATEGY_MEDIA, true, mHardwareOutput);
+    setStrategyMute(STRATEGY_MEDIA, false, mHardwareOutput, 2000);
+
+    if (!a2dpUsedForSonification()) {
+        // unmute music on A2DP output if a notification or ringtone is playing
+        uint32_t refCount = mOutputs.valueFor(mHardwareOutput)->strategyRefCount(STRATEGY_SONIFICATION);
+        for (uint32_t i = 0; i < refCount; i++) {
+            setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput);
+        }
+    }
+    mA2dpDeviceAddress = "";
+    return NO_ERROR;
+}
+
+void AudioPolicyManagerBase::closeA2dpOutputs()
+{
+    LOGV("setDeviceConnectionState() closing A2DP and duplicated output!");
+
+    if (mDuplicatedOutput != 0) {
+        mpClientInterface->closeOutput(mDuplicatedOutput);
+        delete mOutputs.valueFor(mDuplicatedOutput);
+        mOutputs.removeItem(mDuplicatedOutput);
+        mDuplicatedOutput = 0;
+    }
+    if (mA2dpOutput != 0) {
+        AudioParameter param;
+        param.add(String8("closing"), String8("true"));
+        mpClientInterface->setParameters(mA2dpOutput, param.toString());
+        mpClientInterface->closeOutput(mA2dpOutput);
+        delete mOutputs.valueFor(mA2dpOutput);
+        mOutputs.removeItem(mA2dpOutput);
+        mA2dpOutput = 0;
+    }
+}
+
+void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy, uint32_t &newDevice)
+{
+    uint32_t prevDevice = getDeviceForStrategy(strategy);
+    uint32_t curDevice = getDeviceForStrategy(strategy, false);
+    bool a2dpWasUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(prevDevice & ~AudioSystem::DEVICE_OUT_SPEAKER));
+    bool a2dpIsUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(curDevice & ~AudioSystem::DEVICE_OUT_SPEAKER));
+    AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
+    AudioOutputDescriptor *a2dpOutputDesc;
+
+    if (a2dpWasUsed && !a2dpIsUsed) {
+        bool dupUsed = a2dpUsedForSonification() && a2dpWasUsed && (AudioSystem::popCount(prevDevice) == 2);
+
+        if (dupUsed) {
+            LOGV("checkOutputForStrategy() moving strategy %d to duplicated", strategy);
+            a2dpOutputDesc = mOutputs.valueFor(mDuplicatedOutput);
+        } else {
+            LOGV("checkOutputForStrategy() moving strategy %d to a2dp", strategy);
+            a2dpOutputDesc = mOutputs.valueFor(mA2dpOutput);
+        }
+
+        for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+            if (getStrategy((AudioSystem::stream_type)i) == strategy) {
+                mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mHardwareOutput);
+                int refCount = a2dpOutputDesc->mRefCount[i];
+                // in the case of duplicated output, the ref count is first incremented
+                // and then decremented on hardware output tus keeping its value
+                hwOutputDesc->changeRefCount((AudioSystem::stream_type)i, refCount);
+                a2dpOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
+            }
+        }
+        // do not change newDevice is it was already set before this call by a previous call to
+        // getNewDevice() or checkOutputForStrategy() for a strategy with higher priority
+        if (newDevice == 0 && hwOutputDesc->isUsedByStrategy(strategy)) {
+            newDevice = getDeviceForStrategy(strategy, false);
+        }
+    }
+    if (a2dpIsUsed && !a2dpWasUsed) {
+        bool dupUsed = a2dpUsedForSonification() && a2dpIsUsed && (AudioSystem::popCount(curDevice) == 2);
+        audio_io_handle_t a2dpOutput;
+
+        if (dupUsed) {
+            LOGV("checkOutputForStrategy() moving strategy %d from duplicated", strategy);
+            a2dpOutputDesc = mOutputs.valueFor(mDuplicatedOutput);
+            a2dpOutput = mDuplicatedOutput;
+        } else {
+            LOGV("checkOutputForStrategy() moving strategy %d from a2dp", strategy);
+            a2dpOutputDesc = mOutputs.valueFor(mA2dpOutput);
+            a2dpOutput = mA2dpOutput;
+        }
+
+        for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+            if (getStrategy((AudioSystem::stream_type)i) == strategy) {
+                mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, a2dpOutput);
+                int refCount = hwOutputDesc->mRefCount[i];
+                // in the case of duplicated output, the ref count is first incremented
+                // and then decremented on hardware output tus keeping its value
+                a2dpOutputDesc->changeRefCount((AudioSystem::stream_type)i, refCount);
+                hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
+            }
+        }
+    }
+}
+
+void AudioPolicyManagerBase::checkOutputForAllStrategies(uint32_t &newDevice)
+{
+    // Check strategies in order of priority so that once newDevice is set
+    // for a given strategy it is not modified by subsequent calls to
+    // checkOutputForStrategy()
+    checkOutputForStrategy(STRATEGY_PHONE, newDevice);
+    checkOutputForStrategy(STRATEGY_SONIFICATION, newDevice);
+    checkOutputForStrategy(STRATEGY_MEDIA, newDevice);
+    checkOutputForStrategy(STRATEGY_DTMF, newDevice);
+}
+
+#endif
+
+uint32_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache)
+{
+    uint32_t device = 0;
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    // check the following by order of priority to request a routing change if necessary:
+    // 1: we are in call or the strategy phone is active on the hardware output:
+    //      use device for strategy phone
+    // 2: the strategy sonification is active on the hardware output:
+    //      use device for strategy sonification
+    // 3: the strategy media is active on the hardware output:
+    //      use device for strategy media
+    // 4: the strategy DTMF is active on the hardware output:
+    //      use device for strategy DTMF
+    if (mPhoneState == AudioSystem::MODE_IN_CALL ||
+        outputDesc->isUsedByStrategy(STRATEGY_PHONE)) {
+        device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+    } else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) {
+        device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+    } else if (outputDesc->isUsedByStrategy(STRATEGY_MEDIA)) {
+        device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+    } else if (outputDesc->isUsedByStrategy(STRATEGY_DTMF)) {
+        device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+    }
+
+    LOGV("getNewDevice() selected device %x", device);
+    return device;
+}
+
+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::NOTIFICATION:
+    case AudioSystem::ALARM:
+    case AudioSystem::ENFORCED_AUDIBLE:
+        return STRATEGY_SONIFICATION;
+    case AudioSystem::DTMF:
+        return STRATEGY_DTMF;
+    default:
+        LOGE("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;
+    }
+}
+
+uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy, bool fromCache)
+{
+    uint32_t device = 0;
+
+    if (fromCache) {
+        LOGV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]);
+        return mDeviceForStrategy[strategy];
+    }
+
+    switch (strategy) {
+    case STRATEGY_DTMF:
+        if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+            // when off call, DTMF strategy follows the same rules as MEDIA strategy
+            device = getDeviceForStrategy(STRATEGY_MEDIA, false);
+            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 (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
+                device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+            if (device) break;
+            device = mAvailableOutputDevices & AudioSystem::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
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
+            if (device) break;
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
+            if (device) break;
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_EARPIECE;
+            if (device == 0) {
+                LOGE("getDeviceForStrategy() earpiece device not found");
+            }
+            break;
+
+        case AudioSystem::FORCE_SPEAKER:
+            if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
+                device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
+            if (device == 0) {
+                LOGE("getDeviceForStrategy() speaker device not found");
+            }
+            break;
+        }
+    break;
+
+    case STRATEGY_SONIFICATION:
+
+        // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
+        // handleIncallSonification().
+        if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+            device = getDeviceForStrategy(STRATEGY_PHONE, false);
+            break;
+        }
+        device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
+        if (device == 0) {
+            LOGE("getDeviceForStrategy() speaker device not found");
+        }
+        // The second device used for sonification is the same as the device used by media strategy
+        // FALL THROUGH
+
+    case STRATEGY_MEDIA: {
+        uint32_t device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
+#ifdef WITH_A2DP
+        if (mA2dpOutput != 0) {
+            if (strategy == STRATEGY_SONIFICATION && !a2dpUsedForSonification()) {
+                break;
+            }
+            if (device2 == 0) {
+                device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
+            }
+            if (device2 == 0) {
+                device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+            }
+            if (device2 == 0) {
+                device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+            }
+        }
+#endif
+        if (device2 == 0) {
+            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
+        }
+        if (device2 == 0) {
+            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
+        }
+        if (device2 == 0) {
+            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
+        }
+
+        // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION, 0 otherwise
+        device |= device2;
+        if (device == 0) {
+            LOGE("getDeviceForStrategy() speaker device not found");
+        }
+        } break;
+
+    default:
+        LOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+        break;
+    }
+
+    LOGV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
+    return device;
+}
+
+void AudioPolicyManagerBase::updateDeviceForStrategy()
+{
+    for (int i = 0; i < NUM_STRATEGIES; i++) {
+        mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false);
+    }
+}
+
+void AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output, uint32_t device, bool force, int delayMs)
+{
+    LOGV("setOutputDevice() output %d device %x delayMs %d", output, device, delayMs);
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+
+
+    if (outputDesc->isDuplicated()) {
+        setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
+        setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
+        return;
+    }
+#ifdef WITH_A2DP
+    // filter devices according to output selected
+    if (output == mHardwareOutput) {
+        device &= ~AudioSystem::DEVICE_OUT_ALL_A2DP;
+    } else {
+        device &= AudioSystem::DEVICE_OUT_ALL_A2DP;
+    }
+#endif
+
+    uint32_t prevDevice = (uint32_t)outputDesc->device();
+    // Do not change the routing if:
+    //  - the requestede device is 0
+    //  - 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 == 0 ||
+        (device == prevDevice && !force)) {
+        LOGV("setOutputDevice() setting same device %x or null device for output %d", device, output);
+        return;
+    }
+
+    outputDesc->mDevice = device;
+    // mute media streams if both speaker and headset are selected
+    if (output == mHardwareOutput && AudioSystem::popCount(device) == 2) {
+        setStrategyMute(STRATEGY_MEDIA, true, output);
+        // wait for the PCM output buffers to empty before proceeding with the rest of the command
+        usleep(outputDesc->mLatency*2*1000);
+    }
+#ifdef WITH_A2DP
+    // suspend A2D output if SCO device is selected
+    if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)device)) {
+         if (mA2dpOutput && mScoDeviceAddress == mA2dpDeviceAddress) {
+             mpClientInterface->suspendOutput(mA2dpOutput);
+         }
+    }
+#endif
+    // do the routing
+    AudioParameter param = AudioParameter();
+    param.addInt(String8(AudioParameter::keyRouting), (int)device);
+    mpClientInterface->setParameters(mHardwareOutput, param.toString(), delayMs);
+    // update stream volumes according to new device
+    applyStreamVolumes(output, device, delayMs);
+
+#ifdef WITH_A2DP
+    // if disconnecting SCO device, restore A2DP output
+    if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)prevDevice)) {
+         if (mA2dpOutput && mScoDeviceAddress == mA2dpDeviceAddress) {
+             LOGV("restore A2DP output");
+             mpClientInterface->restoreOutput(mA2dpOutput);
+         }
+    }
+#endif
+    // if changing from a combined headset + speaker route, unmute media streams
+    if (output == mHardwareOutput && AudioSystem::popCount(prevDevice) == 2) {
+        setStrategyMute(STRATEGY_MEDIA, false, output, delayMs);
+    }
+}
+
+uint32_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
+{
+    uint32_t device;
+
+    switch(inputSource) {
+    case AUDIO_SOURCE_DEFAULT:
+    case AUDIO_SOURCE_MIC:
+    case AUDIO_SOURCE_VOICE_RECOGNITION:
+        if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
+            mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+            device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+        } else if (mAvailableInputDevices & AudioSystem::DEVICE_IN_WIRED_HEADSET) {
+            device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
+        } else {
+            device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+        }
+        break;
+    case AUDIO_SOURCE_CAMCORDER:
+        if (hasBackMicrophone()) {
+            device = AudioSystem::DEVICE_IN_BACK_MIC;
+        } else {
+            device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+        }
+        break;
+    case AUDIO_SOURCE_VOICE_UPLINK:
+    case AUDIO_SOURCE_VOICE_DOWNLINK:
+    case AUDIO_SOURCE_VOICE_CALL:
+        device = AudioSystem::DEVICE_IN_VOICE_CALL;
+        break;
+    default:
+        LOGW("getInput() invalid input source %d", inputSource);
+        device = 0;
+        break;
+    }
+    LOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+    return device;
+}
+
+audio_io_handle_t AudioPolicyManagerBase::getActiveInput()
+{
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        if (mInputs.valueAt(i)->mRefCount > 0) {
+            return mInputs.keyAt(i);
+        }
+    }
+    return 0;
+}
+
+float AudioPolicyManagerBase::computeVolume(int stream, int index, audio_io_handle_t output, uint32_t device)
+{
+    float volume = 1.0;
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    StreamDescriptor &streamDesc = mStreams[stream];
+
+    if (device == 0) {
+        device = outputDesc->device();
+    }
+
+    int volInt = (100 * (index - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin);
+    volume = AudioSystem::linearToLog(volInt);
+
+    // if a heaset 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.
+    if ((device &
+        (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
+        AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+        AudioSystem::DEVICE_OUT_WIRED_HEADSET |
+        AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) &&
+        (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) &&
+        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 (outputDesc->mRefCount[AudioSystem::MUSIC] || mLimitRingtoneVolume) {
+            float musicVol = computeVolume(AudioSystem::MUSIC, mStreams[AudioSystem::MUSIC].mIndexCur, output, device);
+            float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
+            if (volume > minVol) {
+                volume = minVol;
+                LOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
+            }
+        }
+    }
+
+    return volume;
+}
+
+status_t AudioPolicyManagerBase::checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs, bool force)
+{
+
+    // do not change actual stream volume if the stream is muted
+    if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
+        LOGV("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)) {
+        LOGV("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);
+    // do not set volume if the float value did not change
+    if (volume != mOutputs.valueFor(output)->mCurVolume[stream] || force) {
+        mOutputs.valueFor(output)->mCurVolume[stream] = volume;
+        LOGV("setStreamVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
+        if (stream == AudioSystem::VOICE_CALL ||
+            stream == AudioSystem::DTMF ||
+            stream == AudioSystem::BLUETOOTH_SCO) {
+            float voiceVolume = -1.0;
+            // offset value to reflect actual hardware volume that never reaches 0
+            // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
+            volume = 0.01 + 0.99 * volume;
+            if (stream == AudioSystem::VOICE_CALL) {
+                voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+            } else if (stream == AudioSystem::BLUETOOTH_SCO) {
+                voiceVolume = 1.0;
+            }
+            if (voiceVolume >= 0 && output == mHardwareOutput) {
+                mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+            }
+        }
+        mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
+    }
+
+    return NO_ERROR;
+}
+
+void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs)
+{
+    LOGV("applyStreamVolumes() for output %d and device %x", output, device);
+
+    for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+        checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, device, delayMs);
+    }
+}
+
+void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs)
+{
+    LOGV("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);
+        }
+    }
+}
+
+void AudioPolicyManagerBase::setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs)
+{
+    StreamDescriptor &streamDesc = mStreams[stream];
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+
+    LOGV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d", stream, on, output, outputDesc->mMuteCount[stream]);
+
+    if (on) {
+        if (outputDesc->mMuteCount[stream] == 0) {
+            if (streamDesc.mCanBeMuted) {
+                checkAndSetVolume(stream, 0, output, outputDesc->device(), delayMs);
+            }
+        }
+        // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
+        outputDesc->mMuteCount[stream]++;
+    } else {
+        if (outputDesc->mMuteCount[stream] == 0) {
+            LOGW("setStreamMute() unmuting non muted stream!");
+            return;
+        }
+        if (--outputDesc->mMuteCount[stream] == 0) {
+            checkAndSetVolume(stream, streamDesc.mIndexCur, output, outputDesc->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
+
+    if (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput);
+        LOGV("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)) {
+                LOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
+                for (int i = 0; i < muteCount; i++) {
+                    setStreamMute(stream, starting, mHardwareOutput);
+                }
+            } else {
+                LOGV("handleIncallSonification() high visibility");
+                if (outputDesc->device() & getDeviceForStrategy(STRATEGY_PHONE)) {
+                    LOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
+                    for (int i = 0; i < muteCount; i++) {
+                        setStreamMute(stream, starting, mHardwareOutput);
+                    }
+                }
+                if (starting) {
+                    mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
+                } else {
+                    mpClientInterface->stopTone();
+                }
+            }
+        }
+    }
+}
+
+// --- AudioOutputDescriptor class implementation
+
+AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor()
+    : mId(0), mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0),
+    mFlags((AudioSystem::output_flags)0), mDevice(0), mOutput1(0), mOutput2(0)
+{
+    // 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;
+    }
+}
+
+uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::device()
+{
+    uint32_t device = 0;
+    if (isDuplicated()) {
+        device = mOutput1->mDevice | mOutput2->mDevice;
+    } else {
+        device = mDevice;
+    }
+    return device;
+}
+
+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) {
+        LOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
+        mRefCount[stream] = 0;
+        return;
+    }
+    mRefCount[stream] += delta;
+    LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::refCount()
+{
+    uint32_t refcount = 0;
+    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+        refcount += mRefCount[i];
+    }
+    return refcount;
+}
+
+uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::strategyRefCount(routing_strategy strategy)
+{
+    uint32_t refCount = 0;
+    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+        if (getStrategy((AudioSystem::stream_type)i) == strategy) {
+            refCount += mRefCount[i];
+        }
+    }
+    return refCount;
+}
+
+
+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: %d\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Channels: %08x\n", mChannels);
+    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()
+    : mSamplingRate(0), mFormat(0), mChannels(0),
+     mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(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", mChannels);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Acoustics %08x\n", mAcoustics);
+    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
+
+void AudioPolicyManagerBase::StreamDescriptor::dump(char* buffer, size_t size)
+{
+    snprintf(buffer, size, "      %02d         %02d         %02d         %d\n",
+            mIndexMin,
+            mIndexMax,
+            mIndexCur,
+            mCanBeMuted);
+}
+
+
+}; // namespace android
diff --git a/libs/audioflinger/AudioPolicyManagerGeneric.cpp b/libs/audioflinger/AudioPolicyManagerGeneric.cpp
deleted file mode 100644
index 8cfc204..0000000
--- a/libs/audioflinger/AudioPolicyManagerGeneric.cpp
+++ /dev/null
@@ -1,945 +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 "AudioPolicyManagerGeneric"
-//#define LOG_NDEBUG 0
-#include <utils/Log.h>
-#include "AudioPolicyManagerGeneric.h"
-#include <media/mediarecorder.h>
-
-namespace android {
-
-
-// ----------------------------------------------------------------------------
-// AudioPolicyInterface implementation
-// ----------------------------------------------------------------------------
-
-
-status_t AudioPolicyManagerGeneric::setDeviceConnectionState(AudioSystem::audio_devices device,
-                                                  AudioSystem::device_connection_state state,
-                                                  const char *device_address)
-{
-
-    LOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
-
-    // connect/disconnect only 1 device at a time
-    if (AudioSystem::popCount(device) != 1) return BAD_VALUE;
-
-    if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
-        LOGE("setDeviceConnectionState() invalid address: %s", device_address);
-        return BAD_VALUE;
-    }
-
-    // handle output devices
-    if (AudioSystem::isOutputDevice(device)) {
-        switch (state)
-        {
-        // handle output device connection
-        case AudioSystem::DEVICE_STATE_AVAILABLE:
-            if (mAvailableOutputDevices & device) {
-                LOGW("setDeviceConnectionState() device already connected: %x", device);
-                return INVALID_OPERATION;
-            }
-            LOGV("setDeviceConnectionState() connecting device %x", device);
-
-            // register new device as available
-            mAvailableOutputDevices |= device;
-            break;
-        // handle output device disconnection
-        case AudioSystem::DEVICE_STATE_UNAVAILABLE:
-            if (!(mAvailableOutputDevices & device)) {
-                LOGW("setDeviceConnectionState() device not connected: %x", device);
-                return INVALID_OPERATION;
-            }
-            LOGV("setDeviceConnectionState() disconnecting device %x", device);
-            // remove device from available output devices
-            mAvailableOutputDevices &= ~device;
-            break;
-
-        default:
-            LOGE("setDeviceConnectionState() invalid state: %x", state);
-            return BAD_VALUE;
-        }
-        return NO_ERROR;
-    }
-    // handle input devices
-    if (AudioSystem::isInputDevice(device)) {
-        switch (state)
-        {
-        // handle input device connection
-        case AudioSystem::DEVICE_STATE_AVAILABLE:
-            if (mAvailableInputDevices & device) {
-                LOGW("setDeviceConnectionState() device already connected: %d", device);
-                return INVALID_OPERATION;
-            }
-            mAvailableInputDevices |= device;
-            break;
-
-        // handle input device disconnection
-        case AudioSystem::DEVICE_STATE_UNAVAILABLE:
-            if (!(mAvailableInputDevices & device)) {
-                LOGW("setDeviceConnectionState() device not connected: %d", device);
-                return INVALID_OPERATION;
-            }
-            mAvailableInputDevices &= ~device;
-            break;
-
-        default:
-            LOGE("setDeviceConnectionState() invalid state: %x", state);
-            return BAD_VALUE;
-        }
-        return NO_ERROR;
-    }
-
-    LOGW("setDeviceConnectionState() invalid device: %x", device);
-    return BAD_VALUE;
-}
-
-AudioSystem::device_connection_state AudioPolicyManagerGeneric::getDeviceConnectionState(AudioSystem::audio_devices device,
-                                                  const char *device_address)
-{
-    AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
-    String8 address = String8(device_address);
-    if (AudioSystem::isOutputDevice(device)) {
-        if (device & mAvailableOutputDevices) {
-            state = AudioSystem::DEVICE_STATE_AVAILABLE;
-        }
-    } else if (AudioSystem::isInputDevice(device)) {
-        if (device & mAvailableInputDevices) {
-            state = AudioSystem::DEVICE_STATE_AVAILABLE;
-        }
-    }
-
-    return state;
-}
-
-void AudioPolicyManagerGeneric::setPhoneState(int state)
-{
-    LOGV("setPhoneState() state %d", state);
-    uint32_t newDevice = 0;
-    if (state < 0 || state >= AudioSystem::NUM_MODES) {
-        LOGW("setPhoneState() invalid state %d", state);
-        return;
-    }
-
-    if (state == mPhoneState ) {
-        LOGW("setPhoneState() setting same state %d", state);
-        return;
-    }
-    // store previous phone state for management of sonification strategy below
-    int oldState = mPhoneState;
-    mPhoneState = state;
-
-    // if leaving or entering in call state, handle special case of active streams
-    // pertaining to sonification strategy see handleIncallSonification()
-    if (state == AudioSystem::MODE_IN_CALL ||
-        oldState == AudioSystem::MODE_IN_CALL) {
-        bool starting = (state == AudioSystem::MODE_IN_CALL) ? true : false;
-        LOGV("setPhoneState() in call state management: new state is %d", state);
-        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
-            handleIncallSonification(stream, starting);
-        }
-    }
-}
-
-void AudioPolicyManagerGeneric::setRingerMode(uint32_t mode, uint32_t mask)
-{
-    LOGV("setRingerMode() mode %x, mask %x", mode, mask);
-
-    mRingerMode = mode;
-}
-
-void AudioPolicyManagerGeneric::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
-{
-    LOGV("setForceUse) usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
-    mForceUse[usage] = config;
-}
-
-AudioSystem::forced_config AudioPolicyManagerGeneric::getForceUse(AudioSystem::force_use usage)
-{
-    return mForceUse[usage];
-}
-
-void AudioPolicyManagerGeneric::setSystemProperty(const char* property, const char* value)
-{
-    LOGV("setSystemProperty() property %s, value %s", property, value);
-    if (strcmp(property, "ro.camera.sound.forced") == 0) {
-        if (atoi(value)) {
-            LOGV("ENFORCED_AUDIBLE cannot be muted");
-            mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false;
-        } else {
-            LOGV("ENFORCED_AUDIBLE can be muted");
-            mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true;
-        }
-    }
-}
-
-audio_io_handle_t AudioPolicyManagerGeneric::getOutput(AudioSystem::stream_type stream,
-                                    uint32_t samplingRate,
-                                    uint32_t format,
-                                    uint32_t channels,
-                                    AudioSystem::output_flags flags)
-{
-    LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags);
-
-#ifdef AUDIO_POLICY_TEST
-    if (mCurOutput != 0) {
-        LOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channels %x, mDirectOutput %d",
-                mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
-
-        if (mTestOutputs[mCurOutput] == 0) {
-            LOGV("getOutput() opening test output");
-            AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
-            outputDesc->mDevice = mTestDevice;
-            outputDesc->mSamplingRate = mTestSamplingRate;
-            outputDesc->mFormat = mTestFormat;
-            outputDesc->mChannels = mTestChannels;
-            outputDesc->mLatency = mTestLatencyMs;
-            outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
-            outputDesc->mRefCount[stream] = 0;
-            mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice,
-                                            &outputDesc->mSamplingRate,
-                                            &outputDesc->mFormat,
-                                            &outputDesc->mChannels,
-                                            &outputDesc->mLatency,
-                                            outputDesc->mFlags);
-            if (mTestOutputs[mCurOutput]) {
-                AudioParameter outputCmd = AudioParameter();
-                outputCmd.addInt(String8("set_id"),mCurOutput);
-                mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
-                mOutputs.add(mTestOutputs[mCurOutput], outputDesc);
-            }
-        }
-        return mTestOutputs[mCurOutput];
-    }
-#endif //AUDIO_POLICY_TEST
-    if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
-        (format != 0 && !AudioSystem::isLinearPCM(format)) ||
-        (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO && channels != AudioSystem::CHANNEL_OUT_STEREO)) {
-        return 0;
-    }
-
-    return mHardwareOutput;
-}
-
-status_t AudioPolicyManagerGeneric::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
-{
-    LOGV("startOutput() output %d, stream %d", output, stream);
-    ssize_t index = mOutputs.indexOfKey(output);
-    if (index < 0) {
-        LOGW("startOutput() unknow output %d", output);
-        return BAD_VALUE;
-    }
-
-    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-
-    // handle special case for sonification while in call
-    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
-        handleIncallSonification(stream, true);
-    }
-
-    // incremenent usage count for this stream on the requested output:
-    outputDesc->changeRefCount(stream, 1);
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerGeneric::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
-{
-    LOGV("stopOutput() output %d, stream %d", output, stream);
-    ssize_t index = mOutputs.indexOfKey(output);
-    if (index < 0) {
-        LOGW("stopOutput() unknow output %d", output);
-        return BAD_VALUE;
-    }
-
-    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
-
-    // handle special case for sonification while in call
-    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
-        handleIncallSonification(stream, false);
-    }
-
-    if (outputDesc->isUsedByStream(stream)) {
-        // decrement usage count of this stream on the output
-        outputDesc->changeRefCount(stream, -1);
-        return NO_ERROR;
-    } else {
-        LOGW("stopOutput() refcount is already 0 for output %d", output);
-        return INVALID_OPERATION;
-    }
-}
-
-void AudioPolicyManagerGeneric::releaseOutput(audio_io_handle_t output)
-{
-    LOGV("releaseOutput() %d", output);
-    ssize_t index = mOutputs.indexOfKey(output);
-    if (index < 0) {
-        LOGW("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->refCount() == 0) {
-            mpClientInterface->closeOutput(output);
-            delete mOutputs.valueAt(index);
-            mOutputs.removeItem(output);
-            mTestOutputs[testIndex] = 0;
-        }
-    }
-#endif //AUDIO_POLICY_TEST
-}
-
-audio_io_handle_t AudioPolicyManagerGeneric::getInput(int inputSource,
-                                    uint32_t samplingRate,
-                                    uint32_t format,
-                                    uint32_t channels,
-                                    AudioSystem::audio_in_acoustics acoustics)
-{
-    audio_io_handle_t input = 0;
-    uint32_t device;
-
-    LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics);
-
-    AudioInputDescriptor *inputDesc = new AudioInputDescriptor();
-    inputDesc->mDevice = AudioSystem::DEVICE_IN_BUILTIN_MIC;
-    inputDesc->mSamplingRate = samplingRate;
-    inputDesc->mFormat = format;
-    inputDesc->mChannels = channels;
-    inputDesc->mAcoustics = acoustics;
-    inputDesc->mRefCount = 0;
-    input = mpClientInterface->openInput(&inputDesc->mDevice,
-                                    &inputDesc->mSamplingRate,
-                                    &inputDesc->mFormat,
-                                    &inputDesc->mChannels,
-                                    inputDesc->mAcoustics);
-
-    // only accept input with the exact requested set of parameters
-    if ((samplingRate != inputDesc->mSamplingRate) ||
-        (format != inputDesc->mFormat) ||
-        (channels != inputDesc->mChannels)) {
-        LOGV("getOutput() failed opening input: samplingRate %d, format %d, channels %d",
-                samplingRate, format, channels);
-        mpClientInterface->closeInput(input);
-        delete inputDesc;
-        return 0;
-    }
-    mInputs.add(input, inputDesc);
-    return input;
-}
-
-status_t AudioPolicyManagerGeneric::startInput(audio_io_handle_t input)
-{
-    LOGV("startInput() input %d", input);
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        LOGW("startInput() unknow 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
-        for (size_t i = 0; i < mInputs.size(); i++) {
-            if (mInputs.valueAt(i)->mRefCount > 0) {
-                LOGW("startInput() input %d, other input %d already started", input, mInputs.keyAt(i));
-                return INVALID_OPERATION;
-            }
-        }
-    }
-
-    inputDesc->mRefCount = 1;
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerGeneric::stopInput(audio_io_handle_t input)
-{
-    LOGV("stopInput() input %d", input);
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        LOGW("stopInput() unknow input %d", input);
-        return BAD_VALUE;
-    }
-    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
-    if (inputDesc->mRefCount == 0) {
-        LOGW("stopInput() input %d already stopped", input);
-        return INVALID_OPERATION;
-    } else {
-        inputDesc->mRefCount = 0;
-        return NO_ERROR;
-    }
-}
-
-void AudioPolicyManagerGeneric::releaseInput(audio_io_handle_t input)
-{
-    LOGV("releaseInput() %d", input);
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        LOGW("releaseInput() releasing unknown input %d", input);
-        return;
-    }
-    mpClientInterface->closeInput(input);
-    delete mInputs.valueAt(index);
-    mInputs.removeItem(input);
-}
-
-
-
-void AudioPolicyManagerGeneric::initStreamVolume(AudioSystem::stream_type stream,
-                                            int indexMin,
-                                            int indexMax)
-{
-    LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
-    mStreams[stream].mIndexMin = indexMin;
-    mStreams[stream].mIndexMax = indexMax;
-}
-
-status_t AudioPolicyManagerGeneric::setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
-{
-
-    if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
-        return BAD_VALUE;
-    }
-
-    LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index);
-    mStreams[stream].mIndexCur = index;
-
-    // do not change actual stream volume if the stream is muted
-    if (mStreams[stream].mMuteCount != 0) {
-        return NO_ERROR;
-    }
-
-    // Do not changed 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)) {
-        LOGV("setStreamVolumeIndex() cannot set stream %d volume with force use = %d for comm",
-             stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
-        return INVALID_OPERATION;
-    }
-
-    // compute and apply stream volume on all outputs according to connected device
-    for (size_t i = 0; i < mOutputs.size(); i++) {
-        AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
-        uint32_t device = outputDesc->device();
-
-        float volume = computeVolume((int)stream, index, device);
-
-        LOGV("setStreamVolume() for output %d stream %d, volume %f", mOutputs.keyAt(i), stream, volume);
-        mpClientInterface->setStreamVolume(stream, volume, mOutputs.keyAt(i));
-    }
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerGeneric::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
-{
-    if (index == 0) {
-        return BAD_VALUE;
-    }
-    LOGV("getStreamVolumeIndex() stream %d", stream);
-    *index =  mStreams[stream].mIndexCur;
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerGeneric::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, " Hardware Output: %d\n", mHardwareOutput);
-    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, " Ringer mode: %d\n", mRingerMode);
-    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);
-    write(fd, result.string(), result.size());
-
-    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  Index Min  Index Max  Index Cur  Mute Count  Can be muted\n");
-    write(fd, buffer, strlen(buffer));
-    for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
-        snprintf(buffer, SIZE, " %02d", i);
-        mStreams[i].dump(buffer + 3, SIZE);
-        write(fd, buffer, strlen(buffer));
-    }
-
-    return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManagerGeneric
-// ----------------------------------------------------------------------------
-
-// ---  class factory
-
-AudioPolicyManagerGeneric::AudioPolicyManagerGeneric(AudioPolicyClientInterface *clientInterface)
-    :
-#ifdef AUDIO_POLICY_TEST
-    Thread(false),
-#endif //AUDIO_POLICY_TEST
-    mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0)
-{
-    mpClientInterface = clientInterface;
-
-    for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
-        mForceUse[i] = AudioSystem::FORCE_NONE;
-    }
-
-    // devices available by default are speaker, ear piece and microphone
-    mAvailableOutputDevices = AudioSystem::DEVICE_OUT_SPEAKER;
-    mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
-
-    // open hardware output
-    AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
-    outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
-    mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
-                                    &outputDesc->mSamplingRate,
-                                    &outputDesc->mFormat,
-                                    &outputDesc->mChannels,
-                                    &outputDesc->mLatency,
-                                    outputDesc->mFlags);
-
-    if (mHardwareOutput == 0) {
-        LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d",
-                outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
-    } else {
-        mOutputs.add(mHardwareOutput, outputDesc);
-    }
-
-#ifdef AUDIO_POLICY_TEST
-    AudioParameter outputCmd = AudioParameter();
-    outputCmd.addInt(String8("set_id"), 0);
-    mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
-
-    mTestDevice = AudioSystem::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
-}
-
-AudioPolicyManagerGeneric::~AudioPolicyManagerGeneric()
-{
-#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);
-   }
-   mOutputs.clear();
-   for (size_t i = 0; i < mInputs.size(); i++) {
-        mpClientInterface->closeInput(mInputs.keyAt(i));
-        delete mInputs.valueAt(i);
-   }
-   mInputs.clear();
-}
-
-#ifdef AUDIO_POLICY_TEST
-bool AudioPolicyManagerGeneric::threadLoop()
-{
-    LOGV("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) {
-            LOGV("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"));
-
-                mpClientInterface->closeOutput(mHardwareOutput);
-                delete mOutputs.valueFor(mHardwareOutput);
-                mOutputs.removeItem(mHardwareOutput);
-
-                AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
-                outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
-                mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
-                                                &outputDesc->mSamplingRate,
-                                                &outputDesc->mFormat,
-                                                &outputDesc->mChannels,
-                                                &outputDesc->mLatency,
-                                                outputDesc->mFlags);
-                if (mHardwareOutput == 0) {
-                    LOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
-                            outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
-                } else {
-                    AudioParameter outputCmd = AudioParameter();
-                    outputCmd.addInt(String8("set_id"), 0);
-                    mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
-                    mOutputs.add(mHardwareOutput, outputDesc);
-                }
-            }
-
-
-            mpClientInterface->setParameters(0, String8("test_cmd_policy="));
-        }
-    }
-    return false;
-}
-
-void AudioPolicyManagerGeneric::exit()
-{
-    {
-        AutoMutex _l(mLock);
-        requestExit();
-        mWaitWorkCV.signal();
-    }
-    requestExitAndWait();
-}
-
-int AudioPolicyManagerGeneric::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
-
-// ---
-
-AudioPolicyManagerGeneric::routing_strategy AudioPolicyManagerGeneric::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::NOTIFICATION:
-    case AudioSystem::ALARM:
-    case AudioSystem::ENFORCED_AUDIBLE:
-        return STRATEGY_SONIFICATION;
-    case AudioSystem::DTMF:
-        return STRATEGY_DTMF;
-    default:
-        LOGE("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;
-    }
-}
-
-
-float AudioPolicyManagerGeneric::computeVolume(int stream, int index, uint32_t device)
-{
-    float volume = 1.0;
-
-    StreamDescriptor &streamDesc = mStreams[stream];
-
-    // Force max volume if stream cannot be muted
-    if (!streamDesc.mCanBeMuted) index = streamDesc.mIndexMax;
-
-    int volInt = (100 * (index - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin);
-    volume = AudioSystem::linearToLog(volInt);
-
-    return volume;
-}
-
-void AudioPolicyManagerGeneric::setStreamMute(int stream, bool on, audio_io_handle_t output)
-{
-    LOGV("setStreamMute() stream %d, mute %d, output %d", stream, on, output);
-
-    StreamDescriptor &streamDesc = mStreams[stream];
-
-    if (on) {
-        if (streamDesc.mMuteCount++ == 0) {
-            if (streamDesc.mCanBeMuted) {
-                mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, 0, output);
-            }
-        }
-    } else {
-        if (streamDesc.mMuteCount == 0) {
-            LOGW("setStreamMute() unmuting non muted stream!");
-            return;
-        }
-        if (--streamDesc.mMuteCount == 0) {
-            uint32_t device = mOutputs.valueFor(output)->mDevice;
-            float volume = computeVolume(stream, streamDesc.mIndexCur, device);
-            mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output);
-        }
-    }
-}
-
-void AudioPolicyManagerGeneric::handleIncallSonification(int stream, bool starting)
-{
-    // 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 (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
-        AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput);
-        LOGV("handleIncallSonification() stream %d starting %d device %x", stream, starting, outputDesc->mDevice);
-        if (outputDesc->isUsedByStream((AudioSystem::stream_type)stream)) {
-            if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
-                LOGV("handleIncallSonification() low visibility");
-                setStreamMute(stream, starting, mHardwareOutput);
-            } else {
-                if (starting) {
-                    mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
-                } else {
-                    mpClientInterface->stopTone();
-                }
-            }
-        }
-    }
-}
-
-
-// --- AudioOutputDescriptor class implementation
-
-AudioPolicyManagerGeneric::AudioOutputDescriptor::AudioOutputDescriptor()
-    : mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0),
-    mFlags((AudioSystem::output_flags)0), mDevice(0)
-{
-    // clear usage count for all stream types
-    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
-        mRefCount[i] = 0;
-    }
-}
-
-uint32_t AudioPolicyManagerGeneric::AudioOutputDescriptor::device()
-{
-    return mDevice;
-}
-
-void AudioPolicyManagerGeneric::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type stream, int delta)
-{
-    if ((delta + (int)mRefCount[stream]) < 0) {
-        LOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
-        mRefCount[stream] = 0;
-        return;
-    }
-    mRefCount[stream] += delta;
-    LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
-}
-
-uint32_t AudioPolicyManagerGeneric::AudioOutputDescriptor::refCount()
-{
-    uint32_t refcount = 0;
-    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
-        refcount += mRefCount[i];
-    }
-    return refcount;
-}
-
-status_t AudioPolicyManagerGeneric::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: %d\n", mFormat);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Channels: %08x\n", mChannels);
-    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", mDevice);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Stream refCount\n");
-    result.append(buffer);
-    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
-        snprintf(buffer, SIZE, " %02d     %d\n", i, mRefCount[i]);
-        result.append(buffer);
-    }
-    write(fd, result.string(), result.size());
-
-    return NO_ERROR;
-}
-
-// --- AudioInputDescriptor class implementation
-
-AudioPolicyManagerGeneric::AudioInputDescriptor::AudioInputDescriptor()
-    : mSamplingRate(0), mFormat(0), mChannels(0),
-     mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0)
-{
-}
-
-status_t AudioPolicyManagerGeneric::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", mChannels);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Acoustics %08x\n", mAcoustics);
-    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
-
-void AudioPolicyManagerGeneric::StreamDescriptor::dump(char* buffer, size_t size)
-{
-    snprintf(buffer, size, "      %02d         %02d         %02d         %02d          %d\n",
-            mIndexMin,
-            mIndexMax,
-            mIndexCur,
-            mMuteCount,
-            mCanBeMuted);
-}
-
-}; // namespace android
diff --git a/libs/audioflinger/AudioPolicyManagerGeneric.h b/libs/audioflinger/AudioPolicyManagerGeneric.h
deleted file mode 100644
index 4997cdf..0000000
--- a/libs/audioflinger/AudioPolicyManagerGeneric.h
+++ /dev/null
@@ -1,196 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <utils/Errors.h>
-#include <utils/KeyedVector.h>
-#include <hardware_legacy/AudioPolicyInterface.h>
-#include <utils/threads.h>
-
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-#define MAX_DEVICE_ADDRESS_LEN 20
-#define NUM_TEST_OUTPUTS 5
-
-class AudioPolicyManagerGeneric: public AudioPolicyInterface
-#ifdef AUDIO_POLICY_TEST
-    , public Thread
-#endif //AUDIO_POLICY_TEST
-{
-
-public:
-                AudioPolicyManagerGeneric(AudioPolicyClientInterface *clientInterface);
-        virtual ~AudioPolicyManagerGeneric();
-
-        // AudioPolicyInterface
-        virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
-                                                          AudioSystem::device_connection_state state,
-                                                          const char *device_address);
-        virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
-                                                                              const char *device_address);
-        virtual void setPhoneState(int state);
-        virtual void setRingerMode(uint32_t mode, uint32_t mask);
-        virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
-        virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage);
-        virtual void setSystemProperty(const char* property, const char* value);
-        virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
-                                            uint32_t samplingRate,
-                                            uint32_t format,
-                                            uint32_t channels,
-                                            AudioSystem::output_flags flags);
-        virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
-        virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
-        virtual void releaseOutput(audio_io_handle_t output);
-        virtual audio_io_handle_t getInput(int inputSource,
-                                            uint32_t samplingRate,
-                                            uint32_t format,
-                                            uint32_t channels,
-                                            AudioSystem::audio_in_acoustics acoustics);
-        // indicates to the audio policy manager that the input starts being used.
-        virtual status_t startInput(audio_io_handle_t input);
-        // indicates to the audio policy manager that the input stops being used.
-        virtual status_t stopInput(audio_io_handle_t input);
-        virtual void releaseInput(audio_io_handle_t input);
-        virtual void initStreamVolume(AudioSystem::stream_type stream,
-                                                    int indexMin,
-                                                    int indexMax);
-        virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index);
-        virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index);
-
-        virtual status_t dump(int fd);
-
-private:
-
-        enum routing_strategy {
-            STRATEGY_MEDIA,
-            STRATEGY_PHONE,
-            STRATEGY_SONIFICATION,
-            STRATEGY_DTMF,
-            NUM_STRATEGIES
-        };
-
-        // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
-        // and keep track of the usage of this output by each audio stream type.
-        class AudioOutputDescriptor
-        {
-        public:
-            AudioOutputDescriptor();
-
-            status_t    dump(int fd);
-
-            uint32_t device();
-            void changeRefCount(AudioSystem::stream_type, int delta);
-            bool isUsedByStream(AudioSystem::stream_type stream) { return mRefCount[stream] > 0 ? true : false; }
-            uint32_t refCount();
-
-            uint32_t mSamplingRate;             //
-            uint32_t mFormat;                   //
-            uint32_t mChannels;                 // output configuration
-            uint32_t mLatency;                  //
-            AudioSystem::output_flags mFlags;   //
-            uint32_t mDevice;                   // current device this output is routed to
-            uint32_t mRefCount[AudioSystem::NUM_STREAM_TYPES]; // number of streams of each type using this output
-        };
-
-        // descriptor for audio inputs. Used to maintain current configuration of each opened audio input
-        // and keep track of the usage of this input.
-        class AudioInputDescriptor
-        {
-        public:
-            AudioInputDescriptor();
-
-            status_t    dump(int fd);
-
-            uint32_t mSamplingRate;                     //
-            uint32_t mFormat;                           // input configuration
-            uint32_t mChannels;                         //
-            AudioSystem::audio_in_acoustics mAcoustics; //
-            uint32_t mDevice;                           // current device this input is routed to
-            uint32_t mRefCount;                         // number of AudioRecord clients using this output
-        };
-
-        // stream descriptor used for volume control
-        class StreamDescriptor
-        {
-        public:
-            StreamDescriptor()
-            :   mIndexMin(0), mIndexMax(1), mIndexCur(1), mMuteCount(0), mCanBeMuted(true) {}
-
-            void dump(char* buffer, size_t size);
-
-            int mIndexMin;      // min volume index
-            int mIndexMax;      // max volume index
-            int mIndexCur;      // current volume index
-            int mMuteCount;     // mute request counter
-            bool mCanBeMuted;   // true is the stream can be muted
-        };
-
-        // return the strategy corresponding to a given stream type
-        static routing_strategy getStrategy(AudioSystem::stream_type stream);
-        // return the output handle of an output routed to the specified device, 0 if no output
-        // is routed to the device
-        float computeVolume(int stream, int index, uint32_t device);
-        // Mute or unmute the stream on the specified output
-        void setStreamMute(int stream, bool on, audio_io_handle_t output);
-        // handle special cases for sonification strategy while in call: mute streams or replace by
-        // a special tone in the device used for communication
-        void handleIncallSonification(int stream, bool starting);
-
-
-#ifdef AUDIO_POLICY_TEST
-        virtual     bool        threadLoop();
-                    void        exit();
-        int testOutputIndex(audio_io_handle_t output);
-#endif //AUDIO_POLICY_TEST
-
-
-        AudioPolicyClientInterface *mpClientInterface;  // audio policy client interface
-        audio_io_handle_t mHardwareOutput;              // hardware output handler
-
-        KeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs;   // list ot output descritors
-        KeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs;     // list of input descriptors
-        uint32_t mAvailableOutputDevices;                                   // bit field of all available output devices
-        uint32_t mAvailableInputDevices;                                    // bit field of all available input devices
-        int mPhoneState;                                                    // current phone state
-        uint32_t                 mRingerMode;                               // current ringer mode
-        AudioSystem::forced_config mForceUse[AudioSystem::NUM_FORCE_USE];   // current forced use configuration
-
-        StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES];           // stream descriptors for volume control
-
-#ifdef AUDIO_POLICY_TEST
-        Mutex   mLock;
-        Condition mWaitWorkCV;
-
-        int             mCurOutput;
-        bool            mDirectOutput;
-        audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
-        int             mTestInput;
-        uint32_t        mTestDevice;
-        uint32_t        mTestSamplingRate;
-        uint32_t        mTestFormat;
-        uint32_t        mTestChannels;
-        uint32_t        mTestLatencyMs;
-#endif //AUDIO_POLICY_TEST
-
-};
-
-};
diff --git a/libs/audioflinger/AudioPolicyService.cpp b/libs/audioflinger/AudioPolicyService.cpp
index aa48019..bb3905c 100644
--- a/libs/audioflinger/AudioPolicyService.cpp
+++ b/libs/audioflinger/AudioPolicyService.cpp
@@ -30,9 +30,10 @@
 #include <utils/String16.h>
 #include <utils/threads.h>
 #include "AudioPolicyService.h"
-#include "AudioPolicyManagerGeneric.h"
+#include <hardware_legacy/AudioPolicyManagerBase.h>
 #include <cutils/properties.h>
 #include <dlfcn.h>
+#include <hardware_legacy/power.h>
 
 // ----------------------------------------------------------------------------
 // the sim build doesn't have gettid
@@ -43,8 +44,9 @@
 
 namespace android {
 
-static const char* kDeadlockedString = "AudioPolicyService may be deadlocked\n";
-static const char* kCmdDeadlockedString = "AudioPolicyService command thread may be deadlocked\n";
+
+static const char *kDeadlockedString = "AudioPolicyService may be deadlocked\n";
+static const char *kCmdDeadlockedString = "AudioPolicyService command thread may be deadlocked\n";
 
 static const int kDumpLockRetries = 50;
 static const int kDumpLockSleep = 20000;
@@ -67,18 +69,18 @@
     char value[PROPERTY_VALUE_MAX];
 
     // start tone playback thread
-    mTonePlaybackThread = new AudioCommandThread();
+    mTonePlaybackThread = new AudioCommandThread(String8(""));
     // start audio commands thread
-    mAudioCommandThread = new AudioCommandThread();
+    mAudioCommandThread = new AudioCommandThread(String8("ApmCommandThread"));
 
 #if (defined GENERIC_AUDIO) || (defined AUDIO_POLICY_TEST)
-    mpPolicyManager = new AudioPolicyManagerGeneric(this);
+    mpPolicyManager = new AudioPolicyManagerBase(this);
     LOGV("build for GENERIC_AUDIO - using generic audio policy");
 #else
     // if running in emulation - use the emulator driver
     if (property_get("ro.kernel.qemu", value, 0)) {
         LOGV("Running in emulation - using generic audio policy");
-        mpPolicyManager = new AudioPolicyManagerGeneric(this);
+        mpPolicyManager = new AudioPolicyManagerBase(this);
     }
     else {
         LOGV("Using hardware specific audio policy");
@@ -556,8 +558,8 @@
 
 // -----------  AudioPolicyService::AudioCommandThread implementation ----------
 
-AudioPolicyService::AudioCommandThread::AudioCommandThread()
-    :   Thread(false)
+AudioPolicyService::AudioCommandThread::AudioCommandThread(String8 name)
+    : Thread(false), mName(name)
 {
     mpToneGenerator = NULL;
 }
@@ -565,18 +567,20 @@
 
 AudioPolicyService::AudioCommandThread::~AudioCommandThread()
 {
+    if (mName != "" && !mAudioCommands.isEmpty()) {
+        release_wake_lock(mName.string());
+    }
     mAudioCommands.clear();
     if (mpToneGenerator != NULL) delete mpToneGenerator;
 }
 
 void AudioPolicyService::AudioCommandThread::onFirstRef()
 {
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-
-    snprintf(buffer, SIZE, "AudioCommandThread");
-
-    run(buffer, ANDROID_PRIORITY_AUDIO);
+    if (mName != "") {
+        run(mName.string(), ANDROID_PRIORITY_AUDIO);
+    } else {
+        run("AudioCommandThread", ANDROID_PRIORITY_AUDIO);
+    }
 }
 
 bool AudioPolicyService::AudioCommandThread::threadLoop()
@@ -657,6 +661,10 @@
                 break;
             }
         }
+        // release delayed commands wake lock
+        if (mName != "" && mAudioCommands.isEmpty()) {
+            release_wake_lock(mName.string());
+        }
         LOGV("AudioCommandThread() going to sleep");
         mWaitWorkCV.waitRelative(mLock, waitTime);
         LOGV("AudioCommandThread() waking up");
@@ -815,6 +823,11 @@
 
     command->mTime = systemTime() + milliseconds(delayMs);
 
+    // acquire wake lock to make sure delayed commands are processed
+    if (mName != "" && mAudioCommands.isEmpty()) {
+        acquire_wake_lock(PARTIAL_WAKE_LOCK, mName.string());
+    }
+
     // check same pending commands with later time stamps and eliminate them
     for (i = mAudioCommands.size()-1; i >= 0; i--) {
         AudioCommand *command2 = mAudioCommands[i];
@@ -883,7 +896,7 @@
     removedCommands.clear();
 
     // insert command at the right place according to its time stamp
-    LOGV("inserting command: %d at index %ld, num commands %d", command->mCommand, i+1, mAudioCommands.size());
+    LOGV("inserting command: %d at index %d, num commands %d", command->mCommand, (int)i+1, mAudioCommands.size());
     mAudioCommands.insertAt(command, i + 1);
 }
 
diff --git a/libs/audioflinger/AudioPolicyService.h b/libs/audioflinger/AudioPolicyService.h
index b9234ec..a13d0bd 100644
--- a/libs/audioflinger/AudioPolicyService.h
+++ b/libs/audioflinger/AudioPolicyService.h
@@ -132,7 +132,7 @@
             SET_VOICE_VOLUME
         };
 
-        AudioCommandThread ();
+        AudioCommandThread (String8 name);
         virtual             ~AudioCommandThread();
 
                     status_t    dump(int fd);
@@ -195,7 +195,8 @@
         Condition mWaitWorkCV;
         Vector <AudioCommand *> mAudioCommands; // list of pending commands
         ToneGenerator *mpToneGenerator;     // the tone generator
-        AudioCommand mLastCommand;
+        AudioCommand mLastCommand;          // last processed command (used by dump)
+        String8 mName;                      // string used by wake lock fo delayed commands
     };
 
     // Internal dump utilities.