audioflinger: move legacy audio hw/policy out to libhardware_legacy

Change-Id: I4adcec73d3c08bcbe15bb19e1ba2ff18b195af45
Signed-off-by: Dima Zavin <dima@android.com>

11 files changed, 0 insertions, 4813 deletions

diff --git a/services/audioflinger/A2dpAudioInterface.cpp b/services/audioflinger/A2dpAudioInterface.cpp
deleted file mode 100644
index d926cb1..0000000
--- a/services/audioflinger/A2dpAudioInterface.cpp
+++ /dev/null
@@ -1,498 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT 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 <math.h>
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "A2dpAudioInterface"
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include "A2dpAudioInterface.h"
-#include "audio/liba2dp.h"
-#include <hardware_legacy/power.h>
-
-namespace android {
-
-static const char *sA2dpWakeLock = "A2dpOutputStream";
-#define MAX_WRITE_RETRIES  5
-
-// ----------------------------------------------------------------------------
-
-//AudioHardwareInterface* A2dpAudioInterface::createA2dpInterface()
-//{
-//    AudioHardwareInterface* hw = 0;
-//
-//    hw = AudioHardwareInterface::create();
-//    LOGD("new A2dpAudioInterface(hw: %p)", hw);
-//    hw = new A2dpAudioInterface(hw);
-//    return hw;
-//}
-
-A2dpAudioInterface::A2dpAudioInterface(AudioHardwareInterface* hw) :
-    mOutput(0), mHardwareInterface(hw), mBluetoothEnabled(true), mSuspended(false)
-{
-}
-
-A2dpAudioInterface::~A2dpAudioInterface()
-{
-    closeOutputStream((AudioStreamOut *)mOutput);
-    delete mHardwareInterface;
-}
-
-status_t A2dpAudioInterface::initCheck()
-{
-    if (mHardwareInterface == 0) return NO_INIT;
-    return mHardwareInterface->initCheck();
-}
-
-AudioStreamOut* A2dpAudioInterface::openOutputStream(
-        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
-{
-    if (!AudioSystem::isA2dpDevice((AudioSystem::audio_devices)devices)) {
-        LOGV("A2dpAudioInterface::openOutputStream() open HW device: %x", devices);
-        return mHardwareInterface->openOutputStream(devices, format, channels, sampleRate, status);
-    }
-
-    status_t err = 0;
-
-    // only one output stream allowed
-    if (mOutput) {
-        if (status)
-            *status = -1;
-        return NULL;
-    }
-
-    // create new output stream
-    A2dpAudioStreamOut* out = new A2dpAudioStreamOut();
-    if ((err = out->set(devices, format, channels, sampleRate)) == NO_ERROR) {
-        mOutput = out;
-        mOutput->setBluetoothEnabled(mBluetoothEnabled);
-        mOutput->setSuspended(mSuspended);
-    } else {
-        delete out;
-    }
-
-    if (status)
-        *status = err;
-    return mOutput;
-}
-
-void A2dpAudioInterface::closeOutputStream(AudioStreamOut* out) {
-    if (mOutput == 0 || mOutput != out) {
-        mHardwareInterface->closeOutputStream(out);
-    }
-    else {
-        delete mOutput;
-        mOutput = 0;
-    }
-}
-
-
-AudioStreamIn* A2dpAudioInterface::openInputStream(
-        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status,
-        AudioSystem::audio_in_acoustics acoustics)
-{
-    return mHardwareInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics);
-}
-
-void A2dpAudioInterface::closeInputStream(AudioStreamIn* in)
-{
-    return mHardwareInterface->closeInputStream(in);
-}
-
-status_t A2dpAudioInterface::setMode(int mode)
-{
-    return mHardwareInterface->setMode(mode);
-}
-
-status_t A2dpAudioInterface::setMicMute(bool state)
-{
-    return mHardwareInterface->setMicMute(state);
-}
-
-status_t A2dpAudioInterface::getMicMute(bool* state)
-{
-    return mHardwareInterface->getMicMute(state);
-}
-
-status_t A2dpAudioInterface::setParameters(const String8& keyValuePairs)
-{
-    AudioParameter param = AudioParameter(keyValuePairs);
-    String8 value;
-    String8 key;
-    status_t status = NO_ERROR;
-
-    LOGV("setParameters() %s", keyValuePairs.string());
-
-    key = "bluetooth_enabled";
-    if (param.get(key, value) == NO_ERROR) {
-        mBluetoothEnabled = (value == "true");
-        if (mOutput) {
-            mOutput->setBluetoothEnabled(mBluetoothEnabled);
-        }
-        param.remove(key);
-    }
-    key = String8("A2dpSuspended");
-    if (param.get(key, value) == NO_ERROR) {
-        mSuspended = (value == "true");
-        if (mOutput) {
-            mOutput->setSuspended(mSuspended);
-        }
-        param.remove(key);
-    }
-
-    if (param.size()) {
-        status_t hwStatus = mHardwareInterface->setParameters(param.toString());
-        if (status == NO_ERROR) {
-            status = hwStatus;
-        }
-    }
-
-    return status;
-}
-
-String8 A2dpAudioInterface::getParameters(const String8& keys)
-{
-    AudioParameter param = AudioParameter(keys);
-    AudioParameter a2dpParam = AudioParameter();
-    String8 value;
-    String8 key;
-
-    key = "bluetooth_enabled";
-    if (param.get(key, value) == NO_ERROR) {
-        value = mBluetoothEnabled ? "true" : "false";
-        a2dpParam.add(key, value);
-        param.remove(key);
-    }
-    key = "A2dpSuspended";
-    if (param.get(key, value) == NO_ERROR) {
-        value = mSuspended ? "true" : "false";
-        a2dpParam.add(key, value);
-        param.remove(key);
-    }
-
-    String8 keyValuePairs  = a2dpParam.toString();
-
-    if (param.size()) {
-        if (keyValuePairs != "") {
-            keyValuePairs += ";";
-        }
-        keyValuePairs += mHardwareInterface->getParameters(param.toString());
-    }
-
-    LOGV("getParameters() %s", keyValuePairs.string());
-    return keyValuePairs;
-}
-
-size_t A2dpAudioInterface::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
-{
-    return mHardwareInterface->getInputBufferSize(sampleRate, format, channelCount);
-}
-
-status_t A2dpAudioInterface::setVoiceVolume(float v)
-{
-    return mHardwareInterface->setVoiceVolume(v);
-}
-
-status_t A2dpAudioInterface::setMasterVolume(float v)
-{
-    return mHardwareInterface->setMasterVolume(v);
-}
-
-status_t A2dpAudioInterface::dump(int fd, const Vector<String16>& args)
-{
-    return mHardwareInterface->dumpState(fd, args);
-}
-
-// ----------------------------------------------------------------------------
-
-A2dpAudioInterface::A2dpAudioStreamOut::A2dpAudioStreamOut() :
-    mFd(-1), mStandby(true), mStartCount(0), mRetryCount(0), mData(NULL),
-    // assume BT enabled to start, this is safe because its only the
-    // enabled->disabled transition we are worried about
-    mBluetoothEnabled(true), mDevice(0), mClosing(false), mSuspended(false)
-{
-    // use any address by default
-    strcpy(mA2dpAddress, "00:00:00:00:00:00");
-    init();
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::set(
-        uint32_t device, int *pFormat, uint32_t *pChannels, uint32_t *pRate)
-{
-    int lFormat = pFormat ? *pFormat : 0;
-    uint32_t lChannels = pChannels ? *pChannels : 0;
-    uint32_t lRate = pRate ? *pRate : 0;
-
-    LOGD("A2dpAudioStreamOut::set %x, %d, %d, %d\n", device, lFormat, lChannels, lRate);
-
-    // fix up defaults
-    if (lFormat == 0) lFormat = format();
-    if (lChannels == 0) lChannels = channels();
-    if (lRate == 0) lRate = sampleRate();
-
-    // check values
-    if ((lFormat != format()) ||
-            (lChannels != channels()) ||
-            (lRate != sampleRate())){
-        if (pFormat) *pFormat = format();
-        if (pChannels) *pChannels = channels();
-        if (pRate) *pRate = sampleRate();
-        return BAD_VALUE;
-    }
-
-    if (pFormat) *pFormat = lFormat;
-    if (pChannels) *pChannels = lChannels;
-    if (pRate) *pRate = lRate;
-
-    mDevice = device;
-    mBufferDurationUs = ((bufferSize() * 1000 )/ frameSize() / sampleRate()) * 1000;
-    return NO_ERROR;
-}
-
-A2dpAudioInterface::A2dpAudioStreamOut::~A2dpAudioStreamOut()
-{
-    LOGV("A2dpAudioStreamOut destructor");
-    close();
-    LOGV("A2dpAudioStreamOut destructor returning from close()");
-}
-
-ssize_t A2dpAudioInterface::A2dpAudioStreamOut::write(const void* buffer, size_t bytes)
-{
-    status_t status = -1;
-    {
-        Mutex::Autolock lock(mLock);
-
-        size_t remaining = bytes;
-
-        if (!mBluetoothEnabled || mClosing || mSuspended) {
-            LOGV("A2dpAudioStreamOut::write(), but bluetooth disabled \
-                   mBluetoothEnabled %d, mClosing %d, mSuspended %d",
-                    mBluetoothEnabled, mClosing, mSuspended);
-            goto Error;
-        }
-
-        if (mStandby) {
-            acquire_wake_lock (PARTIAL_WAKE_LOCK, sA2dpWakeLock);
-            mStandby = false;
-            mLastWriteTime = systemTime();
-        }
-
-        status = init();
-        if (status < 0)
-            goto Error;
-
-        int retries = MAX_WRITE_RETRIES;
-        while (remaining > 0 && retries) {
-            status = a2dp_write(mData, buffer, remaining);
-            if (status < 0) {
-                LOGE("a2dp_write failed err: %d\n", status);
-                goto Error;
-            }
-            if (status == 0) {
-                retries--;
-            }
-            remaining -= status;
-            buffer = (char *)buffer + status;
-        }
-
-        // if A2DP sink runs abnormally fast, sleep a little so that audioflinger mixer thread
-        // does no spin and starve other threads.
-        // NOTE: It is likely that the A2DP headset is being disconnected
-        nsecs_t now = systemTime();
-        if ((uint32_t)ns2us(now - mLastWriteTime) < (mBufferDurationUs >> 2)) {
-            LOGV("A2DP sink runs too fast");
-            usleep(mBufferDurationUs - (uint32_t)ns2us(now - mLastWriteTime));
-        }
-        mLastWriteTime = now;
-        return bytes;
-
-    }
-Error:
-
-    standby();
-
-    // Simulate audio output timing in case of error
-    usleep(mBufferDurationUs);
-
-    return status;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::init()
-{
-    if (!mData) {
-        status_t status = a2dp_init(44100, 2, &mData);
-        if (status < 0) {
-            LOGE("a2dp_init failed err: %d\n", status);
-            mData = NULL;
-            return status;
-        }
-        a2dp_set_sink(mData, mA2dpAddress);
-    }
-
-    return 0;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::standby()
-{
-    Mutex::Autolock lock(mLock);
-    return standby_l();
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::standby_l()
-{
-    int result = NO_ERROR;
-
-    if (!mStandby) {
-        LOGV_IF(mClosing || !mBluetoothEnabled, "Standby skip stop: closing %d enabled %d",
-                mClosing, mBluetoothEnabled);
-        if (!mClosing && mBluetoothEnabled) {
-            result = a2dp_stop(mData);
-        }
-        release_wake_lock(sA2dpWakeLock);
-        mStandby = true;
-    }
-
-    return result;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::setParameters(const String8& keyValuePairs)
-{
-    AudioParameter param = AudioParameter(keyValuePairs);
-    String8 value;
-    String8 key = String8("a2dp_sink_address");
-    status_t status = NO_ERROR;
-    int device;
-    LOGV("A2dpAudioStreamOut::setParameters() %s", keyValuePairs.string());
-
-    if (param.get(key, value) == NO_ERROR) {
-        if (value.length() != strlen("00:00:00:00:00:00")) {
-            status = BAD_VALUE;
-        } else {
-            setAddress(value.string());
-        }
-        param.remove(key);
-    }
-    key = String8("closing");
-    if (param.get(key, value) == NO_ERROR) {
-        mClosing = (value == "true");
-        if (mClosing) {
-            standby();
-        }
-        param.remove(key);
-    }
-    key = AudioParameter::keyRouting;
-    if (param.getInt(key, device) == NO_ERROR) {
-        if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)device)) {
-            mDevice = device;
-            status = NO_ERROR;
-        } else {
-            status = BAD_VALUE;
-        }
-        param.remove(key);
-    }
-
-    if (param.size()) {
-        status = BAD_VALUE;
-    }
-    return status;
-}
-
-String8 A2dpAudioInterface::A2dpAudioStreamOut::getParameters(const String8& keys)
-{
-    AudioParameter param = AudioParameter(keys);
-    String8 value;
-    String8 key = String8("a2dp_sink_address");
-
-    if (param.get(key, value) == NO_ERROR) {
-        value = mA2dpAddress;
-        param.add(key, value);
-    }
-    key = AudioParameter::keyRouting;
-    if (param.get(key, value) == NO_ERROR) {
-        param.addInt(key, (int)mDevice);
-    }
-
-    LOGV("A2dpAudioStreamOut::getParameters() %s", param.toString().string());
-    return param.toString();
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::setAddress(const char* address)
-{
-    Mutex::Autolock lock(mLock);
-
-    if (strlen(address) != strlen("00:00:00:00:00:00"))
-        return -EINVAL;
-
-    strcpy(mA2dpAddress, address);
-    if (mData)
-        a2dp_set_sink(mData, mA2dpAddress);
-
-    return NO_ERROR;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::setBluetoothEnabled(bool enabled)
-{
-    LOGD("setBluetoothEnabled %d", enabled);
-
-    Mutex::Autolock lock(mLock);
-
-    mBluetoothEnabled = enabled;
-    if (!enabled) {
-        return close_l();
-    }
-    return NO_ERROR;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::setSuspended(bool onOff)
-{
-    LOGV("setSuspended %d", onOff);
-    mSuspended = onOff;
-    standby();
-    return NO_ERROR;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::close()
-{
-    Mutex::Autolock lock(mLock);
-    LOGV("A2dpAudioStreamOut::close() calling close_l()");
-    return close_l();
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::close_l()
-{
-    standby_l();
-    if (mData) {
-        LOGV("A2dpAudioStreamOut::close_l() calling a2dp_cleanup(mData)");
-        a2dp_cleanup(mData);
-        mData = NULL;
-    }
-    return NO_ERROR;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::dump(int fd, const Vector<String16>& args)
-{
-    return NO_ERROR;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::getRenderPosition(uint32_t *driverFrames)
-{
-    //TODO: enable when supported by driver
-    return INVALID_OPERATION;
-}
-
-}; // namespace android
diff --git a/services/audioflinger/A2dpAudioInterface.h b/services/audioflinger/A2dpAudioInterface.h
deleted file mode 100644
index dbe2c6a..0000000
--- a/services/audioflinger/A2dpAudioInterface.h
+++ /dev/null
@@ -1,138 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef A2DP_AUDIO_HARDWARE_H
-#define A2DP_AUDIO_HARDWARE_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <utils/threads.h>
-
-#include <hardware_legacy/AudioHardwareBase.h>
-
-
-namespace android {
-
-class A2dpAudioInterface : public AudioHardwareBase
-{
-    class A2dpAudioStreamOut;
-
-public:
-                        A2dpAudioInterface(AudioHardwareInterface* hw);
-    virtual             ~A2dpAudioInterface();
-    virtual status_t    initCheck();
-
-    virtual status_t    setVoiceVolume(float volume);
-    virtual status_t    setMasterVolume(float volume);
-
-    virtual status_t    setMode(int mode);
-
-    // mic mute
-    virtual status_t    setMicMute(bool state);
-    virtual status_t    getMicMute(bool* state);
-
-    virtual status_t    setParameters(const String8& keyValuePairs);
-    virtual String8     getParameters(const String8& keys);
-
-    virtual size_t      getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
-
-    // create I/O streams
-    virtual AudioStreamOut* openOutputStream(
-                                uint32_t devices,
-                                int *format=0,
-                                uint32_t *channels=0,
-                                uint32_t *sampleRate=0,
-                                status_t *status=0);
-    virtual    void        closeOutputStream(AudioStreamOut* out);
-
-    virtual AudioStreamIn* openInputStream(
-                                uint32_t devices,
-                                int *format,
-                                uint32_t *channels,
-                                uint32_t *sampleRate,
-                                status_t *status,
-                                AudioSystem::audio_in_acoustics acoustics);
-    virtual    void        closeInputStream(AudioStreamIn* in);
-//    static AudioHardwareInterface* createA2dpInterface();
-
-protected:
-    virtual status_t    dump(int fd, const Vector<String16>& args);
-
-private:
-    class A2dpAudioStreamOut : public AudioStreamOut {
-    public:
-                            A2dpAudioStreamOut();
-        virtual             ~A2dpAudioStreamOut();
-                status_t    set(uint32_t device,
-                                int *pFormat,
-                                uint32_t *pChannels,
-                                uint32_t *pRate);
-        virtual uint32_t    sampleRate() const { return 44100; }
-        // SBC codec wants a multiple of 512
-        virtual size_t      bufferSize() const { return 512 * 20; }
-        virtual uint32_t    channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
-        virtual int         format() const { return AudioSystem::PCM_16_BIT; }
-        virtual uint32_t    latency() const { return ((1000*bufferSize())/frameSize())/sampleRate() + 200; }
-        virtual status_t    setVolume(float left, float right) { return INVALID_OPERATION; }
-        virtual ssize_t     write(const void* buffer, size_t bytes);
-                status_t    standby();
-        virtual status_t    dump(int fd, const Vector<String16>& args);
-        virtual status_t    setParameters(const String8& keyValuePairs);
-        virtual String8     getParameters(const String8& keys);
-        virtual status_t    getRenderPosition(uint32_t *dspFrames);
-
-    private:
-        friend class A2dpAudioInterface;
-                status_t    init();
-                status_t    close();
-                status_t    close_l();
-                status_t    setAddress(const char* address);
-                status_t    setBluetoothEnabled(bool enabled);
-                status_t    setSuspended(bool onOff);
-                status_t    standby_l();
-
-    private:
-                int         mFd;
-                bool        mStandby;
-                int         mStartCount;
-                int         mRetryCount;
-                char        mA2dpAddress[20];
-                void*       mData;
-                Mutex       mLock;
-                bool        mBluetoothEnabled;
-                uint32_t    mDevice;
-                bool        mClosing;
-                bool        mSuspended;
-                nsecs_t     mLastWriteTime;
-                uint32_t    mBufferDurationUs;
-    };
-
-    friend class A2dpAudioStreamOut;
-
-    A2dpAudioStreamOut*     mOutput;
-    AudioHardwareInterface  *mHardwareInterface;
-    char        mA2dpAddress[20];
-    bool        mBluetoothEnabled;
-    bool        mSuspended;
-};
-
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
-
-#endif // A2DP_AUDIO_HARDWARE_H
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 69a4adc..6d78614 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -1,77 +1,5 @@
 LOCAL_PATH:= $(call my-dir)
 
-#AUDIO_POLICY_TEST := true
-#ENABLE_AUDIO_DUMP := true
-
-include $(CLEAR_VARS)
-
-
-ifeq ($(AUDIO_POLICY_TEST),true)
-  ENABLE_AUDIO_DUMP := true
-endif
-
-
-LOCAL_SRC_FILES:= \
-    AudioHardwareGeneric.cpp \
-    AudioHardwareStub.cpp \
-    AudioHardwareInterface.cpp
-
-ifeq ($(ENABLE_AUDIO_DUMP),true)
-  LOCAL_SRC_FILES += AudioDumpInterface.cpp
-  LOCAL_CFLAGS += -DENABLE_AUDIO_DUMP
-endif
-
-LOCAL_SHARED_LIBRARIES := \
-    libcutils \
-    libutils \
-    libbinder \
-    libmedia \
-    libhardware_legacy
-
-ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true)
-  LOCAL_CFLAGS += -DGENERIC_AUDIO
-endif
-
-LOCAL_MODULE:= libaudiointerface
-
-ifeq ($(BOARD_HAVE_BLUETOOTH),true)
-  LOCAL_SRC_FILES += A2dpAudioInterface.cpp
-  LOCAL_SHARED_LIBRARIES += liba2dp
-  LOCAL_CFLAGS += -DWITH_BLUETOOTH -DWITH_A2DP
-  LOCAL_C_INCLUDES += $(call include-path-for, bluez)
-endif
-
-include $(BUILD_STATIC_LIBRARY)
-
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:=               \
-    AudioPolicyManagerBase.cpp
-
-LOCAL_SHARED_LIBRARIES := \
-    libcutils \
-    libutils \
-    libmedia
-
-ifeq ($(TARGET_SIMULATOR),true)
- LOCAL_LDLIBS += -ldl
-else
- LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-LOCAL_MODULE:= libaudiopolicybase
-
-ifeq ($(BOARD_HAVE_BLUETOOTH),true)
-  LOCAL_CFLAGS += -DWITH_A2DP
-endif
-
-ifeq ($(AUDIO_POLICY_TEST),true)
-  LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
-endif
-
-include $(BUILD_STATIC_LIBRARY)
-
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES:=               \
@@ -90,12 +18,6 @@ LOCAL_SHARED_LIBRARIES := \
     libhardware_legacy \
     libeffects
 
-ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true)
-  LOCAL_STATIC_LIBRARIES += libaudiointerface libaudiopolicybase
-  LOCAL_CFLAGS += -DGENERIC_AUDIO
-else
-  LOCAL_SHARED_LIBRARIES += libaudio libaudiopolicy
-endif
 
 ifeq ($(TARGET_SIMULATOR),true)
  LOCAL_LDLIBS += -ldl
@@ -105,15 +27,6 @@ endif
 
 LOCAL_MODULE:= libaudioflinger
 
-ifeq ($(BOARD_HAVE_BLUETOOTH),true)
-  LOCAL_CFLAGS += -DWITH_BLUETOOTH -DWITH_A2DP
-  LOCAL_SHARED_LIBRARIES += liba2dp
-endif
-
-ifeq ($(AUDIO_POLICY_TEST),true)
-  LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
-endif
-
 ifeq ($(TARGET_SIMULATOR),true)
     ifeq ($(HOST_OS),linux)
         LOCAL_LDLIBS += -lrt -lpthread
diff --git a/services/audioflinger/AudioDumpInterface.cpp b/services/audioflinger/AudioDumpInterface.cpp
deleted file mode 100644
index 6c11114..0000000
--- a/services/audioflinger/AudioDumpInterface.cpp
+++ /dev/null
@@ -1,573 +0,0 @@
-/* //device/servers/AudioFlinger/AudioDumpInterface.cpp
-**
-** Copyright 2008, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT 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 "AudioFlingerDump"
-//#define LOG_NDEBUG 0
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <utils/Log.h>
-
-#include <stdlib.h>
-#include <unistd.h>
-
-#include "AudioDumpInterface.h"
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-AudioDumpInterface::AudioDumpInterface(AudioHardwareInterface* hw)
-    : mPolicyCommands(String8("")), mFileName(String8(""))
-{
-    if(hw == 0) {
-        LOGE("Dump construct hw = 0");
-    }
-    mFinalInterface = hw;
-    LOGV("Constructor %p, mFinalInterface %p", this, mFinalInterface);
-}
-
-
-AudioDumpInterface::~AudioDumpInterface()
-{
-    for (size_t i = 0; i < mOutputs.size(); i++) {
-        closeOutputStream((AudioStreamOut *)mOutputs[i]);
-    }
-
-    for (size_t i = 0; i < mInputs.size(); i++) {
-        closeInputStream((AudioStreamIn *)mInputs[i]);
-    }
-
-    if(mFinalInterface) delete mFinalInterface;
-}
-
-
-AudioStreamOut* AudioDumpInterface::openOutputStream(
-        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
-{
-    AudioStreamOut* outFinal = NULL;
-    int lFormat = AudioSystem::PCM_16_BIT;
-    uint32_t lChannels = AudioSystem::CHANNEL_OUT_STEREO;
-    uint32_t lRate = 44100;
-
-
-    outFinal = mFinalInterface->openOutputStream(devices, format, channels, sampleRate, status);
-    if (outFinal != 0) {
-        lFormat = outFinal->format();
-        lChannels = outFinal->channels();
-        lRate = outFinal->sampleRate();
-    } else {
-        if (format != 0) {
-            if (*format != 0) {
-                lFormat = *format;
-            } else {
-                *format = lFormat;
-            }
-        }
-        if (channels != 0) {
-            if (*channels != 0) {
-                lChannels = *channels;
-            } else {
-                *channels = lChannels;
-            }
-        }
-        if (sampleRate != 0) {
-            if (*sampleRate != 0) {
-                lRate = *sampleRate;
-            } else {
-                *sampleRate = lRate;
-            }
-        }
-        if (status) *status = NO_ERROR;
-    }
-    LOGV("openOutputStream(), outFinal %p", outFinal);
-
-    AudioStreamOutDump *dumOutput = new AudioStreamOutDump(this, mOutputs.size(), outFinal,
-            devices, lFormat, lChannels, lRate);
-    mOutputs.add(dumOutput);
-
-    return dumOutput;
-}
-
-void AudioDumpInterface::closeOutputStream(AudioStreamOut* out)
-{
-    AudioStreamOutDump *dumpOut = (AudioStreamOutDump *)out;
-
-    if (mOutputs.indexOf(dumpOut) < 0) {
-        LOGW("Attempt to close invalid output stream");
-        return;
-    }
-
-    LOGV("closeOutputStream() output %p", out);
-
-    dumpOut->standby();
-    if (dumpOut->finalStream() != NULL) {
-        mFinalInterface->closeOutputStream(dumpOut->finalStream());
-    }
-
-    mOutputs.remove(dumpOut);
-    delete dumpOut;
-}
-
-AudioStreamIn* AudioDumpInterface::openInputStream(uint32_t devices, int *format, uint32_t *channels,
-        uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics)
-{
-    AudioStreamIn* inFinal = NULL;
-    int lFormat = AudioSystem::PCM_16_BIT;
-    uint32_t lChannels = AudioSystem::CHANNEL_IN_MONO;
-    uint32_t lRate = 8000;
-
-    inFinal = mFinalInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics);
-    if (inFinal != 0) {
-        lFormat = inFinal->format();
-        lChannels = inFinal->channels();
-        lRate = inFinal->sampleRate();
-    } else {
-        if (format != 0) {
-            if (*format != 0) {
-                lFormat = *format;
-            } else {
-                *format = lFormat;
-            }
-        }
-        if (channels != 0) {
-            if (*channels != 0) {
-                lChannels = *channels;
-            } else {
-                *channels = lChannels;
-            }
-        }
-        if (sampleRate != 0) {
-            if (*sampleRate != 0) {
-                lRate = *sampleRate;
-            } else {
-                *sampleRate = lRate;
-            }
-        }
-        if (status) *status = NO_ERROR;
-    }
-    LOGV("openInputStream(), inFinal %p", inFinal);
-
-    AudioStreamInDump *dumInput = new AudioStreamInDump(this, mInputs.size(), inFinal,
-            devices, lFormat, lChannels, lRate);
-    mInputs.add(dumInput);
-
-    return dumInput;
-}
-void AudioDumpInterface::closeInputStream(AudioStreamIn* in)
-{
-    AudioStreamInDump *dumpIn = (AudioStreamInDump *)in;
-
-    if (mInputs.indexOf(dumpIn) < 0) {
-        LOGW("Attempt to close invalid input stream");
-        return;
-    }
-    dumpIn->standby();
-    if (dumpIn->finalStream() != NULL) {
-        mFinalInterface->closeInputStream(dumpIn->finalStream());
-    }
-
-    mInputs.remove(dumpIn);
-    delete dumpIn;
-}
-
-
-status_t AudioDumpInterface::setParameters(const String8& keyValuePairs)
-{
-    AudioParameter param = AudioParameter(keyValuePairs);
-    String8 value;
-    int valueInt;
-    LOGV("setParameters %s", keyValuePairs.string());
-
-    if (param.get(String8("test_cmd_file_name"), value) == NO_ERROR) {
-        mFileName = value;
-        param.remove(String8("test_cmd_file_name"));
-    }
-    if (param.get(String8("test_cmd_policy"), value) == NO_ERROR) {
-        Mutex::Autolock _l(mLock);
-        param.remove(String8("test_cmd_policy"));
-        mPolicyCommands = param.toString();
-        LOGV("test_cmd_policy command %s written", mPolicyCommands.string());
-        return NO_ERROR;
-    }
-
-    if (mFinalInterface != 0 ) return mFinalInterface->setParameters(keyValuePairs);
-    return NO_ERROR;
-}
-
-String8 AudioDumpInterface::getParameters(const String8& keys)
-{
-    AudioParameter param = AudioParameter(keys);
-    AudioParameter response;
-    String8 value;
-
-//    LOGV("getParameters %s", keys.string());
-    if (param.get(String8("test_cmd_policy"), value) == NO_ERROR) {
-        Mutex::Autolock _l(mLock);
-        if (mPolicyCommands.length() != 0) {
-            response = AudioParameter(mPolicyCommands);
-            response.addInt(String8("test_cmd_policy"), 1);
-        } else {
-            response.addInt(String8("test_cmd_policy"), 0);
-        }
-        param.remove(String8("test_cmd_policy"));
-//        LOGV("test_cmd_policy command %s read", mPolicyCommands.string());
-    }
-
-    if (param.get(String8("test_cmd_file_name"), value) == NO_ERROR) {
-        response.add(String8("test_cmd_file_name"), mFileName);
-        param.remove(String8("test_cmd_file_name"));
-    }
-
-    String8 keyValuePairs = response.toString();
-
-    if (param.size() && mFinalInterface != 0 ) {
-        keyValuePairs += ";";
-        keyValuePairs += mFinalInterface->getParameters(param.toString());
-    }
-
-    return keyValuePairs;
-}
-
-status_t AudioDumpInterface::setMode(int mode)
-{
-    return mFinalInterface->setMode(mode);
-}
-
-size_t AudioDumpInterface::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
-{
-    return mFinalInterface->getInputBufferSize(sampleRate, format, channelCount);
-}
-
-// ----------------------------------------------------------------------------
-
-AudioStreamOutDump::AudioStreamOutDump(AudioDumpInterface *interface,
-                                        int id,
-                                        AudioStreamOut* finalStream,
-                                        uint32_t devices,
-                                        int format,
-                                        uint32_t channels,
-                                        uint32_t sampleRate)
-    : mInterface(interface), mId(id),
-      mSampleRate(sampleRate), mFormat(format), mChannels(channels), mLatency(0), mDevice(devices),
-      mBufferSize(1024), mFinalStream(finalStream), mFile(0), mFileCount(0)
-{
-    LOGV("AudioStreamOutDump Constructor %p, mInterface %p, mFinalStream %p", this, mInterface, mFinalStream);
-}
-
-
-AudioStreamOutDump::~AudioStreamOutDump()
-{
-    LOGV("AudioStreamOutDump destructor");
-    Close();
-}
-
-ssize_t AudioStreamOutDump::write(const void* buffer, size_t bytes)
-{
-    ssize_t ret;
-
-    if (mFinalStream) {
-        ret = mFinalStream->write(buffer, bytes);
-    } else {
-        usleep((((bytes * 1000) / frameSize()) / sampleRate()) * 1000);
-        ret = bytes;
-    }
-    if(!mFile) {
-        if (mInterface->fileName() != "") {
-            char name[255];
-            sprintf(name, "%s_out_%d_%d.pcm", mInterface->fileName().string(), mId, ++mFileCount);
-            mFile = fopen(name, "wb");
-            LOGV("Opening dump file %s, fh %p", name, mFile);
-        }
-    }
-    if (mFile) {
-        fwrite(buffer, bytes, 1, mFile);
-    }
-    return ret;
-}
-
-status_t AudioStreamOutDump::standby()
-{
-    LOGV("AudioStreamOutDump standby(), mFile %p, mFinalStream %p", mFile, mFinalStream);
-
-    Close();
-    if (mFinalStream != 0 ) return mFinalStream->standby();
-    return NO_ERROR;
-}
-
-uint32_t AudioStreamOutDump::sampleRate() const
-{
-    if (mFinalStream != 0 ) return mFinalStream->sampleRate();
-    return mSampleRate;
-}
-
-size_t AudioStreamOutDump::bufferSize() const
-{
-    if (mFinalStream != 0 ) return mFinalStream->bufferSize();
-    return mBufferSize;
-}
-
-uint32_t AudioStreamOutDump::channels() const
-{
-    if (mFinalStream != 0 ) return mFinalStream->channels();
-    return mChannels;
-}
-int AudioStreamOutDump::format() const
-{
-    if (mFinalStream != 0 ) return mFinalStream->format();
-    return mFormat;
-}
-uint32_t AudioStreamOutDump::latency() const
-{
-    if (mFinalStream != 0 ) return mFinalStream->latency();
-    return 0;
-}
-status_t AudioStreamOutDump::setVolume(float left, float right)
-{
-    if (mFinalStream != 0 ) return mFinalStream->setVolume(left, right);
-    return NO_ERROR;
-}
-status_t AudioStreamOutDump::setParameters(const String8& keyValuePairs)
-{
-    LOGV("AudioStreamOutDump::setParameters %s", keyValuePairs.string());
-
-    if (mFinalStream != 0 ) {
-        return mFinalStream->setParameters(keyValuePairs);
-    }
-
-    AudioParameter param = AudioParameter(keyValuePairs);
-    String8 value;
-    int valueInt;
-    status_t status = NO_ERROR;
-
-    if (param.getInt(String8("set_id"), valueInt) == NO_ERROR) {
-        mId = valueInt;
-    }
-
-    if (param.getInt(String8("format"), valueInt) == NO_ERROR) {
-        if (mFile == 0) {
-            mFormat = valueInt;
-        } else {
-            status = INVALID_OPERATION;
-        }
-    }
-    if (param.getInt(String8("channels"), valueInt) == NO_ERROR) {
-        if (valueInt == AudioSystem::CHANNEL_OUT_STEREO || valueInt == AudioSystem::CHANNEL_OUT_MONO) {
-            mChannels = valueInt;
-        } else {
-            status = BAD_VALUE;
-        }
-    }
-    if (param.getInt(String8("sampling_rate"), valueInt) == NO_ERROR) {
-        if (valueInt > 0 && valueInt <= 48000) {
-            if (mFile == 0) {
-                mSampleRate = valueInt;
-            } else {
-                status = INVALID_OPERATION;
-            }
-        } else {
-            status = BAD_VALUE;
-        }
-    }
-    return status;
-}
-
-String8 AudioStreamOutDump::getParameters(const String8& keys)
-{
-    if (mFinalStream != 0 ) return mFinalStream->getParameters(keys);
-
-    AudioParameter param = AudioParameter(keys);
-    return param.toString();
-}
-
-status_t AudioStreamOutDump::dump(int fd, const Vector<String16>& args)
-{
-    if (mFinalStream != 0 ) return mFinalStream->dump(fd, args);
-    return NO_ERROR;
-}
-
-void AudioStreamOutDump::Close()
-{
-    if(mFile) {
-        fclose(mFile);
-        mFile = 0;
-    }
-}
-
-status_t AudioStreamOutDump::getRenderPosition(uint32_t *dspFrames)
-{
-    if (mFinalStream != 0 ) return mFinalStream->getRenderPosition(dspFrames);
-    return INVALID_OPERATION;
-}
-
-// ----------------------------------------------------------------------------
-
-AudioStreamInDump::AudioStreamInDump(AudioDumpInterface *interface,
-                                        int id,
-                                        AudioStreamIn* finalStream,
-                                        uint32_t devices,
-                                        int format,
-                                        uint32_t channels,
-                                        uint32_t sampleRate)
-    : mInterface(interface), mId(id),
-      mSampleRate(sampleRate), mFormat(format), mChannels(channels), mDevice(devices),
-      mBufferSize(1024), mFinalStream(finalStream), mFile(0), mFileCount(0)
-{
-    LOGV("AudioStreamInDump Constructor %p, mInterface %p, mFinalStream %p", this, mInterface, mFinalStream);
-}
-
-
-AudioStreamInDump::~AudioStreamInDump()
-{
-    Close();
-}
-
-ssize_t AudioStreamInDump::read(void* buffer, ssize_t bytes)
-{
-    ssize_t ret;
-
-    if (mFinalStream) {
-        ret = mFinalStream->read(buffer, bytes);
-        if(!mFile) {
-            if (mInterface->fileName() != "") {
-                char name[255];
-                sprintf(name, "%s_in_%d_%d.pcm", mInterface->fileName().string(), mId, ++mFileCount);
-                mFile = fopen(name, "wb");
-                LOGV("Opening input dump file %s, fh %p", name, mFile);
-            }
-        }
-        if (mFile) {
-            fwrite(buffer, bytes, 1, mFile);
-        }
-    } else {
-        usleep((((bytes * 1000) / frameSize()) / sampleRate()) * 1000);
-        ret = bytes;
-        if(!mFile) {
-            char name[255];
-            strcpy(name, "/sdcard/music/sine440");
-            if (channels() == AudioSystem::CHANNEL_IN_MONO) {
-                strcat(name, "_mo");
-            } else {
-                strcat(name, "_st");
-            }
-            if (format() == AudioSystem::PCM_16_BIT) {
-                strcat(name, "_16b");
-            } else {
-                strcat(name, "_8b");
-            }
-            if (sampleRate() < 16000) {
-                strcat(name, "_8k");
-            } else if (sampleRate() < 32000) {
-                strcat(name, "_22k");
-            } else if (sampleRate() < 48000) {
-                strcat(name, "_44k");
-            } else {
-                strcat(name, "_48k");
-            }
-            strcat(name, ".wav");
-            mFile = fopen(name, "rb");
-            LOGV("Opening input read file %s, fh %p", name, mFile);
-            if (mFile) {
-                fseek(mFile, AUDIO_DUMP_WAVE_HDR_SIZE, SEEK_SET);
-            }
-        }
-        if (mFile) {
-            ssize_t bytesRead = fread(buffer, bytes, 1, mFile);
-            if (bytesRead >=0 && bytesRead < bytes) {
-                fseek(mFile, AUDIO_DUMP_WAVE_HDR_SIZE, SEEK_SET);
-                fread((uint8_t *)buffer+bytesRead, bytes-bytesRead, 1, mFile);
-            }
-        }
-    }
-
-    return ret;
-}
-
-status_t AudioStreamInDump::standby()
-{
-    LOGV("AudioStreamInDump standby(), mFile %p, mFinalStream %p", mFile, mFinalStream);
-
-    Close();
-    if (mFinalStream != 0 ) return mFinalStream->standby();
-    return NO_ERROR;
-}
-
-status_t AudioStreamInDump::setGain(float gain)
-{
-    if (mFinalStream != 0 ) return mFinalStream->setGain(gain);
-    return NO_ERROR;
-}
-
-uint32_t AudioStreamInDump::sampleRate() const
-{
-    if (mFinalStream != 0 ) return mFinalStream->sampleRate();
-    return mSampleRate;
-}
-
-size_t AudioStreamInDump::bufferSize() const
-{
-    if (mFinalStream != 0 ) return mFinalStream->bufferSize();
-    return mBufferSize;
-}
-
-uint32_t AudioStreamInDump::channels() const
-{
-    if (mFinalStream != 0 ) return mFinalStream->channels();
-    return mChannels;
-}
-
-int AudioStreamInDump::format() const
-{
-    if (mFinalStream != 0 ) return mFinalStream->format();
-    return mFormat;
-}
-
-status_t AudioStreamInDump::setParameters(const String8& keyValuePairs)
-{
-    LOGV("AudioStreamInDump::setParameters()");
-    if (mFinalStream != 0 ) return mFinalStream->setParameters(keyValuePairs);
-    return NO_ERROR;
-}
-
-String8 AudioStreamInDump::getParameters(const String8& keys)
-{
-    if (mFinalStream != 0 ) return mFinalStream->getParameters(keys);
-
-    AudioParameter param = AudioParameter(keys);
-    return param.toString();
-}
-
-unsigned int AudioStreamInDump::getInputFramesLost() const
-{
-    if (mFinalStream != 0 ) return mFinalStream->getInputFramesLost();
-    return 0;
-}
-
-status_t AudioStreamInDump::dump(int fd, const Vector<String16>& args)
-{
-    if (mFinalStream != 0 ) return mFinalStream->dump(fd, args);
-    return NO_ERROR;
-}
-
-void AudioStreamInDump::Close()
-{
-    if(mFile) {
-        fclose(mFile);
-        mFile = 0;
-    }
-}
-}; // namespace android
diff --git a/services/audioflinger/AudioDumpInterface.h b/services/audioflinger/AudioDumpInterface.h
deleted file mode 100644
index 814ce5f..0000000
--- a/services/audioflinger/AudioDumpInterface.h
+++ /dev/null
@@ -1,170 +0,0 @@
-/* //device/servers/AudioFlinger/AudioDumpInterface.h
-**
-** Copyright 2008, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#ifndef ANDROID_AUDIO_DUMP_INTERFACE_H
-#define ANDROID_AUDIO_DUMP_INTERFACE_H
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <utils/String8.h>
-#include <utils/SortedVector.h>
-
-#include <hardware_legacy/AudioHardwareBase.h>
-
-namespace android {
-
-#define AUDIO_DUMP_WAVE_HDR_SIZE 44
-
-class AudioDumpInterface;
-
-class AudioStreamOutDump : public AudioStreamOut {
-public:
-                        AudioStreamOutDump(AudioDumpInterface *interface,
-                                            int id,
-                                            AudioStreamOut* finalStream,
-                                            uint32_t devices,
-                                            int format,
-                                            uint32_t channels,
-                                            uint32_t sampleRate);
-                        ~AudioStreamOutDump();
-
-    virtual ssize_t     write(const void* buffer, size_t bytes);
-    virtual uint32_t    sampleRate() const;
-    virtual size_t      bufferSize() const;
-    virtual uint32_t    channels() const;
-    virtual int         format() const;
-    virtual uint32_t    latency() const;
-    virtual status_t    setVolume(float left, float right);
-    virtual status_t    standby();
-    virtual status_t    setParameters(const String8& keyValuePairs);
-    virtual String8     getParameters(const String8& keys);
-    virtual status_t    dump(int fd, const Vector<String16>& args);
-    void                Close(void);
-    AudioStreamOut*     finalStream() { return mFinalStream; }
-    uint32_t            device() { return mDevice; }
-    int                 getId()  { return mId; }
-    virtual status_t    getRenderPosition(uint32_t *dspFrames);
-
-private:
-    AudioDumpInterface *mInterface;
-    int                  mId;
-    uint32_t mSampleRate;               //
-    uint32_t mFormat;                   //
-    uint32_t mChannels;                 // output configuration
-    uint32_t mLatency;                  //
-    uint32_t mDevice;                   // current device this output is routed to
-    size_t  mBufferSize;
-    AudioStreamOut      *mFinalStream;
-    FILE                *mFile;      // output file
-    int                 mFileCount;
-};
-
-class AudioStreamInDump : public AudioStreamIn {
-public:
-                        AudioStreamInDump(AudioDumpInterface *interface,
-                                            int id,
-                                            AudioStreamIn* finalStream,
-                                            uint32_t devices,
-                                            int format,
-                                            uint32_t channels,
-                                            uint32_t sampleRate);
-                        ~AudioStreamInDump();
-
-    virtual uint32_t    sampleRate() const;
-    virtual size_t      bufferSize() const;
-    virtual uint32_t    channels() const;
-    virtual int         format() const;
-
-    virtual status_t    setGain(float gain);
-    virtual ssize_t     read(void* buffer, ssize_t bytes);
-    virtual status_t    standby();
-    virtual status_t    setParameters(const String8& keyValuePairs);
-    virtual String8     getParameters(const String8& keys);
-    virtual unsigned int  getInputFramesLost() const;
-    virtual status_t    dump(int fd, const Vector<String16>& args);
-    void                Close(void);
-    AudioStreamIn*     finalStream() { return mFinalStream; }
-    uint32_t            device() { return mDevice; }
-
-private:
-    AudioDumpInterface *mInterface;
-    int                  mId;
-    uint32_t mSampleRate;               //
-    uint32_t mFormat;                   //
-    uint32_t mChannels;                 // output configuration
-    uint32_t mDevice;                   // current device this output is routed to
-    size_t  mBufferSize;
-    AudioStreamIn      *mFinalStream;
-    FILE                *mFile;      // output file
-    int                 mFileCount;
-};
-
-class AudioDumpInterface : public AudioHardwareBase
-{
-
-public:
-                        AudioDumpInterface(AudioHardwareInterface* hw);
-    virtual AudioStreamOut* openOutputStream(
-                                uint32_t devices,
-                                int *format=0,
-                                uint32_t *channels=0,
-                                uint32_t *sampleRate=0,
-                                status_t *status=0);
-    virtual    void        closeOutputStream(AudioStreamOut* out);
-
-    virtual             ~AudioDumpInterface();
-
-    virtual status_t    initCheck()
-                            {return mFinalInterface->initCheck();}
-    virtual status_t    setVoiceVolume(float volume)
-                            {return mFinalInterface->setVoiceVolume(volume);}
-    virtual status_t    setMasterVolume(float volume)
-                            {return mFinalInterface->setMasterVolume(volume);}
-
-    virtual status_t    setMode(int mode);
-
-    // mic mute
-    virtual status_t    setMicMute(bool state)
-                            {return mFinalInterface->setMicMute(state);}
-    virtual status_t    getMicMute(bool* state)
-                            {return mFinalInterface->getMicMute(state);}
-
-    virtual status_t    setParameters(const String8& keyValuePairs);
-    virtual String8     getParameters(const String8& keys);
-
-    virtual size_t      getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
-
-    virtual AudioStreamIn* openInputStream(uint32_t devices, int *format, uint32_t *channels,
-            uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics);
-    virtual    void        closeInputStream(AudioStreamIn* in);
-
-    virtual status_t    dump(int fd, const Vector<String16>& args) { return mFinalInterface->dumpState(fd, args); }
-
-            String8     fileName() const { return mFileName; }
-protected:
-
-    AudioHardwareInterface          *mFinalInterface;
-    SortedVector<AudioStreamOutDump *>   mOutputs;
-    SortedVector<AudioStreamInDump *>    mInputs;
-    Mutex                           mLock;
-    String8                         mPolicyCommands;
-    String8                         mFileName;
-};
-
-}; // namespace android
-
-#endif // ANDROID_AUDIO_DUMP_INTERFACE_H
diff --git a/services/audioflinger/AudioHardwareGeneric.cpp b/services/audioflinger/AudioHardwareGeneric.cpp
deleted file mode 100644
index d63c031..0000000
--- a/services/audioflinger/AudioHardwareGeneric.cpp
+++ /dev/null
@@ -1,411 +0,0 @@
-/*
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <unistd.h>
-#include <sched.h>
-#include <fcntl.h>
-#include <sys/ioctl.h>
-
-#define LOG_TAG "AudioHardware"
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include "AudioHardwareGeneric.h"
-#include <media/AudioRecord.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-static char const * const kAudioDeviceName = "/dev/eac";
-
-// ----------------------------------------------------------------------------
-
-AudioHardwareGeneric::AudioHardwareGeneric()
-    : mOutput(0), mInput(0),  mFd(-1), mMicMute(false)
-{
-    mFd = ::open(kAudioDeviceName, O_RDWR);
-}
-
-AudioHardwareGeneric::~AudioHardwareGeneric()
-{
-    if (mFd >= 0) ::close(mFd);
-    closeOutputStream((AudioStreamOut *)mOutput);
-    closeInputStream((AudioStreamIn *)mInput);
-}
-
-status_t AudioHardwareGeneric::initCheck()
-{
-    if (mFd >= 0) {
-        if (::access(kAudioDeviceName, O_RDWR) == NO_ERROR)
-            return NO_ERROR;
-    }
-    return NO_INIT;
-}
-
-AudioStreamOut* AudioHardwareGeneric::openOutputStream(
-        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
-{
-    AutoMutex lock(mLock);
-
-    // only one output stream allowed
-    if (mOutput) {
-        if (status) {
-            *status = INVALID_OPERATION;
-        }
-        return 0;
-    }
-
-    // create new output stream
-    AudioStreamOutGeneric* out = new AudioStreamOutGeneric();
-    status_t lStatus = out->set(this, mFd, devices, format, channels, sampleRate);
-    if (status) {
-        *status = lStatus;
-    }
-    if (lStatus == NO_ERROR) {
-        mOutput = out;
-    } else {
-        delete out;
-    }
-    return mOutput;
-}
-
-void AudioHardwareGeneric::closeOutputStream(AudioStreamOut* out) {
-    if (mOutput && out == mOutput) {
-        delete mOutput;
-        mOutput = 0;
-    }
-}
-
-AudioStreamIn* AudioHardwareGeneric::openInputStream(
-        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate,
-        status_t *status, AudioSystem::audio_in_acoustics acoustics)
-{
-    // check for valid input source
-    if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) {
-        return 0;
-    }
-
-    AutoMutex lock(mLock);
-
-    // only one input stream allowed
-    if (mInput) {
-        if (status) {
-            *status = INVALID_OPERATION;
-        }
-        return 0;
-    }
-
-    // create new output stream
-    AudioStreamInGeneric* in = new AudioStreamInGeneric();
-    status_t lStatus = in->set(this, mFd, devices, format, channels, sampleRate, acoustics);
-    if (status) {
-        *status = lStatus;
-    }
-    if (lStatus == NO_ERROR) {
-        mInput = in;
-    } else {
-        delete in;
-    }
-    return mInput;
-}
-
-void AudioHardwareGeneric::closeInputStream(AudioStreamIn* in) {
-    if (mInput && in == mInput) {
-        delete mInput;
-        mInput = 0;
-    }
-}
-
-status_t AudioHardwareGeneric::setVoiceVolume(float v)
-{
-    // Implement: set voice volume
-    return NO_ERROR;
-}
-
-status_t AudioHardwareGeneric::setMasterVolume(float v)
-{
-    // Implement: set master volume
-    // return error - software mixer will handle it
-    return INVALID_OPERATION;
-}
-
-status_t AudioHardwareGeneric::setMicMute(bool state)
-{
-    mMicMute = state;
-    return NO_ERROR;
-}
-
-status_t AudioHardwareGeneric::getMicMute(bool* state)
-{
-    *state = mMicMute;
-    return NO_ERROR;
-}
-
-status_t AudioHardwareGeneric::dumpInternals(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-    result.append("AudioHardwareGeneric::dumpInternals\n");
-    snprintf(buffer, SIZE, "\tmFd: %d mMicMute: %s\n",  mFd, mMicMute? "true": "false");
-    result.append(buffer);
-    ::write(fd, result.string(), result.size());
-    return NO_ERROR;
-}
-
-status_t AudioHardwareGeneric::dump(int fd, const Vector<String16>& args)
-{
-    dumpInternals(fd, args);
-    if (mInput) {
-        mInput->dump(fd, args);
-    }
-    if (mOutput) {
-        mOutput->dump(fd, args);
-    }
-    return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-
-status_t AudioStreamOutGeneric::set(
-        AudioHardwareGeneric *hw,
-        int fd,
-        uint32_t devices,
-        int *pFormat,
-        uint32_t *pChannels,
-        uint32_t *pRate)
-{
-    int lFormat = pFormat ? *pFormat : 0;
-    uint32_t lChannels = pChannels ? *pChannels : 0;
-    uint32_t lRate = pRate ? *pRate : 0;
-
-    // fix up defaults
-    if (lFormat == 0) lFormat = format();
-    if (lChannels == 0) lChannels = channels();
-    if (lRate == 0) lRate = sampleRate();
-
-    // check values
-    if ((lFormat != format()) ||
-            (lChannels != channels()) ||
-            (lRate != sampleRate())) {
-        if (pFormat) *pFormat = format();
-        if (pChannels) *pChannels = channels();
-        if (pRate) *pRate = sampleRate();
-        return BAD_VALUE;
-    }
-
-    if (pFormat) *pFormat = lFormat;
-    if (pChannels) *pChannels = lChannels;
-    if (pRate) *pRate = lRate;
-
-    mAudioHardware = hw;
-    mFd = fd;
-    mDevice = devices;
-    return NO_ERROR;
-}
-
-AudioStreamOutGeneric::~AudioStreamOutGeneric()
-{
-}
-
-ssize_t AudioStreamOutGeneric::write(const void* buffer, size_t bytes)
-{
-    Mutex::Autolock _l(mLock);
-    return ssize_t(::write(mFd, buffer, bytes));
-}
-
-status_t AudioStreamOutGeneric::standby()
-{
-    // Implement: audio hardware to standby mode
-    return NO_ERROR;
-}
-
-status_t AudioStreamOutGeneric::dump(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-    snprintf(buffer, SIZE, "AudioStreamOutGeneric::dump\n");
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tformat: %d\n", format());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tdevice: %d\n", mDevice);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tmAudioHardware: %p\n", mAudioHardware);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tmFd: %d\n", mFd);
-    result.append(buffer);
-    ::write(fd, result.string(), result.size());
-    return NO_ERROR;
-}
-
-status_t AudioStreamOutGeneric::setParameters(const String8& keyValuePairs)
-{
-    AudioParameter param = AudioParameter(keyValuePairs);
-    String8 key = String8(AudioParameter::keyRouting);
-    status_t status = NO_ERROR;
-    int device;
-    LOGV("setParameters() %s", keyValuePairs.string());
-
-    if (param.getInt(key, device) == NO_ERROR) {
-        mDevice = device;
-        param.remove(key);
-    }
-
-    if (param.size()) {
-        status = BAD_VALUE;
-    }
-    return status;
-}
-
-String8 AudioStreamOutGeneric::getParameters(const String8& keys)
-{
-    AudioParameter param = AudioParameter(keys);
-    String8 value;
-    String8 key = String8(AudioParameter::keyRouting);
-
-    if (param.get(key, value) == NO_ERROR) {
-        param.addInt(key, (int)mDevice);
-    }
-
-    LOGV("getParameters() %s", param.toString().string());
-    return param.toString();
-}
-
-status_t AudioStreamOutGeneric::getRenderPosition(uint32_t *dspFrames)
-{
-    return INVALID_OPERATION;
-}
-
-// ----------------------------------------------------------------------------
-
-// record functions
-status_t AudioStreamInGeneric::set(
-        AudioHardwareGeneric *hw,
-        int fd,
-        uint32_t devices,
-        int *pFormat,
-        uint32_t *pChannels,
-        uint32_t *pRate,
-        AudioSystem::audio_in_acoustics acoustics)
-{
-    if (pFormat == 0 || pChannels == 0 || pRate == 0) return BAD_VALUE;
-    LOGV("AudioStreamInGeneric::set(%p, %d, %d, %d, %u)", hw, fd, *pFormat, *pChannels, *pRate);
-    // check values
-    if ((*pFormat != format()) ||
-        (*pChannels != channels()) ||
-        (*pRate != sampleRate())) {
-        LOGE("Error opening input channel");
-        *pFormat = format();
-        *pChannels = channels();
-        *pRate = sampleRate();
-        return BAD_VALUE;
-    }
-
-    mAudioHardware = hw;
-    mFd = fd;
-    mDevice = devices;
-    return NO_ERROR;
-}
-
-AudioStreamInGeneric::~AudioStreamInGeneric()
-{
-}
-
-ssize_t AudioStreamInGeneric::read(void* buffer, ssize_t bytes)
-{
-    AutoMutex lock(mLock);
-    if (mFd < 0) {
-        LOGE("Attempt to read from unopened device");
-        return NO_INIT;
-    }
-    return ::read(mFd, buffer, bytes);
-}
-
-status_t AudioStreamInGeneric::dump(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-    snprintf(buffer, SIZE, "AudioStreamInGeneric::dump\n");
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tformat: %d\n", format());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tdevice: %d\n", mDevice);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tmAudioHardware: %p\n", mAudioHardware);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tmFd: %d\n", mFd);
-    result.append(buffer);
-    ::write(fd, result.string(), result.size());
-    return NO_ERROR;
-}
-
-status_t AudioStreamInGeneric::setParameters(const String8& keyValuePairs)
-{
-    AudioParameter param = AudioParameter(keyValuePairs);
-    String8 key = String8(AudioParameter::keyRouting);
-    status_t status = NO_ERROR;
-    int device;
-    LOGV("setParameters() %s", keyValuePairs.string());
-
-    if (param.getInt(key, device) == NO_ERROR) {
-        mDevice = device;
-        param.remove(key);
-    }
-
-    if (param.size()) {
-        status = BAD_VALUE;
-    }
-    return status;
-}
-
-String8 AudioStreamInGeneric::getParameters(const String8& keys)
-{
-    AudioParameter param = AudioParameter(keys);
-    String8 value;
-    String8 key = String8(AudioParameter::keyRouting);
-
-    if (param.get(key, value) == NO_ERROR) {
-        param.addInt(key, (int)mDevice);
-    }
-
-    LOGV("getParameters() %s", param.toString().string());
-    return param.toString();
-}
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
diff --git a/services/audioflinger/AudioHardwareGeneric.h b/services/audioflinger/AudioHardwareGeneric.h
deleted file mode 100644
index aa4e78d..0000000
--- a/services/audioflinger/AudioHardwareGeneric.h
+++ /dev/null
@@ -1,151 +0,0 @@
-/*
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#ifndef ANDROID_AUDIO_HARDWARE_GENERIC_H
-#define ANDROID_AUDIO_HARDWARE_GENERIC_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <utils/threads.h>
-
-#include <hardware_legacy/AudioHardwareBase.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-class AudioHardwareGeneric;
-
-class AudioStreamOutGeneric : public AudioStreamOut {
-public:
-                        AudioStreamOutGeneric() : mAudioHardware(0), mFd(-1) {}
-    virtual             ~AudioStreamOutGeneric();
-
-    virtual status_t    set(
-            AudioHardwareGeneric *hw,
-            int mFd,
-            uint32_t devices,
-            int *pFormat,
-            uint32_t *pChannels,
-            uint32_t *pRate);
-
-    virtual uint32_t    sampleRate() const { return 44100; }
-    virtual size_t      bufferSize() const { return 4096; }
-    virtual uint32_t    channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
-    virtual int         format() const { return AudioSystem::PCM_16_BIT; }
-    virtual uint32_t    latency() const { return 20; }
-    virtual status_t    setVolume(float left, float right) { return INVALID_OPERATION; }
-    virtual ssize_t     write(const void* buffer, size_t bytes);
-    virtual status_t    standby();
-    virtual status_t    dump(int fd, const Vector<String16>& args);
-    virtual status_t    setParameters(const String8& keyValuePairs);
-    virtual String8     getParameters(const String8& keys);
-    virtual status_t    getRenderPosition(uint32_t *dspFrames);
-
-private:
-    AudioHardwareGeneric *mAudioHardware;
-    Mutex   mLock;
-    int     mFd;
-    uint32_t mDevice;
-};
-
-class AudioStreamInGeneric : public AudioStreamIn {
-public:
-                        AudioStreamInGeneric() : mAudioHardware(0), mFd(-1) {}
-    virtual             ~AudioStreamInGeneric();
-
-    virtual status_t    set(
-            AudioHardwareGeneric *hw,
-            int mFd,
-            uint32_t devices,
-            int *pFormat,
-            uint32_t *pChannels,
-            uint32_t *pRate,
-            AudioSystem::audio_in_acoustics acoustics);
-
-    virtual uint32_t    sampleRate() const { return 8000; }
-    virtual size_t      bufferSize() const { return 320; }
-    virtual uint32_t    channels() const { return AudioSystem::CHANNEL_IN_MONO; }
-    virtual int         format() const { return AudioSystem::PCM_16_BIT; }
-    virtual status_t    setGain(float gain) { return INVALID_OPERATION; }
-    virtual ssize_t     read(void* buffer, ssize_t bytes);
-    virtual status_t    dump(int fd, const Vector<String16>& args);
-    virtual status_t    standby() { return NO_ERROR; }
-    virtual status_t    setParameters(const String8& keyValuePairs);
-    virtual String8     getParameters(const String8& keys);
-    virtual unsigned int  getInputFramesLost() const { return 0; }
-
-private:
-    AudioHardwareGeneric *mAudioHardware;
-    Mutex   mLock;
-    int     mFd;
-    uint32_t mDevice;
-};
-
-
-class AudioHardwareGeneric : public AudioHardwareBase
-{
-public:
-                        AudioHardwareGeneric();
-    virtual             ~AudioHardwareGeneric();
-    virtual status_t    initCheck();
-    virtual status_t    setVoiceVolume(float volume);
-    virtual status_t    setMasterVolume(float volume);
-
-    // mic mute
-    virtual status_t    setMicMute(bool state);
-    virtual status_t    getMicMute(bool* state);
-
-    // create I/O streams
-    virtual AudioStreamOut* openOutputStream(
-            uint32_t devices,
-            int *format=0,
-            uint32_t *channels=0,
-            uint32_t *sampleRate=0,
-            status_t *status=0);
-    virtual    void        closeOutputStream(AudioStreamOut* out);
-
-    virtual AudioStreamIn* openInputStream(
-            uint32_t devices,
-            int *format,
-            uint32_t *channels,
-            uint32_t *sampleRate,
-            status_t *status,
-            AudioSystem::audio_in_acoustics acoustics);
-    virtual    void        closeInputStream(AudioStreamIn* in);
-
-            void            closeOutputStream(AudioStreamOutGeneric* out);
-            void            closeInputStream(AudioStreamInGeneric* in);
-protected:
-    virtual status_t        dump(int fd, const Vector<String16>& args);
-
-private:
-    status_t                dumpInternals(int fd, const Vector<String16>& args);
-
-    Mutex                   mLock;
-    AudioStreamOutGeneric   *mOutput;
-    AudioStreamInGeneric    *mInput;
-    int                     mFd;
-    bool                    mMicMute;
-};
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
-
-#endif // ANDROID_AUDIO_HARDWARE_GENERIC_H
diff --git a/services/audioflinger/AudioHardwareInterface.cpp b/services/audioflinger/AudioHardwareInterface.cpp
deleted file mode 100644
index f58e4c0..0000000
--- a/services/audioflinger/AudioHardwareInterface.cpp
+++ /dev/null
@@ -1,183 +0,0 @@
-/*
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License"); 
-** you may not use this file except in compliance with the License. 
-** You may obtain a copy of the License at 
-**
-**     http://www.apache.org/licenses/LICENSE-2.0 
-**
-** Unless required by applicable law or agreed to in writing, software 
-** distributed under the License is distributed on an "AS IS" BASIS, 
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 
-** See the License for the specific language governing permissions and 
-** limitations under the License.
-*/
-
-#include <cutils/properties.h>
-#include <string.h>
-#include <unistd.h>
-//#define LOG_NDEBUG 0
-
-#define LOG_TAG "AudioHardwareInterface"
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include "AudioHardwareStub.h"
-#include "AudioHardwareGeneric.h"
-#ifdef WITH_A2DP
-#include "A2dpAudioInterface.h"
-#endif
-
-#ifdef ENABLE_AUDIO_DUMP
-#include "AudioDumpInterface.h"
-#endif
-
-
-// change to 1 to log routing calls
-#define LOG_ROUTING_CALLS 1
-
-namespace android {
-
-#if LOG_ROUTING_CALLS
-static const char* routingModeStrings[] =
-{
-    "OUT OF RANGE",
-    "INVALID",
-    "CURRENT",
-    "NORMAL",
-    "RINGTONE",
-    "IN_CALL",
-    "IN_COMMUNICATION"
-};
-
-static const char* routeNone = "NONE";
-
-static const char* displayMode(int mode)
-{
-    if ((mode < AudioSystem::MODE_INVALID) || (mode >= AudioSystem::NUM_MODES))
-        return routingModeStrings[0];
-    return routingModeStrings[mode+3];
-}
-#endif
-
-// ----------------------------------------------------------------------------
-
-AudioHardwareInterface* AudioHardwareInterface::create()
-{
-    /*
-     * FIXME: This code needs to instantiate the correct audio device
-     * interface. For now - we use compile-time switches.
-     */
-    AudioHardwareInterface* hw = 0;
-    char value[PROPERTY_VALUE_MAX];
-
-#ifdef GENERIC_AUDIO
-    hw = new AudioHardwareGeneric();
-#else
-    // if running in emulation - use the emulator driver
-    if (property_get("ro.kernel.qemu", value, 0)) {
-        LOGD("Running in emulation - using generic audio driver");
-        hw = new AudioHardwareGeneric();
-    }
-    else {
-        LOGV("Creating Vendor Specific AudioHardware");
-        hw = createAudioHardware();
-    }
-#endif
-    if (hw->initCheck() != NO_ERROR) {
-        LOGW("Using stubbed audio hardware. No sound will be produced.");
-        delete hw;
-        hw = new AudioHardwareStub();
-    }
-    
-#ifdef WITH_A2DP
-    hw = new A2dpAudioInterface(hw);
-#endif
-
-#ifdef ENABLE_AUDIO_DUMP
-    // This code adds a record of buffers in a file to write calls made by AudioFlinger.
-    // It replaces the current AudioHardwareInterface object by an intermediate one which
-    // will record buffers in a file (after sending them to hardware) for testing purpose.
-    // This feature is enabled by defining symbol ENABLE_AUDIO_DUMP.
-    // The output file is set with setParameters("test_cmd_file_name=<name>"). Pause are not recorded in the file.
-    LOGV("opening PCM dump interface");
-    hw = new AudioDumpInterface(hw);    // replace interface
-#endif
-    return hw;
-}
-
-AudioStreamOut::~AudioStreamOut()
-{
-}
-
-AudioStreamIn::~AudioStreamIn() {}
-
-AudioHardwareBase::AudioHardwareBase()
-{
-    mMode = 0;
-}
-
-status_t AudioHardwareBase::setMode(int mode)
-{
-#if LOG_ROUTING_CALLS
-    LOGD("setMode(%s)", displayMode(mode));
-#endif
-    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
-        return BAD_VALUE;
-    if (mMode == mode)
-        return ALREADY_EXISTS;
-    mMode = mode;
-    return NO_ERROR;
-}
-
-// default implementation
-status_t AudioHardwareBase::setParameters(const String8& keyValuePairs)
-{
-    return NO_ERROR;
-}
-
-// default implementation
-String8 AudioHardwareBase::getParameters(const String8& keys)
-{
-    AudioParameter param = AudioParameter(keys);
-    return param.toString();
-}
-
-// default implementation
-size_t AudioHardwareBase::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
-{
-    if (sampleRate != 8000) {
-        LOGW("getInputBufferSize bad sampling rate: %d", sampleRate);
-        return 0;
-    }
-    if (format != AudioSystem::PCM_16_BIT) {
-        LOGW("getInputBufferSize bad format: %d", format);
-        return 0;
-    }
-    if (channelCount != 1) {
-        LOGW("getInputBufferSize bad channel count: %d", channelCount);
-        return 0;
-    }
-
-    return 320;
-}
-
-status_t AudioHardwareBase::dumpState(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-    snprintf(buffer, SIZE, "AudioHardwareBase::dumpState\n");
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tmMode: %d\n", mMode);
-    result.append(buffer);
-    ::write(fd, result.string(), result.size());
-    dump(fd, args);  // Dump the state of the concrete child.
-    return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
diff --git a/services/audioflinger/AudioHardwareStub.cpp b/services/audioflinger/AudioHardwareStub.cpp
deleted file mode 100644
index d481150..0000000
--- a/services/audioflinger/AudioHardwareStub.cpp
+++ /dev/null
@@ -1,209 +0,0 @@
-/* //device/servers/AudioFlinger/AudioHardwareStub.cpp
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <stdlib.h>
-#include <unistd.h>
-#include <utils/String8.h>
-
-#include "AudioHardwareStub.h"
-#include <media/AudioRecord.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-AudioHardwareStub::AudioHardwareStub() : mMicMute(false)
-{
-}
-
-AudioHardwareStub::~AudioHardwareStub()
-{
-}
-
-status_t AudioHardwareStub::initCheck()
-{
-    return NO_ERROR;
-}
-
-AudioStreamOut* AudioHardwareStub::openOutputStream(
-        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
-{
-    AudioStreamOutStub* out = new AudioStreamOutStub();
-    status_t lStatus = out->set(format, channels, sampleRate);
-    if (status) {
-        *status = lStatus;
-    }
-    if (lStatus == NO_ERROR)
-        return out;
-    delete out;
-    return 0;
-}
-
-void AudioHardwareStub::closeOutputStream(AudioStreamOut* out)
-{
-    delete out;
-}
-
-AudioStreamIn* AudioHardwareStub::openInputStream(
-        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate,
-        status_t *status, AudioSystem::audio_in_acoustics acoustics)
-{
-    // check for valid input source
-    if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) {
-        return 0;
-    }
-
-    AudioStreamInStub* in = new AudioStreamInStub();
-    status_t lStatus = in->set(format, channels, sampleRate, acoustics);
-    if (status) {
-        *status = lStatus;
-    }
-    if (lStatus == NO_ERROR)
-        return in;
-    delete in;
-    return 0;
-}
-
-void AudioHardwareStub::closeInputStream(AudioStreamIn* in)
-{
-    delete in;
-}
-
-status_t AudioHardwareStub::setVoiceVolume(float volume)
-{
-    return NO_ERROR;
-}
-
-status_t AudioHardwareStub::setMasterVolume(float volume)
-{
-    return NO_ERROR;
-}
-
-status_t AudioHardwareStub::dumpInternals(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-    result.append("AudioHardwareStub::dumpInternals\n");
-    snprintf(buffer, SIZE, "\tmMicMute: %s\n", mMicMute? "true": "false");
-    result.append(buffer);
-    ::write(fd, result.string(), result.size());
-    return NO_ERROR;
-}
-
-status_t AudioHardwareStub::dump(int fd, const Vector<String16>& args)
-{
-    dumpInternals(fd, args);
-    return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-
-status_t AudioStreamOutStub::set(int *pFormat, uint32_t *pChannels, uint32_t *pRate)
-{
-    if (pFormat) *pFormat = format();
-    if (pChannels) *pChannels = channels();
-    if (pRate) *pRate = sampleRate();
-
-    return NO_ERROR;
-}
-
-ssize_t AudioStreamOutStub::write(const void* buffer, size_t bytes)
-{
-    // fake timing for audio output
-    usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate());
-    return bytes;
-}
-
-status_t AudioStreamOutStub::standby()
-{
-    return NO_ERROR;
-}
-
-status_t AudioStreamOutStub::dump(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-    snprintf(buffer, SIZE, "AudioStreamOutStub::dump\n");
-    snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
-    snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
-    snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
-    snprintf(buffer, SIZE, "\tformat: %d\n", format());
-    result.append(buffer);
-    ::write(fd, result.string(), result.size());
-    return NO_ERROR;
-}
-
-String8 AudioStreamOutStub::getParameters(const String8& keys)
-{
-    AudioParameter param = AudioParameter(keys);
-    return param.toString();
-}
-
-status_t AudioStreamOutStub::getRenderPosition(uint32_t *dspFrames)
-{
-    return INVALID_OPERATION;
-}
-
-// ----------------------------------------------------------------------------
-
-status_t AudioStreamInStub::set(int *pFormat, uint32_t *pChannels, uint32_t *pRate,
-                AudioSystem::audio_in_acoustics acoustics)
-{
-    return NO_ERROR;
-}
-
-ssize_t AudioStreamInStub::read(void* buffer, ssize_t bytes)
-{
-    // fake timing for audio input
-    usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate());
-    memset(buffer, 0, bytes);
-    return bytes;
-}
-
-status_t AudioStreamInStub::dump(int fd, const Vector<String16>& args)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-    snprintf(buffer, SIZE, "AudioStreamInStub::dump\n");
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
-    result.append(buffer);
-    snprintf(buffer, SIZE, "\tformat: %d\n", format());
-    result.append(buffer);
-    ::write(fd, result.string(), result.size());
-    return NO_ERROR;
-}
-
-String8 AudioStreamInStub::getParameters(const String8& keys)
-{
-    AudioParameter param = AudioParameter(keys);
-    return param.toString();
-}
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
diff --git a/services/audioflinger/AudioHardwareStub.h b/services/audioflinger/AudioHardwareStub.h
deleted file mode 100644
index 06a29de..0000000
--- a/services/audioflinger/AudioHardwareStub.h
+++ /dev/null
@@ -1,106 +0,0 @@
-/* //device/servers/AudioFlinger/AudioHardwareStub.h
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#ifndef ANDROID_AUDIO_HARDWARE_STUB_H
-#define ANDROID_AUDIO_HARDWARE_STUB_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <hardware_legacy/AudioHardwareBase.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-class AudioStreamOutStub : public AudioStreamOut {
-public:
-    virtual status_t    set(int *pFormat, uint32_t *pChannels, uint32_t *pRate);
-    virtual uint32_t    sampleRate() const { return 44100; }
-    virtual size_t      bufferSize() const { return 4096; }
-    virtual uint32_t    channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
-    virtual int         format() const { return AudioSystem::PCM_16_BIT; }
-    virtual uint32_t    latency() const { return 0; }
-    virtual status_t    setVolume(float left, float right) { return NO_ERROR; }
-    virtual ssize_t     write(const void* buffer, size_t bytes);
-    virtual status_t    standby();
-    virtual status_t    dump(int fd, const Vector<String16>& args);
-    virtual status_t    setParameters(const String8& keyValuePairs) { return NO_ERROR;}
-    virtual String8     getParameters(const String8& keys);
-    virtual status_t    getRenderPosition(uint32_t *dspFrames);
-};
-
-class AudioStreamInStub : public AudioStreamIn {
-public:
-    virtual status_t    set(int *pFormat, uint32_t *pChannels, uint32_t *pRate, AudioSystem::audio_in_acoustics acoustics);
-    virtual uint32_t    sampleRate() const { return 8000; }
-    virtual size_t      bufferSize() const { return 320; }
-    virtual uint32_t    channels() const { return AudioSystem::CHANNEL_IN_MONO; }
-    virtual int         format() const { return AudioSystem::PCM_16_BIT; }
-    virtual status_t    setGain(float gain) { return NO_ERROR; }
-    virtual ssize_t     read(void* buffer, ssize_t bytes);
-    virtual status_t    dump(int fd, const Vector<String16>& args);
-    virtual status_t    standby() { return NO_ERROR; }
-    virtual status_t    setParameters(const String8& keyValuePairs) { return NO_ERROR;}
-    virtual String8     getParameters(const String8& keys);
-    virtual unsigned int  getInputFramesLost() const { return 0; }
-};
-
-class AudioHardwareStub : public  AudioHardwareBase
-{
-public:
-                        AudioHardwareStub();
-    virtual             ~AudioHardwareStub();
-    virtual status_t    initCheck();
-    virtual status_t    setVoiceVolume(float volume);
-    virtual status_t    setMasterVolume(float volume);
-
-    // mic mute
-    virtual status_t    setMicMute(bool state) { mMicMute = state;  return  NO_ERROR; }
-    virtual status_t    getMicMute(bool* state) { *state = mMicMute ; return NO_ERROR; }
-
-    // create I/O streams
-    virtual AudioStreamOut* openOutputStream(
-                                uint32_t devices,
-                                int *format=0,
-                                uint32_t *channels=0,
-                                uint32_t *sampleRate=0,
-                                status_t *status=0);
-    virtual    void        closeOutputStream(AudioStreamOut* out);
-
-    virtual AudioStreamIn* openInputStream(
-                                uint32_t devices,
-                                int *format,
-                                uint32_t *channels,
-                                uint32_t *sampleRate,
-                                status_t *status,
-                                AudioSystem::audio_in_acoustics acoustics);
-    virtual    void        closeInputStream(AudioStreamIn* in);
-
-protected:
-    virtual status_t    dump(int fd, const Vector<String16>& args);
-
-            bool        mMicMute;
-private:
-    status_t            dumpInternals(int fd, const Vector<String16>& args);
-};
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
-
-#endif // ANDROID_AUDIO_HARDWARE_STUB_H
diff --git a/services/audioflinger/AudioPolicyManagerBase.cpp b/services/audioflinger/AudioPolicyManagerBase.cpp
deleted file mode 100644
index 32d92dc..0000000
--- a/services/audioflinger/AudioPolicyManagerBase.cpp
+++ /dev/null
@@ -1,2287 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "AudioPolicyManagerBase"
-//#define LOG_NDEBUG 0
-#include <utils/Log.h>
-#include <hardware_legacy/AudioPolicyManagerBase.h>
-#include <media/mediarecorder.h>
-#include <math.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);
-                }
-            }
-            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 = "";
-                }
-            }
-            } 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
-        checkA2dpSuspend();
-        checkOutputForAllStrategies();
-        // 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 (isInCall()) {
-        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 (!isStateInCall(oldState) && isStateInCall(state)) {
-        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 (isStateInCall(oldState) && !isStateInCall(state)) {
-        LOGV("  Exiting call in setPhoneState()");
-        // force routing command to audio hardware when exiting a call
-        // even if no device change is needed
-        force = true;
-    } else if (isStateInCall(state) && (state != oldState)) {
-        LOGV("  Switching between telephony and VoIP in setPhoneState()");
-        // force routing command to audio hardware when switching between telephony and VoIP
-        // even if no device change is needed
-        force = true;
-    }
-
-    // check for device and output changes triggered by new phone state
-    newDevice = getNewDevice(mHardwareOutput, false);
-#ifdef WITH_A2DP
-    checkA2dpSuspend();
-    checkOutputForAllStrategies();
-#endif
-    updateDeviceForStrategy();
-
-    AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
-
-    // force routing command to audio hardware when ending call
-    // even if no device change is needed
-    if (isStateInCall(oldState) && newDevice == 0) {
-        newDevice = hwOutputDesc->device();
-    }
-
-    // when changing from ring tone to in call mode, mute the ringing tone
-    // immediately and delay the route change to avoid sending the ring tone
-    // tail into the earpiece or headset.
-    int delayMs = 0;
-    if (isStateInCall(state) && oldState == AudioSystem::MODE_RINGTONE) {
-        // delay the device change command by twice the output latency to have some margin
-        // and be sure that audio buffers not yet affected by the mute are out when
-        // we actually apply the route change
-        delayMs = hwOutputDesc->mLatency*2;
-        setStreamMute(AudioSystem::RING, true, mHardwareOutput);
-    }
-
-    // change routing is necessary
-    setOutputDevice(mHardwareOutput, newDevice, force, delayMs);
-
-    // if entering in call state, handle special case of active streams
-    // pertaining to sonification strategy see handleIncallSonification()
-    if (isStateInCall(state)) {
-        LOGV("setPhoneState() in call state management: new state is %d", state);
-        // unmute the ringing tone after a sufficient delay if it was muted before
-        // setting output device above
-        if (oldState == AudioSystem::MODE_RINGTONE) {
-            setStreamMute(AudioSystem::RING, false, mHardwareOutput, MUTE_TIME_MS);
-        }
-        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
-            handleIncallSonification(stream, true, true);
-        }
-    }
-
-    // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
-    if (state == AudioSystem::MODE_RINGTONE &&
-        isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
-        mLimitRingtoneVolume = true;
-    } else {
-        mLimitRingtoneVolume = false;
-    }
-}
-
-void AudioPolicyManagerBase::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);
-
-    bool forceVolumeReeval = false;
-    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;
-        }
-        forceVolumeReeval = true;
-        mForceUse[usage] = config;
-        break;
-    case AudioSystem::FOR_MEDIA:
-        if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP &&
-            config != AudioSystem::FORCE_WIRED_ACCESSORY &&
-            config != AudioSystem::FORCE_ANALOG_DOCK &&
-            config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE) {
-            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 &&
-            config != AudioSystem::FORCE_ANALOG_DOCK &&
-            config != AudioSystem::FORCE_DIGITAL_DOCK) {
-            LOGW("setForceUse() invalid config %d for FOR_DOCK", config);
-        }
-        forceVolumeReeval = true;
-        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
-    checkA2dpSuspend();
-    checkOutputForAllStrategies();
-#endif
-    updateDeviceForStrategy();
-    setOutputDevice(mHardwareOutput, newDevice);
-    if (forceVolumeReeval) {
-        applyStreamVolumes(mHardwareOutput, newDevice, 0, true);
-    }
-
-    audio_io_handle_t activeInput = getActiveInput();
-    if (activeInput != 0) {
-        AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
-        newDevice = getDeviceForInputSource(inputDesc->mInputSource);
-        if (newDevice != inputDesc->mDevice) {
-            LOGV("setForceUse() changing device from %x to %x for input %d",
-                    inputDesc->mDevice, newDevice, activeInput);
-            inputDesc->mDevice = newDevice;
-            AudioParameter param = AudioParameter();
-            param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
-            mpClientInterface->setParameters(activeInput, param.toString());
-        }
-    }
-
-}
-
-AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
-{
-    return mForceUse[usage];
-}
-
-void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value)
-{
-    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 required by specified parameters
-    if (needsDirectOuput(stream, samplingRate, format, channels, flags, device)) {
-
-        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] = 0;
-        outputDesc->mStopTime[stream] = 0;
-        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,
-                                             int session)
-{
-    LOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
-    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 (isInCall()) {
-        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,
-                                            int session)
-{
-    LOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
-    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 (isInCall()) {
-        handleIncallSonification(stream, false, false);
-    }
-
-    if (outputDesc->mRefCount[stream] > 0) {
-        // decrement usage count of this stream on the output
-        outputDesc->changeRefCount(stream, -1);
-        // store time at which the stream was stopped - see isStreamActive()
-        outputDesc->mStopTime[stream] = systemTime();
-
-        setOutputDevice(output, getNewDevice(output), false, outputDesc->mLatency*2);
-
-#ifdef WITH_A2DP
-        if (mA2dpOutput != 0 && !a2dpUsedForSonification() &&
-                strategy == STRATEGY_SONIFICATION) {
-            setStrategyMute(STRATEGY_MEDIA,
-                            false,
-                            mA2dpOutput,
-                            mOutputs.valueFor(mHardwareOutput)->mLatency*2);
-        }
-#endif
-        if (output != mHardwareOutput) {
-            setOutputDevice(mHardwareOutput, getNewDevice(mHardwareOutput), true);
-        }
-        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);
-
-    param.addInt(String8(AudioParameter::keyInputSource), (int)inputDesc->mInputSource);
-    LOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
-
-    mpClientInterface->setParameters(input, param.toString());
-
-    inputDesc->mRefCount = 1;
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input)
-{
-    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;
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getOutputForEffect(effect_descriptor_t *desc)
-{
-    LOGV("getOutputForEffect()");
-    // apply simple rule where global effects are attached to the same output as MUSIC streams
-    return getOutput(AudioSystem::MUSIC);
-}
-
-status_t AudioPolicyManagerBase::registerEffect(effect_descriptor_t *desc,
-                                audio_io_handle_t output,
-                                uint32_t strategy,
-                                int session,
-                                int id)
-{
-    ssize_t index = mOutputs.indexOfKey(output);
-    if (index < 0) {
-        LOGW("registerEffect() unknown output %d", output);
-        return INVALID_OPERATION;
-    }
-
-    if (mTotalEffectsCpuLoad + desc->cpuLoad > getMaxEffectsCpuLoad()) {
-        LOGW("registerEffect() CPU Load limit exceeded for Fx %s, CPU %f MIPS",
-                desc->name, (float)desc->cpuLoad/10);
-        return INVALID_OPERATION;
-    }
-    if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
-        LOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
-                desc->name, desc->memoryUsage);
-        return INVALID_OPERATION;
-    }
-    mTotalEffectsCpuLoad += desc->cpuLoad;
-    mTotalEffectsMemory += desc->memoryUsage;
-    LOGV("registerEffect() effect %s, output %d, strategy %d session %d id %d",
-            desc->name, output, strategy, session, id);
-
-    LOGV("registerEffect() CPU %d, memory %d", desc->cpuLoad, desc->memoryUsage);
-    LOGV("  total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory);
-
-    EffectDescriptor *pDesc = new EffectDescriptor();
-    memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t));
-    pDesc->mOutput = output;
-    pDesc->mStrategy = (routing_strategy)strategy;
-    pDesc->mSession = session;
-    mEffects.add(id, pDesc);
-
-    return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::unregisterEffect(int id)
-{
-    ssize_t index = mEffects.indexOfKey(id);
-    if (index < 0) {
-        LOGW("unregisterEffect() unknown effect ID %d", id);
-        return INVALID_OPERATION;
-    }
-
-    EffectDescriptor *pDesc = mEffects.valueAt(index);
-
-    if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) {
-        LOGW("unregisterEffect() CPU load %d too high for total %d",
-                pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
-        pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
-    }
-    mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad;
-    if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) {
-        LOGW("unregisterEffect() memory %d too big for total %d",
-                pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
-        pDesc->mDesc.memoryUsage = mTotalEffectsMemory;
-    }
-    mTotalEffectsMemory -= pDesc->mDesc.memoryUsage;
-    LOGV("unregisterEffect() effect %s, ID %d, CPU %d, memory %d",
-            pDesc->mDesc.name, id, pDesc->mDesc.cpuLoad, pDesc->mDesc.memoryUsage);
-    LOGV("  total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory);
-
-    mEffects.removeItem(id);
-    delete pDesc;
-
-    return NO_ERROR;
-}
-
-bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const
-{
-    nsecs_t sysTime = systemTime();
-    for (size_t i = 0; i < mOutputs.size(); i++) {
-        if (mOutputs.valueAt(i)->mRefCount[stream] != 0 ||
-            ns2ms(sysTime - mOutputs.valueAt(i)->mStopTime[stream]) < inPastMs) {
-            return true;
-        }
-    }
-    return false;
-}
-
-
-status_t AudioPolicyManagerBase::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " 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));
-    }
-
-    snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
-            (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
-    write(fd, buffer, strlen(buffer));
-
-    snprintf(buffer, SIZE, "Registered effects:\n");
-    write(fd, buffer, strlen(buffer));
-    for (size_t i = 0; i < mEffects.size(); i++) {
-        snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
-        write(fd, buffer, strlen(buffer));
-        mEffects.valueAt(i)->dump(fd);
-    }
-
-
-    return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManagerBase
-// ----------------------------------------------------------------------------
-
-AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
-    :
-#ifdef AUDIO_POLICY_TEST
-    Thread(false),
-#endif //AUDIO_POLICY_TEST
-    mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0),
-    mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
-    mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
-    mA2dpSuspended(false)
-{
-    mpClientInterface = clientInterface;
-
-    for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
-        mForceUse[i] = AudioSystem::FORCE_NONE;
-    }
-
-    initializeVolumeCurves();
-
-    // 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);
-        //TODO: configure audio effect output stage here
-    }
-
-    updateDeviceForStrategy();
-#ifdef AUDIO_POLICY_TEST
-    if (mHardwareOutput != 0) {
-        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();
-}
-
-status_t AudioPolicyManagerBase::initCheck()
-{
-    return (mHardwareOutput == 0) ? NO_INIT : NO_ERROR;
-}
-
-#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);
-
-        //TODO: configure audio effect output stage here
-
-        // 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 (!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);
-        }
-    }
-
-    mA2dpSuspended = false;
-
-    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 strategy to avoid outputting 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, MUTE_TIME_MS);
-
-    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 = "";
-    mA2dpSuspended = false;
-    return NO_ERROR;
-}
-
-void AudioPolicyManagerBase::closeA2dpOutputs()
-{
-
-    LOGV("setDeviceConnectionState() closing A2DP and duplicated output!");
-
-    if (mDuplicatedOutput != 0) {
-        AudioOutputDescriptor *dupOutputDesc = mOutputs.valueFor(mDuplicatedOutput);
-        AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
-        // As all active tracks on duplicated output will be deleted,
-        // and as they were also referenced on hardware output, the reference
-        // count for their stream type must be adjusted accordingly on
-        // hardware output.
-        for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
-            int refCount = dupOutputDesc->mRefCount[i];
-            hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
-        }
-
-        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 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));
-    audio_io_handle_t srcOutput = 0;
-    audio_io_handle_t dstOutput = 0;
-
-    if (a2dpWasUsed && !a2dpIsUsed) {
-        bool dupUsed = a2dpUsedForSonification() && a2dpWasUsed && (AudioSystem::popCount(prevDevice) == 2);
-        dstOutput = mHardwareOutput;
-        if (dupUsed) {
-            LOGV("checkOutputForStrategy() moving strategy %d from duplicated", strategy);
-            srcOutput = mDuplicatedOutput;
-        } else {
-            LOGV("checkOutputForStrategy() moving strategy %d from a2dp", strategy);
-            srcOutput = mA2dpOutput;
-        }
-    }
-    if (a2dpIsUsed && !a2dpWasUsed) {
-        bool dupUsed = a2dpUsedForSonification() && a2dpIsUsed && (AudioSystem::popCount(curDevice) == 2);
-        srcOutput = mHardwareOutput;
-        if (dupUsed) {
-            LOGV("checkOutputForStrategy() moving strategy %d to duplicated", strategy);
-            dstOutput = mDuplicatedOutput;
-        } else {
-            LOGV("checkOutputForStrategy() moving strategy %d to a2dp", strategy);
-            dstOutput = mA2dpOutput;
-        }
-    }
-
-    if (srcOutput != 0 && dstOutput != 0) {
-        // Move effects associated to this strategy from previous output to new output
-        for (size_t i = 0; i < mEffects.size(); i++) {
-            EffectDescriptor *desc = mEffects.valueAt(i);
-            if (desc->mSession != AudioSystem::SESSION_OUTPUT_STAGE &&
-                    desc->mStrategy == strategy &&
-                    desc->mOutput == srcOutput) {
-                LOGV("checkOutputForStrategy() moving effect %d to output %d", mEffects.keyAt(i), dstOutput);
-                mpClientInterface->moveEffects(desc->mSession, srcOutput, dstOutput);
-                desc->mOutput = dstOutput;
-            }
-        }
-        // Move tracks associated to this strategy from previous output to new output
-        for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
-            if (getStrategy((AudioSystem::stream_type)i) == strategy) {
-                mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, dstOutput);
-            }
-        }
-    }
-}
-
-void AudioPolicyManagerBase::checkOutputForAllStrategies()
-{
-    checkOutputForStrategy(STRATEGY_PHONE);
-    checkOutputForStrategy(STRATEGY_SONIFICATION);
-    checkOutputForStrategy(STRATEGY_MEDIA);
-    checkOutputForStrategy(STRATEGY_DTMF);
-}
-
-void AudioPolicyManagerBase::checkA2dpSuspend()
-{
-    // suspend A2DP output if:
-    //      (NOT already suspended) &&
-    //      ((SCO device is connected &&
-    //       (forced usage for communication || for record is SCO))) ||
-    //      (phone state is ringing || in call)
-    //
-    // restore A2DP output if:
-    //      (Already suspended) &&
-    //      ((SCO device is NOT connected ||
-    //       (forced usage NOT for communication && NOT for record is SCO))) &&
-    //      (phone state is NOT ringing && NOT in call)
-    //
-    if (mA2dpOutput == 0) {
-        return;
-    }
-
-    if (mA2dpSuspended) {
-        if (((mScoDeviceAddress == "") ||
-             ((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) &&
-              (mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) &&
-             ((mPhoneState != AudioSystem::MODE_IN_CALL) &&
-              (mPhoneState != AudioSystem::MODE_RINGTONE))) {
-
-            mpClientInterface->restoreOutput(mA2dpOutput);
-            mA2dpSuspended = false;
-        }
-    } else {
-        if (((mScoDeviceAddress != "") &&
-             ((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
-              (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) ||
-             ((mPhoneState == AudioSystem::MODE_IN_CALL) ||
-              (mPhoneState == AudioSystem::MODE_RINGTONE))) {
-
-            mpClientInterface->suspendOutput(mA2dpOutput);
-            mA2dpSuspended = true;
-        }
-    }
-}
-
-
-#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 (isInCall() ||
-        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;
-}
-
-uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) {
-    return (uint32_t)getStrategy(stream);
-}
-
-uint32_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) {
-    uint32_t devices;
-    // By checking the range of stream before calling getStrategy, we avoid
-    // getStrategy's behavior for invalid streams.  getStrategy would do a LOGE
-    // and then return STRATEGY_MEDIA, but we want to return the empty set.
-    if (stream < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
-        devices = 0;
-    } else {
-        AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream);
-        devices = getDeviceForStrategy(strategy, true);
-    }
-    return devices;
-}
-
-AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy(
-        AudioSystem::stream_type stream) {
-    // stream to strategy mapping
-    switch (stream) {
-    case AudioSystem::VOICE_CALL:
-    case AudioSystem::BLUETOOTH_SCO:
-        return STRATEGY_PHONE;
-    case AudioSystem::RING:
-    case AudioSystem::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 (!isInCall()) {
-            // 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 (!isInCall() || 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;
-#ifdef WITH_A2DP
-            // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
-            if (!isInCall() && !mA2dpSuspended) {
-                device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
-                if (device) break;
-                device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
-                if (device) break;
-            }
-#endif
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_EARPIECE;
-            if (device == 0) {
-                LOGE("getDeviceForStrategy() earpiece device not found");
-            }
-            break;
-
-        case AudioSystem::FORCE_SPEAKER:
-#ifdef WITH_A2DP
-            // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
-            // A2DP speaker when forcing to speaker output
-            if (!isInCall() && !mA2dpSuspended) {
-                device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
-                if (device) break;
-            }
-#endif
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
-            if (device) break;
-            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET;
-            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 (isInCall()) {
-            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_WIRED_HEADPHONE;
-        if (device2 == 0) {
-            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
-        }
-#ifdef WITH_A2DP
-        if ((mA2dpOutput != 0) && !mA2dpSuspended &&
-                (strategy != STRATEGY_SONIFICATION || a2dpUsedForSonification())) {
-            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_AUX_DIGITAL;
-        }
-        if (device2 == 0) {
-            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
-        }
-        if (device2 == 0) {
-            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_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 == mA2dpOutput) {
-        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);
-    }
-
-    // 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);
-
-    // 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:
-    case AUDIO_SOURCE_VOICE_COMMUNICATION:
-        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::volIndexToAmpl(uint32_t device, const StreamDescriptor& streamDesc,
-        int indexInUi) {
-    // the volume index in the UI is relative to the min and max volume indices for this stream type
-    int nbSteps = 1 + streamDesc.mVolIndex[StreamDescriptor::VOLMAX] -
-            streamDesc.mVolIndex[StreamDescriptor::VOLMIN];
-    int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
-            (streamDesc.mIndexMax - streamDesc.mIndexMin);
-
-    // find what part of the curve this index volume belongs to, or if it's out of bounds
-    int segment = 0;
-    if (volIdx < streamDesc.mVolIndex[StreamDescriptor::VOLMIN]) {         // out of bounds
-        return 0.0f;
-    } else if (volIdx < streamDesc.mVolIndex[StreamDescriptor::VOLKNEE1]) {
-        segment = 0;
-    } else if (volIdx < streamDesc.mVolIndex[StreamDescriptor::VOLKNEE2]) {
-        segment = 1;
-    } else if (volIdx <= streamDesc.mVolIndex[StreamDescriptor::VOLMAX]) {
-        segment = 2;
-    } else {                                                               // out of bounds
-        return 1.0f;
-    }
-
-    // linear interpolation in the attenuation table in dB
-    float decibels = streamDesc.mVolDbAtt[segment] +
-            ((float)(volIdx - streamDesc.mVolIndex[segment])) *
-                ( (streamDesc.mVolDbAtt[segment+1] - streamDesc.mVolDbAtt[segment]) /
-                    ((float)(streamDesc.mVolIndex[segment+1] - streamDesc.mVolIndex[segment])) );
-
-    float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
-
-    LOGV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
-            streamDesc.mVolIndex[segment], volIdx, streamDesc.mVolIndex[segment+1],
-            streamDesc.mVolDbAtt[segment], decibels, streamDesc.mVolDbAtt[segment+1],
-            amplification);
-
-    return amplification;
-}
-
-void AudioPolicyManagerBase::initializeVolumeCurves() {
-    // initialize the volume curves to a (-49.5 - 0 dB) attenuation in 0.5dB steps
-    for (int i=0 ; i< AudioSystem::NUM_STREAM_TYPES ; i++) {
-        mStreams[i].mVolIndex[StreamDescriptor::VOLMIN] = 1;
-        mStreams[i].mVolDbAtt[StreamDescriptor::VOLMIN] = -49.5f;
-        mStreams[i].mVolIndex[StreamDescriptor::VOLKNEE1] = 33;
-        mStreams[i].mVolDbAtt[StreamDescriptor::VOLKNEE1] = -33.5f;
-        mStreams[i].mVolIndex[StreamDescriptor::VOLKNEE2] = 66;
-        mStreams[i].mVolDbAtt[StreamDescriptor::VOLKNEE2] = -17.0f;
-        // here we use 100 steps to avoid rounding errors
-        // when computing the volume in volIndexToAmpl()
-        mStreams[i].mVolIndex[StreamDescriptor::VOLMAX] = 100;
-        mStreams[i].mVolDbAtt[StreamDescriptor::VOLMAX] = 0.0f;
-    }
-
-    // Modification for music: more attenuation for lower volumes, finer steps at high volumes
-    mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLMIN] = 1;
-    mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLMIN] = -58.0f;
-    mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLKNEE1] = 20;
-    mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLKNEE1] = -40.0f;
-    mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLKNEE2] = 60;
-    mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLKNEE2] = -17.0f;
-    mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLMAX] = 100;
-    mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLMAX] = 0.0f;
-}
-
-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();
-    }
-
-    // if volume is not 0 (not muted), force media volume to max on digital output
-    if (stream == AudioSystem::MUSIC &&
-        index != mStreams[stream].mIndexMin &&
-        device == AudioSystem::DEVICE_OUT_AUX_DIGITAL) {
-        return 1.0;
-    }
-
-    volume = volIndexToAmpl(device, streamDesc, index);
-
-    // if a headset is connected, apply the following rules to ring tones and notifications
-    // to avoid sound level bursts in user's ears:
-    // - always attenuate ring tones and notifications volume by 6dB
-    // - if music is playing, always limit the volume to current music volume,
-    // with a minimum threshold at -36dB so that notification is always perceived.
-    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) ||
-         (stream == AudioSystem::SYSTEM)) &&
-        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);
-    // We actually change the volume if:
-    // - the float value returned by computeVolume() changed
-    // - the force flag is set
-    if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
-            force) {
-        mOutputs.valueFor(output)->mCurVolume[stream] = volume;
-        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) {
-            // 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;
-            // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
-            // enabled
-            if (stream == AudioSystem::BLUETOOTH_SCO) {
-                mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs);
-            }
-        }
-
-        mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
-    }
-
-    if (stream == AudioSystem::VOICE_CALL ||
-        stream == AudioSystem::BLUETOOTH_SCO) {
-        float voiceVolume;
-        // Force voice volume to max for bluetooth SCO as volume is managed by the headset
-        if (stream == AudioSystem::VOICE_CALL) {
-            voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
-        } else {
-            voiceVolume = 1.0;
-        }
-
-        if (voiceVolume != mLastVoiceVolume && output == mHardwareOutput) {
-            mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
-            mLastVoiceVolume = voiceVolume;
-        }
-    }
-
-    return NO_ERROR;
-}
-
-void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs, bool force)
-{
-    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, force);
-    }
-}
-
-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();
-                }
-            }
-        }
-    }
-}
-
-bool AudioPolicyManagerBase::isInCall()
-{
-    return isStateInCall(mPhoneState);
-}
-
-bool AudioPolicyManagerBase::isStateInCall(int state) {
-    return ((state == AudioSystem::MODE_IN_CALL) ||
-            (state == AudioSystem::MODE_IN_COMMUNICATION));
-}
-
-bool AudioPolicyManagerBase::needsDirectOuput(AudioSystem::stream_type stream,
-                                    uint32_t samplingRate,
-                                    uint32_t format,
-                                    uint32_t channels,
-                                    AudioSystem::output_flags flags,
-                                    uint32_t device)
-{
-   return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
-          (format != 0 && !AudioSystem::isLinearPCM(format)));
-}
-
-uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad()
-{
-    return MAX_EFFECTS_CPU_LOAD;
-}
-
-uint32_t AudioPolicyManagerBase::getMaxEffectsMemory()
-{
-    return MAX_EFFECTS_MEMORY;
-}
-
-// --- 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;
-        mStopTime[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() delta %d, stream %d, refCount %d", delta, 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),
-      mInputSource(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);
-}
-
-// --- EffectDescriptor class implementation
-
-status_t AudioPolicyManagerBase::EffectDescriptor::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, " Output: %d\n", mOutput);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Session: %d\n", mSession);
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Name: %s\n",  mDesc.name);
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-
-    return NO_ERROR;
-}
-
-
-
-}; // namespace android