legacy: move legacy audio code from frameworks/base here

Change-Id: Ic5da0130af44354dffdf85c30cd99f57c6ee163c
Signed-off-by: Dima Zavin <dima@android.com>

10 files changed, 4726 insertions, 0 deletions

diff --git a/audio/A2dpAudioInterface.cpp b/audio/A2dpAudioInterface.cpp
new file mode 100644
index 0000000..d926cb1
--- /dev/null
+++ b/audio/A2dpAudioInterface.cpp
@@ -0,0 +1,498 @@
+/*
+ * 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/audio/A2dpAudioInterface.h b/audio/A2dpAudioInterface.h
new file mode 100644
index 0000000..dbe2c6a
--- /dev/null
+++ b/audio/A2dpAudioInterface.h
@@ -0,0 +1,138 @@
+/*
+ * 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/audio/AudioDumpInterface.cpp b/audio/AudioDumpInterface.cpp
new file mode 100644
index 0000000..6c11114
--- /dev/null
+++ b/audio/AudioDumpInterface.cpp
@@ -0,0 +1,573 @@
+/* //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/audio/AudioDumpInterface.h b/audio/AudioDumpInterface.h
new file mode 100644
index 0000000..814ce5f
--- /dev/null
+++ b/audio/AudioDumpInterface.h
@@ -0,0 +1,170 @@
+/* //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/audio/AudioHardwareGeneric.cpp b/audio/AudioHardwareGeneric.cpp
new file mode 100644
index 0000000..d63c031
--- /dev/null
+++ b/audio/AudioHardwareGeneric.cpp
@@ -0,0 +1,411 @@
+/*
+**
+** 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/audio/AudioHardwareGeneric.h b/audio/AudioHardwareGeneric.h
new file mode 100644
index 0000000..aa4e78d
--- /dev/null
+++ b/audio/AudioHardwareGeneric.h
@@ -0,0 +1,151 @@
+/*
+**
+** 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/audio/AudioHardwareInterface.cpp b/audio/AudioHardwareInterface.cpp
new file mode 100644
index 0000000..f58e4c0
--- /dev/null
+++ b/audio/AudioHardwareInterface.cpp
@@ -0,0 +1,183 @@
+/*
+**
+** 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/audio/AudioHardwareStub.cpp b/audio/AudioHardwareStub.cpp
new file mode 100644
index 0000000..d481150
--- /dev/null
+++ b/audio/AudioHardwareStub.cpp
@@ -0,0 +1,209 @@
+/* //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/audio/AudioHardwareStub.h b/audio/AudioHardwareStub.h
new file mode 100644
index 0000000..06a29de
--- /dev/null
+++ b/audio/AudioHardwareStub.h
@@ -0,0 +1,106 @@
+/* //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/audio/AudioPolicyManagerBase.cpp b/audio/AudioPolicyManagerBase.cpp
new file mode 100644
index 0000000..32d92dc
--- /dev/null
+++ b/audio/AudioPolicyManagerBase.cpp
@@ -0,0 +1,2287 @@
+/*
+ * 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