diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
new file mode 100644
index 0000000..92c40e9
--- /dev/null
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -0,0 +1,2474 @@
+/* //device/include/server/AudioFlinger/AudioFlinger.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.
+*/
+
+
+#define LOG_TAG "AudioFlinger"
+//#define LOG_NDEBUG 0
+
+#include <math.h>
+#include <signal.h>
+#include <sys/time.h>
+#include <sys/resource.h>
+
+#include <utils/IServiceManager.h>
+#include <utils/Log.h>
+#include <utils/Parcel.h>
+#include <utils/IPCThreadState.h>
+#include <utils/String16.h>
+#include <utils/threads.h>
+
+#include <cutils/properties.h>
+
+#include <media/AudioTrack.h>
+#include <media/AudioRecord.h>
+
+#include <private/media/AudioTrackShared.h>
+
+#include <hardware_legacy/AudioHardwareInterface.h>
+
+#include "AudioMixer.h"
+#include "AudioFlinger.h"
+
+#ifdef WITH_A2DP
+#include "A2dpAudioInterface.h"
+#endif
+
+// ----------------------------------------------------------------------------
+// the sim build doesn't have gettid
+
+#ifndef HAVE_GETTID
+# define gettid getpid
+#endif
+
+// ----------------------------------------------------------------------------
+
+namespace android {
+
+//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
+static const unsigned long kBufferRecoveryInUsecs = 2000;
+static const unsigned long kMaxBufferRecoveryInUsecs = 20000;
+static const float MAX_GAIN = 4096.0f;
+
+// retry counts for buffer fill timeout
+// 50 * ~20msecs = 1 second
+static const int8_t kMaxTrackRetries = 50;
+static const int8_t kMaxTrackStartupRetries = 50;
+
+static const int kStartSleepTime = 30000;
+static const int kStopSleepTime = 30000;
+
+// Maximum number of pending buffers allocated by OutputTrack::write()
+static const uint8_t kMaxOutputTrackBuffers = 5;
+
+
+#define AUDIOFLINGER_SECURITY_ENABLED 1
+
+// ----------------------------------------------------------------------------
+
+static bool recordingAllowed() {
+#ifndef HAVE_ANDROID_OS
+    return true;
+#endif
+#if AUDIOFLINGER_SECURITY_ENABLED
+    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
+    bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
+    if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
+    return ok;
+#else
+    if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO")))
+        LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest");
+    return true;
+#endif
+}
+
+static bool settingsAllowed() {
+#ifndef HAVE_ANDROID_OS
+    return true;
+#endif
+#if AUDIOFLINGER_SECURITY_ENABLED
+    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
+    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
+    if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
+    return ok;
+#else
+    if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")))
+        LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest");
+    return true;
+#endif
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::AudioFlinger()
+    : BnAudioFlinger(),
+        mAudioHardware(0), mA2dpAudioInterface(0),
+        mA2dpEnabled(false), mA2dpEnabledReq(false),
+        mForcedSpeakerCount(0), mForcedRoute(0), mRouteRestoreTime(0), mMusicMuteSaved(false)
+{
+    mHardwareStatus = AUDIO_HW_IDLE;
+    mAudioHardware = AudioHardwareInterface::create();
+    mHardwareStatus = AUDIO_HW_INIT;
+    if (mAudioHardware->initCheck() == NO_ERROR) {
+        // open 16-bit output stream for s/w mixer
+        mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
+        status_t status;
+        AudioStreamOut *hwOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
+        mHardwareStatus = AUDIO_HW_IDLE;
+        if (hwOutput) {
+            mHardwareMixerThread = new MixerThread(this, hwOutput, AudioSystem::AUDIO_OUTPUT_HARDWARE);
+        } else {
+            LOGE("Failed to initialize hardware output stream, status: %d", status);
+        }
+        
+#ifdef WITH_A2DP
+        // Create A2DP interface
+        mA2dpAudioInterface = new A2dpAudioInterface();
+        AudioStreamOut *a2dpOutput = mA2dpAudioInterface->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
+        if (a2dpOutput) {
+            mA2dpMixerThread = new MixerThread(this, a2dpOutput, AudioSystem::AUDIO_OUTPUT_A2DP);
+            if (hwOutput) {  
+                uint32_t frameCount = ((a2dpOutput->bufferSize()/a2dpOutput->frameSize()) * hwOutput->sampleRate()) / a2dpOutput->sampleRate();
+                MixerThread::OutputTrack *a2dpOutTrack = new MixerThread::OutputTrack(mA2dpMixerThread,
+                                                            hwOutput->sampleRate(),
+                                                            AudioSystem::PCM_16_BIT,
+                                                            hwOutput->channelCount(),
+                                                            frameCount);
+                mHardwareMixerThread->setOuputTrack(a2dpOutTrack);                
+            }
+        } else {
+            LOGE("Failed to initialize A2DP output stream, status: %d", status);
+        }
+#endif
+ 
+        // FIXME - this should come from settings
+        setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
+        setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
+        setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL);
+        setMode(AudioSystem::MODE_NORMAL);
+
+        setMasterVolume(1.0f);
+        setMasterMute(false);
+
+        // Start record thread
+        mAudioRecordThread = new AudioRecordThread(mAudioHardware);
+        if (mAudioRecordThread != 0) {
+            mAudioRecordThread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO);            
+        }
+     } else {
+        LOGE("Couldn't even initialize the stubbed audio hardware!");
+    }
+}
+
+AudioFlinger::~AudioFlinger()
+{
+    if (mAudioRecordThread != 0) {
+        mAudioRecordThread->exit();
+        mAudioRecordThread.clear();        
+    }
+    mHardwareMixerThread.clear();
+    delete mAudioHardware;
+    // deleting mA2dpAudioInterface also deletes mA2dpOutput;
+#ifdef WITH_A2DP
+    mA2dpMixerThread.clear();
+    delete mA2dpAudioInterface;
+#endif
+}
+
+
+#ifdef WITH_A2DP
+void AudioFlinger::setA2dpEnabled(bool enable)
+{
+    LOGV_IF(enable, "set output to A2DP\n");
+    LOGV_IF(!enable, "set output to hardware audio\n");
+
+    mA2dpEnabledReq = enable;
+    mA2dpMixerThread->wakeUp();
+}
+#endif // WITH_A2DP
+
+bool AudioFlinger::streamForcedToSpeaker(int streamType)
+{
+    // NOTE that streams listed here must not be routed to A2DP by default:
+    // AudioSystem::routedToA2dpOutput(streamType) == false
+    return (streamType == AudioSystem::RING ||
+            streamType == AudioSystem::ALARM ||
+            streamType == AudioSystem::NOTIFICATION);
+}
+
+status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    result.append("Clients:\n");
+    for (size_t i = 0; i < mClients.size(); ++i) {
+        wp<Client> wClient = mClients.valueAt(i);
+        if (wClient != 0) {
+            sp<Client> client = wClient.promote();
+            if (client != 0) {
+                snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
+                result.append(buffer);
+            }
+        }
+    }
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+
+status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "Hardware status: %d\n", mHardwareStatus);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    snprintf(buffer, SIZE, "Permission Denial: "
+            "can't dump AudioFlinger from pid=%d, uid=%d\n",
+            IPCThreadState::self()->getCallingPid(),
+            IPCThreadState::self()->getCallingUid());
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
+{
+    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+        dumpPermissionDenial(fd, args);
+    } else {
+        AutoMutex lock(&mLock);
+
+        dumpClients(fd, args);
+        dumpInternals(fd, args);
+        mHardwareMixerThread->dump(fd, args);
+#ifdef WITH_A2DP
+        mA2dpMixerThread->dump(fd, args);
+#endif
+
+        // dump record client
+        if (mAudioRecordThread != 0) mAudioRecordThread->dump(fd, args);
+
+        if (mAudioHardware) {
+            mAudioHardware->dumpState(fd, args);
+        }
+    }
+    return NO_ERROR;
+}
+
+// IAudioFlinger interface
+
+
+sp<IAudioTrack> AudioFlinger::createTrack(
+        pid_t pid,
+        int streamType,
+        uint32_t sampleRate,
+        int format,
+        int channelCount,
+        int frameCount,
+        uint32_t flags,
+        const sp<IMemory>& sharedBuffer,
+        status_t *status)
+{
+    sp<MixerThread::Track> track;
+    sp<TrackHandle> trackHandle;
+    sp<Client> client;
+    wp<Client> wclient;
+    status_t lStatus;
+
+    if (streamType >= AudioSystem::NUM_STREAM_TYPES) {
+        LOGE("invalid stream type");
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    {
+        Mutex::Autolock _l(mLock);
+
+        wclient = mClients.valueFor(pid);
+
+        if (wclient != NULL) {
+            client = wclient.promote();
+        } else {
+            client = new Client(this, pid);
+            mClients.add(pid, client);
+        }
+#ifdef WITH_A2DP
+        if (isA2dpEnabled() && AudioSystem::routedToA2dpOutput(streamType)) {
+            track = mA2dpMixerThread->createTrack(client, streamType, sampleRate, format,
+                    channelCount, frameCount, sharedBuffer, &lStatus);            
+        } else 
+#endif
+        {
+            track = mHardwareMixerThread->createTrack(client, streamType, sampleRate, format,
+                    channelCount, frameCount, sharedBuffer, &lStatus);            
+        }
+        if (track != NULL) {
+            trackHandle = new TrackHandle(track);
+            lStatus = NO_ERROR;            
+        }
+    }
+
+Exit:
+    if(status) {
+        *status = lStatus;
+    }
+    return trackHandle;
+}
+
+uint32_t AudioFlinger::sampleRate(int output) const
+{
+#ifdef WITH_A2DP
+     if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
+         return mA2dpMixerThread->sampleRate();
+     }
+#endif
+     return mHardwareMixerThread->sampleRate();
+}
+
+int AudioFlinger::channelCount(int output) const
+{
+#ifdef WITH_A2DP
+     if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
+         return mA2dpMixerThread->channelCount();
+     }
+#endif
+     return mHardwareMixerThread->channelCount();
+}
+
+int AudioFlinger::format(int output) const
+{
+#ifdef WITH_A2DP
+     if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
+         return mA2dpMixerThread->format();
+     }
+#endif
+     return mHardwareMixerThread->format();
+}
+
+size_t AudioFlinger::frameCount(int output) const
+{
+#ifdef WITH_A2DP
+     if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
+         return mA2dpMixerThread->frameCount();
+     }
+#endif
+     return mHardwareMixerThread->frameCount();
+}
+
+uint32_t AudioFlinger::latency(int output) const
+{
+#ifdef WITH_A2DP
+     if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
+         return mA2dpMixerThread->latency();
+     }
+#endif
+     return mHardwareMixerThread->latency();
+}
+
+status_t AudioFlinger::setMasterVolume(float value)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    // when hw supports master volume, don't scale in sw mixer
+    AutoMutex lock(mHardwareLock);
+    mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+    if (mAudioHardware->setMasterVolume(value) == NO_ERROR) {
+        value = 1.0f;
+    }
+    mHardwareStatus = AUDIO_HW_IDLE;
+    mHardwareMixerThread->setMasterVolume(value);
+#ifdef WITH_A2DP
+    mA2dpMixerThread->setMasterVolume(value);
+#endif
+    
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
+{
+    status_t err = NO_ERROR;
+
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) {
+        LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask);
+        return BAD_VALUE;
+    }
+
+#ifdef WITH_A2DP
+    LOGD("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(), IPCThreadState::self()->getCallingPid());
+    if (mode == AudioSystem::MODE_NORMAL && 
+            (mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) {
+        AutoMutex lock(&mLock);
+
+        bool enableA2dp = false;
+        if (routes & AudioSystem::ROUTE_BLUETOOTH_A2DP) {
+            enableA2dp = true;
+        }
+        setA2dpEnabled(enableA2dp);
+        LOGV("setOutput done\n");
+    }
+#endif
+
+    // do nothing if only A2DP routing is affected
+    mask &= ~AudioSystem::ROUTE_BLUETOOTH_A2DP;
+    if (mask) {
+        AutoMutex lock(mHardwareLock);
+        mHardwareStatus = AUDIO_HW_GET_ROUTING;
+        uint32_t r;
+        err = mAudioHardware->getRouting(mode, &r);
+        if (err == NO_ERROR) {
+            r = (r & ~mask) | (routes & mask);
+            if (mode == AudioSystem::MODE_NORMAL || 
+                (mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
+                mSavedRoute = r;
+                r |= mForcedRoute;
+                LOGV("setRouting mSavedRoute %08x mForcedRoute %08x\n", mSavedRoute, mForcedRoute);
+            }
+            mHardwareStatus = AUDIO_HW_SET_ROUTING;
+            err = mAudioHardware->setRouting(mode, r);
+        }
+        mHardwareStatus = AUDIO_HW_IDLE;
+    }
+    return err;
+}
+
+uint32_t AudioFlinger::getRouting(int mode) const
+{
+    uint32_t routes = 0;
+    if ((mode >= AudioSystem::MODE_CURRENT) && (mode < AudioSystem::NUM_MODES)) {
+        if (mode == AudioSystem::MODE_NORMAL || 
+            (mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
+            routes = mSavedRoute;                
+        } else {
+            mHardwareStatus = AUDIO_HW_GET_ROUTING;
+            mAudioHardware->getRouting(mode, &routes);
+            mHardwareStatus = AUDIO_HW_IDLE;
+        }
+    } else {
+        LOGW("Illegal value: getRouting(%d)", mode);
+    }
+    return routes;
+}
+
+status_t AudioFlinger::setMode(int mode)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
+        LOGW("Illegal value: setMode(%d)", mode);
+        return BAD_VALUE;
+    }
+
+    AutoMutex lock(mHardwareLock);
+    mHardwareStatus = AUDIO_HW_SET_MODE;
+    status_t ret = mAudioHardware->setMode(mode);
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return ret;
+}
+
+int AudioFlinger::getMode() const
+{
+    int mode = AudioSystem::MODE_INVALID;
+    mHardwareStatus = AUDIO_HW_SET_MODE;
+    mAudioHardware->getMode(&mode);
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return mode;
+}
+
+status_t AudioFlinger::setMicMute(bool state)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    AutoMutex lock(mHardwareLock);
+    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
+    status_t ret = mAudioHardware->setMicMute(state);
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return ret;
+}
+
+bool AudioFlinger::getMicMute() const
+{
+    bool state = AudioSystem::MODE_INVALID;
+    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
+    mAudioHardware->getMicMute(&state);
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return state;
+}
+
+status_t AudioFlinger::setMasterMute(bool muted)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    mHardwareMixerThread->setMasterMute(muted);
+#ifdef WITH_A2DP
+    mA2dpMixerThread->setMasterMute(muted);
+#endif
+    return NO_ERROR;
+}
+
+float AudioFlinger::masterVolume() const
+{
+    return mHardwareMixerThread->masterVolume();
+}
+
+bool AudioFlinger::masterMute() const
+{
+    return mHardwareMixerThread->masterMute();
+}
+
+status_t AudioFlinger::setStreamVolume(int stream, float value)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+        return BAD_VALUE;
+    }
+
+    mHardwareMixerThread->setStreamVolume(stream, value);
+#ifdef WITH_A2DP
+    mA2dpMixerThread->setStreamVolume(stream, value);
+#endif
+
+    status_t ret = NO_ERROR;
+    if (stream == AudioSystem::VOICE_CALL ||
+        stream == AudioSystem::BLUETOOTH_SCO) {
+        
+        if (stream == AudioSystem::VOICE_CALL) {
+            value = (float)AudioSystem::logToLinear(value)/100.0f;
+        } else { // (type == AudioSystem::BLUETOOTH_SCO)
+            value = 1.0f;
+        }
+
+        AutoMutex lock(mHardwareLock);
+        mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
+        ret = mAudioHardware->setVoiceVolume(value);
+        mHardwareStatus = AUDIO_HW_IDLE;
+    }
+
+    return ret;
+}
+
+status_t AudioFlinger::setStreamMute(int stream, bool muted)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+        return BAD_VALUE;
+    }
+    
+#ifdef WITH_A2DP
+    mA2dpMixerThread->setStreamMute(stream, muted);
+#endif
+    if (stream == AudioSystem::MUSIC) 
+    {
+        AutoMutex lock(&mHardwareLock);
+        if (mForcedRoute != 0)
+            mMusicMuteSaved = muted;
+        else
+            mHardwareMixerThread->setStreamMute(stream, muted);
+    } else {
+        mHardwareMixerThread->setStreamMute(stream, muted);
+    }
+
+    
+
+    return NO_ERROR;
+}
+
+float AudioFlinger::streamVolume(int stream) const
+{
+    if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+        return 0.0f;
+    }
+    return mHardwareMixerThread->streamVolume(stream);
+}
+
+bool AudioFlinger::streamMute(int stream) const
+{
+    if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+        return true;
+    }
+    
+    if (stream == AudioSystem::MUSIC && mForcedRoute != 0) 
+    {
+        return mMusicMuteSaved;
+    }
+    return mHardwareMixerThread->streamMute(stream);
+}
+
+bool AudioFlinger::isMusicActive() const
+{
+ #ifdef WITH_A2DP
+     if (isA2dpEnabled()) {
+         return mA2dpMixerThread->isMusicActive();
+     }
+ #endif
+    return mHardwareMixerThread->isMusicActive();
+}
+
+status_t AudioFlinger::setParameter(const char* key, const char* value)
+{
+    status_t result, result2;
+    AutoMutex lock(mHardwareLock);
+    mHardwareStatus = AUDIO_SET_PARAMETER;
+    
+    LOGV("setParameter() key %s, value %s, tid %d, calling tid %d", key, value, gettid(), IPCThreadState::self()->getCallingPid());
+    result = mAudioHardware->setParameter(key, value);
+    if (mA2dpAudioInterface) {
+        result2 = mA2dpAudioInterface->setParameter(key, value);
+        if (result2)
+            result = result2;
+    }
+    mHardwareStatus = AUDIO_HW_IDLE;
+    return result;
+}
+
+size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+{
+    return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
+}
+
+void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
+{
+    
+    LOGV("registerClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid());
+    Mutex::Autolock _l(mLock);
+
+    sp<IBinder> binder = client->asBinder();
+    if (mNotificationClients.indexOf(binder) < 0) {
+        LOGV("Adding notification client %p", binder.get());
+        binder->linkToDeath(this);
+        mNotificationClients.add(binder);
+        client->a2dpEnabledChanged(isA2dpEnabled());
+    }
+}
+
+void AudioFlinger::binderDied(const wp<IBinder>& who) {
+    
+    LOGV("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid());
+    Mutex::Autolock _l(mLock);
+
+    IBinder *binder = who.unsafe_get();
+
+    if (binder != NULL) {
+        int index = mNotificationClients.indexOf(binder);
+        if (index >= 0) {
+            LOGV("Removing notification client %p", binder);
+            mNotificationClients.removeAt(index);
+        }
+    }
+}
+
+void AudioFlinger::handleOutputSwitch()
+{
+    if (mA2dpEnabled != mA2dpEnabledReq)
+    {
+        Mutex::Autolock _l(mLock);
+
+        if (mA2dpEnabled != mA2dpEnabledReq)
+        {
+            mA2dpEnabled = mA2dpEnabledReq;
+            SortedVector < sp<MixerThread::Track> > tracks;
+            SortedVector < wp<MixerThread::Track> > activeTracks;
+            
+            // We hold mA2dpMixerThread mLock already 
+            Mutex::Autolock _l(mHardwareMixerThread->mLock);
+            
+            // Transfer tracks playing on MUSIC stream from one mixer to the other
+            if (mA2dpEnabled) {
+                mHardwareMixerThread->getTracks(tracks, activeTracks);
+                mA2dpMixerThread->putTracks(tracks, activeTracks);
+            } else {
+                mA2dpMixerThread->getTracks(tracks, activeTracks);
+                mHardwareMixerThread->putTracks(tracks, activeTracks);
+            }            
+            
+            // Notify AudioSystem of the A2DP activation/deactivation
+            size_t size = mNotificationClients.size();
+            for (size_t i = 0; i < size; i++) {
+                sp<IBinder> binder = mNotificationClients.itemAt(i).promote();
+                if (binder != NULL) {
+                    LOGV("Notifying output change to client %p", binder.get());
+                    sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder);
+                    client->a2dpEnabledChanged(mA2dpEnabled);
+                }
+            }
+
+            mHardwareMixerThread->wakeUp();
+        }
+    }
+}
+
+void AudioFlinger::removeClient(pid_t pid)
+{
+    LOGV("removeClient() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
+    Mutex::Autolock _l(mLock);
+    mClients.removeItem(pid);
+}
+
+void AudioFlinger::wakeUp()
+{
+    mHardwareMixerThread->wakeUp();
+#ifdef WITH_A2DP
+    mA2dpMixerThread->wakeUp();
+#endif // WITH_A2DP
+}
+
+bool AudioFlinger::isA2dpEnabled() const
+{
+    return mA2dpEnabledReq;
+}
+
+void AudioFlinger::handleForcedSpeakerRoute(int command)
+{
+    switch(command) {
+    case ACTIVE_TRACK_ADDED:
+        {
+            AutoMutex lock(mHardwareLock);
+            if (mForcedSpeakerCount++ == 0) {
+                mRouteRestoreTime = 0;
+                mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC);
+                if (mForcedRoute == 0 && !(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
+                    LOGV("Route forced to Speaker ON %08x", mSavedRoute | AudioSystem::ROUTE_SPEAKER);
+                    mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, true);
+                    mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+                    mAudioHardware->setMasterVolume(0);
+                    usleep(mHardwareMixerThread->latency()*1000);
+                    mHardwareStatus = AUDIO_HW_SET_ROUTING;
+                    mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute | AudioSystem::ROUTE_SPEAKER);
+                    mHardwareStatus = AUDIO_HW_IDLE;
+                    // delay track start so that audio hardware has time to siwtch routes
+                    usleep(kStartSleepTime);
+                    mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+                    mAudioHardware->setMasterVolume(mHardwareMixerThread->masterVolume());
+                    mHardwareStatus = AUDIO_HW_IDLE;
+                }
+                mForcedRoute = AudioSystem::ROUTE_SPEAKER;
+            }
+            LOGV("mForcedSpeakerCount incremented to %d", mForcedSpeakerCount);
+        }
+        break;
+    case ACTIVE_TRACK_REMOVED:
+        {
+            AutoMutex lock(mHardwareLock);
+            if (mForcedSpeakerCount > 0){
+                if (--mForcedSpeakerCount == 0) {
+                    mRouteRestoreTime = systemTime() + milliseconds(kStopSleepTime/1000);
+                }
+                LOGV("mForcedSpeakerCount decremented to %d", mForcedSpeakerCount);            
+            } else {
+                LOGE("mForcedSpeakerCount is already zero");            
+            }            
+        }
+        break;
+    case CHECK_ROUTE_RESTORE_TIME:
+    case FORCE_ROUTE_RESTORE:
+        if (mRouteRestoreTime) {
+            AutoMutex lock(mHardwareLock);
+            if (mRouteRestoreTime && 
+               (systemTime() > mRouteRestoreTime || command == FORCE_ROUTE_RESTORE)) {
+                mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, mMusicMuteSaved);
+                mForcedRoute = 0;
+                if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
+                    mHardwareStatus = AUDIO_HW_SET_ROUTING;
+                    mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute);
+                    mHardwareStatus = AUDIO_HW_IDLE;
+                    LOGV("Route forced to Speaker OFF %08x", mSavedRoute);
+                }
+                mRouteRestoreTime = 0;
+            }
+        }
+        break;
+    }
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int outputType)
+    :   Thread(false),
+        mAudioFlinger(audioFlinger), mAudioMixer(0), mOutput(output), mOutputType(outputType), 
+        mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0), mMixBuffer(0),
+        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mStandby(false),
+        mInWrite(false)
+{
+    mSampleRate = output->sampleRate();
+    mChannelCount = output->channelCount();
+
+    // FIXME - Current mixer implementation only supports stereo output
+    if (mChannelCount == 1) {
+        LOGE("Invalid audio hardware channel count");
+    }
+
+    mFormat = output->format();
+    mFrameCount = output->bufferSize() / output->channelCount() / sizeof(int16_t);
+    mAudioMixer = new AudioMixer(mFrameCount, output->sampleRate());
+
+    // FIXME - Current mixer implementation only supports stereo output: Always
+    // Allocate a stereo buffer even if HW output is mono.
+    mMixBuffer = new int16_t[mFrameCount * 2];
+    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
+}
+
+AudioFlinger::MixerThread::~MixerThread()
+{
+    delete [] mMixBuffer;
+    delete mAudioMixer;
+}
+
+status_t AudioFlinger::MixerThread::dump(int fd, const Vector<String16>& args)
+{
+    dumpInternals(fd, args);
+    dumpTracks(fd, args);
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::MixerThread::dumpTracks(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "Output %d mixer thread tracks\n", mOutputType);
+    result.append(buffer);
+    result.append("   Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        wp<Track> wTrack = mTracks[i];
+        if (wTrack != 0) {
+            sp<Track> track = wTrack.promote();
+            if (track != 0) {
+                track->dump(buffer, SIZE);
+                result.append(buffer);
+            }
+        }
+    }
+
+    snprintf(buffer, SIZE, "Output %d mixer thread active tracks\n", mOutputType);
+    result.append(buffer);
+    result.append("   Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
+    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
+        wp<Track> wTrack = mTracks[i];
+        if (wTrack != 0) {
+            sp<Track> track = wTrack.promote();
+            if (track != 0) {
+                track->dump(buffer, SIZE);
+                result.append(buffer);
+            }
+        }
+    }
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "Output %d mixer thread internals\n", mOutputType);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
+    result.append(buffer);
+    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+// Thread virtuals
+bool AudioFlinger::MixerThread::threadLoop()
+{
+    unsigned long sleepTime = kBufferRecoveryInUsecs;
+    int16_t* curBuf = mMixBuffer;
+    Vector< sp<Track> > tracksToRemove;
+    size_t enabledTracks = 0;
+    nsecs_t standbyTime = systemTime();   
+    size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t);
+    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
+
+#ifdef WITH_A2DP
+    bool outputTrackActive = false;
+#endif
+
+    do {
+        enabledTracks = 0;
+        { // scope for the mLock
+        
+            Mutex::Autolock _l(mLock);
+
+#ifdef WITH_A2DP
+            if (mOutputType == AudioSystem::AUDIO_OUTPUT_A2DP) {
+                mAudioFlinger->handleOutputSwitch();
+            }
+            if (mOutputTrack != NULL && !mAudioFlinger->isA2dpEnabled()) {
+                if (outputTrackActive) {
+                    mOutputTrack->stop();
+                    outputTrackActive = false;
+                }
+            }
+#endif
+
+            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+
+            // put audio hardware into standby after short delay
+            if UNLIKELY(!activeTracks.size() && systemTime() > standbyTime) {
+                // wait until we have something to do...
+                LOGV("Audio hardware entering standby, output %d\n", mOutputType);
+//                mAudioFlinger->mHardwareStatus = AUDIO_HW_STANDBY;
+                if (!mStandby) {
+                    mOutput->standby();
+                    mStandby = true;
+                }
+                
+#ifdef WITH_A2DP
+                if (outputTrackActive) {
+                    mOutputTrack->stop();
+                    outputTrackActive = false;
+                }
+#endif
+                if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
+                    mAudioFlinger->handleForcedSpeakerRoute(FORCE_ROUTE_RESTORE);
+                }                
+//                mHardwareStatus = AUDIO_HW_IDLE;
+                // we're about to wait, flush the binder command buffer
+                IPCThreadState::self()->flushCommands();
+                mWaitWorkCV.wait(mLock);
+                LOGV("Audio hardware exiting standby, output %d\n", mOutputType);
+                
+                if (mMasterMute == false) {
+                    char value[PROPERTY_VALUE_MAX];
+                    property_get("ro.audio.silent", value, "0");
+                    if (atoi(value)) {
+                        LOGD("Silence is golden");
+                        setMasterMute(true);
+                    }                    
+                }
+                
+                standbyTime = systemTime() + kStandbyTimeInNsecs;
+                continue;
+            }
+
+            // Forced route to speaker is handled by hardware mixer thread
+            if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
+                mAudioFlinger->handleForcedSpeakerRoute(CHECK_ROUTE_RESTORE_TIME);
+            }
+
+            // find out which tracks need to be processed
+            size_t count = activeTracks.size();
+            for (size_t i=0 ; i<count ; i++) {
+                sp<Track> t = activeTracks[i].promote();
+                if (t == 0) continue;
+
+                Track* const track = t.get();
+                audio_track_cblk_t* cblk = track->cblk();
+
+                // The first time a track is added we wait
+                // for all its buffers to be filled before processing it
+                mAudioMixer->setActiveTrack(track->name());
+                if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
+                        !track->isPaused())
+                {
+                    //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+                    // compute volume for this track
+                    int16_t left, right;
+                    if (track->isMuted() || mMasterMute || track->isPausing()) {
+                        left = right = 0;
+                        if (track->isPausing()) {
+                            LOGV("paused(%d)", track->name());
+                            track->setPaused();
+                        }
+                    } else {
+                        float typeVolume = mStreamTypes[track->type()].volume;
+                        float v = mMasterVolume * typeVolume;
+                        float v_clamped = v * cblk->volume[0];
+                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+                        left = int16_t(v_clamped);
+                        v_clamped = v * cblk->volume[1];
+                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+                        right = int16_t(v_clamped);
+                    }
+
+                    // XXX: these things DON'T need to be done each time
+                    mAudioMixer->setBufferProvider(track);
+                    mAudioMixer->enable(AudioMixer::MIXING);
+
+                    int param;
+                    if ( track->mFillingUpStatus == Track::FS_FILLED) {
+                        // no ramp for the first volume setting
+                        track->mFillingUpStatus = Track::FS_ACTIVE;
+                        if (track->mState == TrackBase::RESUMING) {
+                            track->mState = TrackBase::ACTIVE;
+                            param = AudioMixer::RAMP_VOLUME;
+                        } else {
+                            param = AudioMixer::VOLUME;
+                        }
+                    } else {
+                        param = AudioMixer::RAMP_VOLUME;
+                    }
+                    mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
+                    mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
+                    mAudioMixer->setParameter(
+                        AudioMixer::TRACK,
+                        AudioMixer::FORMAT, track->format());
+                    mAudioMixer->setParameter(
+                        AudioMixer::TRACK,
+                        AudioMixer::CHANNEL_COUNT, track->channelCount());
+                    mAudioMixer->setParameter(
+                        AudioMixer::RESAMPLE,
+                        AudioMixer::SAMPLE_RATE,
+                        int(cblk->sampleRate));
+
+                    // reset retry count
+                    track->mRetryCount = kMaxTrackRetries;
+                    enabledTracks++;
+                } else {
+                    //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+                    if (track->isStopped()) {
+                        track->reset();
+                    }
+                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
+                        // We have consumed all the buffers of this track.
+                        // Remove it from the list of active tracks.
+                        LOGV("remove(%d) from active list", track->name());
+                        tracksToRemove.add(track);
+                    } else {
+                        // No buffers for this track. Give it a few chances to
+                        // fill a buffer, then remove it from active list.
+                        if (--(track->mRetryCount) <= 0) {
+                            LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+                            tracksToRemove.add(track);
+                        }
+                    }
+                    // LOGV("disable(%d)", track->name());
+                    mAudioMixer->disable(AudioMixer::MIXING);
+                }
+            }
+
+            // remove all the tracks that need to be...
+            count = tracksToRemove.size();
+            if (UNLIKELY(count)) {
+                for (size_t i=0 ; i<count ; i++) {
+                    const sp<Track>& track = tracksToRemove[i];
+                    removeActiveTrack(track);
+                    if (track->isTerminated()) {
+                        mTracks.remove(track);
+                        deleteTrackName(track->mName);
+                    }
+                }
+            }  
+       }
+        
+        if (LIKELY(enabledTracks)) {
+            // mix buffers...
+            mAudioMixer->process(curBuf);
+
+#ifdef WITH_A2DP
+            if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
+                if (!outputTrackActive) {
+                    LOGV("starting output track in mixer for output %d", mOutputType);
+                    mOutputTrack->start();
+                    outputTrackActive = true;
+                }
+                mOutputTrack->write(curBuf, mFrameCount);
+            }
+#endif
+
+            // output audio to hardware
+            mLastWriteTime = systemTime();
+            mInWrite = true;
+            mOutput->write(curBuf, mixBufferSize);
+            mNumWrites++;
+            mInWrite = false;
+            mStandby = false;
+            nsecs_t temp = systemTime();
+            standbyTime = temp + kStandbyTimeInNsecs;
+            nsecs_t delta = temp - mLastWriteTime;
+            if (delta > maxPeriod) {
+                LOGW("write blocked for %llu msecs", ns2ms(delta));
+                mNumDelayedWrites++;
+            }
+            sleepTime = kBufferRecoveryInUsecs;
+        } else {         
+#ifdef WITH_A2DP
+            if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
+                if (outputTrackActive) {
+                    mOutputTrack->write(curBuf, 0);
+                    if (mOutputTrack->bufferQueueEmpty()) {
+                        mOutputTrack->stop();
+                        outputTrackActive = false;
+                    } else {
+                        standbyTime = systemTime() + kStandbyTimeInNsecs;
+                    }
+                }
+            }
+#endif
+            // There was nothing to mix this round, which means all
+            // active tracks were late. Sleep a little bit to give
+            // them another chance. If we're too late, the audio
+            // hardware will zero-fill for us.
+            //LOGV("no buffers - usleep(%lu)", sleepTime);
+            usleep(sleepTime);
+            if (sleepTime < kMaxBufferRecoveryInUsecs) {
+                sleepTime += kBufferRecoveryInUsecs;
+            }
+        }
+
+        // finally let go of all our tracks, without the lock held
+        // since we can't guarantee the destructors won't acquire that
+        // same lock.
+        tracksToRemove.clear();
+    } while (true);
+
+    return false;
+}
+
+status_t AudioFlinger::MixerThread::readyToRun()
+{
+    if (mSampleRate == 0) {
+        LOGE("No working audio driver found.");
+        return NO_INIT;
+    }
+    LOGI("AudioFlinger's thread ready to run for output %d", mOutputType);
+    return NO_ERROR;
+}
+
+void AudioFlinger::MixerThread::onFirstRef()
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+
+    snprintf(buffer, SIZE, "Mixer Thread for output %d", mOutputType);
+
+    run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
+}
+
+
+sp<AudioFlinger::MixerThread::Track>  AudioFlinger::MixerThread::createTrack(
+        const sp<AudioFlinger::Client>& client,
+        int streamType,
+        uint32_t sampleRate,
+        int format,
+        int channelCount,
+        int frameCount,
+        const sp<IMemory>& sharedBuffer,
+        status_t *status)
+{
+    sp<Track> track;
+    status_t lStatus;
+    
+    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
+    if (sampleRate > MAX_SAMPLE_RATE || sampleRate > mSampleRate*2) {
+        LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    {
+        Mutex::Autolock _l(mLock);
+
+        if (mSampleRate == 0) {
+            LOGE("Audio driver not initialized.");
+            lStatus = NO_INIT;
+            goto Exit;
+        }
+
+        track = new Track(this, client, streamType, sampleRate, format,
+                channelCount, frameCount, sharedBuffer);
+        if (track->getCblk() == NULL) {
+            track.clear();
+            lStatus = NO_MEMORY;
+            goto Exit;
+        }
+        mTracks.add(track);
+        lStatus = NO_ERROR;
+    }
+
+Exit:
+    if(status) {
+        *status = lStatus;
+    }
+    return track;
+}
+
+void AudioFlinger::MixerThread::getTracks(
+        SortedVector < sp<Track> >& tracks,
+        SortedVector < wp<Track> >& activeTracks)
+{
+    size_t size = mTracks.size();
+    LOGV ("MixerThread::getTracks() for output %d, mTracks.size %d, mActiveTracks.size %d", mOutputType,  mTracks.size(), mActiveTracks.size());
+    for (size_t i = 0; i < size; i++) {
+        sp<Track> t = mTracks[i];
+        if (AudioSystem::routedToA2dpOutput(t->mStreamType)) {
+            tracks.add(t);
+            int j = mActiveTracks.indexOf(t);
+            if (j >= 0) {
+                t = mActiveTracks[j].promote();
+                if (t != NULL) {
+                    activeTracks.add(t);                                    
+                }                            
+            }
+        }
+    }
+
+    size = activeTracks.size();
+    for (size_t i = 0; i < size; i++) {
+        removeActiveTrack(activeTracks[i]);
+    }
+    
+    size = tracks.size();
+    for (size_t i = 0; i < size; i++) {
+        sp<Track> t = tracks[i];
+        mTracks.remove(t);
+        deleteTrackName(t->name());
+    }
+}
+
+void AudioFlinger::MixerThread::putTracks(
+        SortedVector < sp<Track> >& tracks,
+        SortedVector < wp<Track> >& activeTracks)
+{
+
+    LOGV ("MixerThread::putTracks() for output %d, tracks.size %d, activeTracks.size %d", mOutputType,  tracks.size(), activeTracks.size());
+
+    size_t size = tracks.size();
+    for (size_t i = 0; i < size ; i++) {
+        sp<Track> t = tracks[i];
+        int name = getTrackName();
+
+        if (name < 0) return;
+        
+        t->mName = name;
+        t->mMixerThread = this;
+        mTracks.add(t);
+
+        int j = activeTracks.indexOf(t);
+        if (j >= 0) {
+            addActiveTrack(t);
+        }            
+    }
+}
+
+uint32_t AudioFlinger::MixerThread::sampleRate() const
+{
+    return mSampleRate;
+}
+
+int AudioFlinger::MixerThread::channelCount() const
+{
+    return mChannelCount;
+}
+
+int AudioFlinger::MixerThread::format() const
+{
+    return mFormat;
+}
+
+size_t AudioFlinger::MixerThread::frameCount() const
+{
+    return mFrameCount;
+}
+
+uint32_t AudioFlinger::MixerThread::latency() const
+{
+    if (mOutput) {
+        return mOutput->latency();
+    }
+    else {
+        return 0;
+    }
+}
+
+status_t AudioFlinger::MixerThread::setMasterVolume(float value)
+{
+    mMasterVolume = value;
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::MixerThread::setMasterMute(bool muted)
+{
+    mMasterMute = muted;
+    return NO_ERROR;
+}
+
+float AudioFlinger::MixerThread::masterVolume() const
+{
+    return mMasterVolume;
+}
+
+bool AudioFlinger::MixerThread::masterMute() const
+{
+    return mMasterMute;
+}
+
+status_t AudioFlinger::MixerThread::setStreamVolume(int stream, float value)
+{
+    mStreamTypes[stream].volume = value;
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::MixerThread::setStreamMute(int stream, bool muted)
+{
+    mStreamTypes[stream].mute = muted;
+    return NO_ERROR;
+}
+
+float AudioFlinger::MixerThread::streamVolume(int stream) const
+{
+    return mStreamTypes[stream].volume;
+}
+
+bool AudioFlinger::MixerThread::streamMute(int stream) const
+{
+    return mStreamTypes[stream].mute;
+}
+
+bool AudioFlinger::MixerThread::isMusicActive() const
+{
+    size_t count = mActiveTracks.size();
+    for (size_t i = 0 ; i < count ; ++i) {
+        sp<Track> t = mActiveTracks[i].promote();
+        if (t == 0) continue;
+        Track* const track = t.get();
+        if (t->mStreamType == AudioSystem::MUSIC)
+            return true;
+    }
+    return false;
+}
+
+status_t AudioFlinger::MixerThread::addTrack(const sp<Track>& track)
+{
+    status_t status = ALREADY_EXISTS;
+    Mutex::Autolock _l(mLock);
+
+    // here the track could be either new, or restarted
+    // in both cases "unstop" the track
+    if (track->isPaused()) {
+        track->mState = TrackBase::RESUMING;
+        LOGV("PAUSED => RESUMING (%d)", track->name());
+    } else {
+        track->mState = TrackBase::ACTIVE;
+        LOGV("? => ACTIVE (%d)", track->name());
+    }
+    // set retry count for buffer fill
+    track->mRetryCount = kMaxTrackStartupRetries;
+    if (mActiveTracks.indexOf(track) < 0) {
+        // the track is newly added, make sure it fills up all its
+        // buffers before playing. This is to ensure the client will
+        // effectively get the latency it requested.
+        track->mFillingUpStatus = Track::FS_FILLING;
+        track->mResetDone = false;
+        addActiveTrack(track);
+        status = NO_ERROR;
+    }
+    
+    LOGV("mWaitWorkCV.broadcast");
+    mWaitWorkCV.broadcast();
+
+    return status;
+}
+
+void AudioFlinger::MixerThread::removeTrack(wp<Track> track, int name)
+{
+    Mutex::Autolock _l(mLock);
+    sp<Track> t = track.promote();
+    if (t!=NULL && (t->mState <= TrackBase::STOPPED)) {
+        remove_track_l(track, name);
+    }
+}
+
+void AudioFlinger::MixerThread::remove_track_l(wp<Track> track, int name)
+{
+    sp<Track> t = track.promote();
+    if (t!=NULL) {
+        t->reset();
+    }
+    deleteTrackName(name);
+    removeActiveTrack(track);
+    mWaitWorkCV.broadcast();
+}
+
+void AudioFlinger::MixerThread::destroyTrack(const sp<Track>& track)
+{
+    // NOTE: We're acquiring a strong reference on the track before
+    // acquiring the lock, this is to make sure removing it from
+    // mTracks won't cause the destructor to be called while the lock is
+    // held (note that technically, 'track' could be a reference to an item
+    // in mTracks, which is why we need to do this).
+    sp<Track> keep(track);
+    Mutex::Autolock _l(mLock);
+    track->mState = TrackBase::TERMINATED;
+    if (mActiveTracks.indexOf(track) < 0) {
+        LOGV("remove track (%d) and delete from mixer", track->name());
+        mTracks.remove(track);
+        deleteTrackName(keep->name());
+    }
+}
+
+
+void AudioFlinger::MixerThread::addActiveTrack(const wp<Track>& t)
+{
+    mActiveTracks.add(t);
+
+    // Force routing to speaker for certain stream types
+    // The forced routing to speaker is managed by hardware mixer
+    if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
+        sp<Track> track = t.promote();
+        if (track == NULL) return;
+   
+        if (streamForcedToSpeaker(track->type())) {
+            mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_ADDED);
+        }        
+    }
+}
+
+void AudioFlinger::MixerThread::removeActiveTrack(const wp<Track>& t)
+{
+    mActiveTracks.remove(t);
+
+    // Force routing to speaker for certain stream types
+    // The forced routing to speaker is managed by hardware mixer
+    if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
+        sp<Track> track = t.promote();
+        if (track == NULL) return;
+
+        if (streamForcedToSpeaker(track->type())) {
+            mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_REMOVED);
+        }
+    }
+}
+
+int AudioFlinger::MixerThread::getTrackName()
+{
+    return mAudioMixer->getTrackName();
+}
+
+void AudioFlinger::MixerThread::deleteTrackName(int name)
+{
+    mAudioMixer->deleteTrackName(name);
+}
+
+size_t AudioFlinger::MixerThread::getOutputFrameCount() 
+{
+    return mOutput->bufferSize() / mOutput->channelCount() / sizeof(int16_t);
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::TrackBase::TrackBase(
+            const sp<MixerThread>& mixerThread,
+            const sp<Client>& client,
+            int streamType,
+            uint32_t sampleRate,
+            int format,
+            int channelCount,
+            int frameCount,
+            uint32_t flags,
+            const sp<IMemory>& sharedBuffer)
+    :   RefBase(),
+        mMixerThread(mixerThread),
+        mClient(client),
+        mStreamType(streamType),
+        mFrameCount(0),
+        mState(IDLE),
+        mClientTid(-1),
+        mFormat(format),
+        mFlags(flags & ~SYSTEM_FLAGS_MASK)
+{
+    mName = mixerThread->getTrackName();
+    LOGV("TrackBase contructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+    if (mName < 0) {
+        LOGE("no more track names availlable");
+        return;
+    }
+
+    LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
+
+    // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
+   size_t size = sizeof(audio_track_cblk_t);
+   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
+   if (sharedBuffer == 0) {
+       size += bufferSize;
+   }
+
+   if (client != NULL) {
+        mCblkMemory = client->heap()->allocate(size);
+        if (mCblkMemory != 0) {
+            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
+            if (mCblk) { // construct the shared structure in-place.
+                new(mCblk) audio_track_cblk_t();
+                // clear all buffers
+                mCblk->frameCount = frameCount;
+                mCblk->sampleRate = sampleRate;
+                mCblk->channels = channelCount;
+                if (sharedBuffer == 0) {
+                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
+                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
+                    // Force underrun condition to avoid false underrun callback until first data is
+                    // written to buffer
+                    mCblk->flowControlFlag = 1;
+                } else {
+                    mBuffer = sharedBuffer->pointer();
+                }
+                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
+            }
+        } else {
+            LOGE("not enough memory for AudioTrack size=%u", size);
+            client->heap()->dump("AudioTrack");
+            return;
+        }
+   } else {
+       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
+       if (mCblk) { // construct the shared structure in-place.
+           new(mCblk) audio_track_cblk_t();
+           // clear all buffers
+           mCblk->frameCount = frameCount;
+           mCblk->sampleRate = sampleRate;
+           mCblk->channels = channelCount;
+           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
+           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
+           // Force underrun condition to avoid false underrun callback until first data is
+           // written to buffer
+           mCblk->flowControlFlag = 1;
+           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
+       }
+   }
+}
+
+AudioFlinger::MixerThread::TrackBase::~TrackBase()
+{
+    if (mCblk) {
+        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.        
+    }
+    mCblkMemory.clear();            // and free the shared memory
+    mClient.clear();
+}
+
+void AudioFlinger::MixerThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    buffer->raw = 0;
+    mFrameCount = buffer->frameCount;
+    step();
+    buffer->frameCount = 0;
+}
+
+bool AudioFlinger::MixerThread::TrackBase::step() {
+    bool result;
+    audio_track_cblk_t* cblk = this->cblk();
+
+    result = cblk->stepServer(mFrameCount);
+    if (!result) {
+        LOGV("stepServer failed acquiring cblk mutex");
+        mFlags |= STEPSERVER_FAILED;
+    }
+    return result;
+}
+
+void AudioFlinger::MixerThread::TrackBase::reset() {
+    audio_track_cblk_t* cblk = this->cblk();
+
+    cblk->user = 0;
+    cblk->server = 0;
+    cblk->userBase = 0;
+    cblk->serverBase = 0;
+    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
+    LOGV("TrackBase::reset");
+}
+
+sp<IMemory> AudioFlinger::MixerThread::TrackBase::getCblk() const
+{
+    return mCblkMemory;
+}
+
+int AudioFlinger::MixerThread::TrackBase::sampleRate() const {
+    return mCblk->sampleRate;
+}
+
+int AudioFlinger::MixerThread::TrackBase::channelCount() const {
+    return mCblk->channels;
+}
+
+void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
+    audio_track_cblk_t* cblk = this->cblk();
+    int16_t *bufferStart = (int16_t *)mBuffer + (offset-cblk->serverBase)*cblk->channels;
+    int16_t *bufferEnd = bufferStart + frames * cblk->channels;
+
+    // Check validity of returned pointer in case the track control block would have been corrupted.
+    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd) {
+        LOGW("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
+                server %d, serverBase %d, user %d, userBase %d",
+                bufferStart, bufferEnd, mBuffer, mBufferEnd,
+                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
+        return 0;
+    }
+
+    return bufferStart;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::Track::Track(
+            const sp<MixerThread>& mixerThread,
+            const sp<Client>& client,
+            int streamType,
+            uint32_t sampleRate,
+            int format,
+            int channelCount,
+            int frameCount,
+            const sp<IMemory>& sharedBuffer)
+    :   TrackBase(mixerThread, client, streamType, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
+{
+    mVolume[0] = 1.0f;
+    mVolume[1] = 1.0f;
+    mMute = false;
+    mSharedBuffer = sharedBuffer;
+}
+
+AudioFlinger::MixerThread::Track::~Track()
+{
+    wp<Track> weak(this); // never create a strong ref from the dtor
+    mState = TERMINATED;
+    mMixerThread->removeTrack(weak, mName);
+}
+
+void AudioFlinger::MixerThread::Track::destroy()
+{
+    mMixerThread->destroyTrack(this);
+}
+
+void AudioFlinger::MixerThread::Track::dump(char* buffer, size_t size)
+{
+    snprintf(buffer, size, "  %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n",
+            mName - AudioMixer::TRACK0,
+            (mClient == NULL) ? getpid() : mClient->pid(),
+            mStreamType,
+            mFormat,
+            mCblk->channels,
+            mFrameCount,
+            mState,
+            mMute,
+            mFillingUpStatus,
+            mCblk->sampleRate,
+            mCblk->volume[0],
+            mCblk->volume[1],
+            mCblk->server,
+            mCblk->user);
+}
+
+status_t AudioFlinger::MixerThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+     audio_track_cblk_t* cblk = this->cblk();
+     uint32_t framesReady;
+     uint32_t framesReq = buffer->frameCount;
+
+     // Check if last stepServer failed, try to step now
+     if (mFlags & TrackBase::STEPSERVER_FAILED) {
+         if (!step())  goto getNextBuffer_exit;
+         LOGV("stepServer recovered");
+         mFlags &= ~TrackBase::STEPSERVER_FAILED;
+     }
+
+     framesReady = cblk->framesReady();
+
+     if (LIKELY(framesReady)) {
+        uint32_t s = cblk->server;
+        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
+
+        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
+        if (framesReq > framesReady) {
+            framesReq = framesReady;
+        }
+        if (s + framesReq > bufferEnd) {
+            framesReq = bufferEnd - s;
+        }
+
+         buffer->raw = getBuffer(s, framesReq);
+         if (buffer->raw == 0) goto getNextBuffer_exit;
+
+         buffer->frameCount = framesReq;
+        return NO_ERROR;
+     }
+
+getNextBuffer_exit:
+     buffer->raw = 0;
+     buffer->frameCount = 0;
+     return NOT_ENOUGH_DATA;
+}
+
+bool AudioFlinger::MixerThread::Track::isReady() const {
+    if (mFillingUpStatus != FS_FILLING) return true;
+
+    if (mCblk->framesReady() >= mCblk->frameCount ||
+        mCblk->forceReady) {
+        mFillingUpStatus = FS_FILLED;
+        mCblk->forceReady = 0;
+        LOGV("Track::isReady() track %d for output %d", mName, mMixerThread->mOutputType);
+        return true;
+    }
+    return false;
+}
+
+status_t AudioFlinger::MixerThread::Track::start()
+{
+    LOGV("start(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
+    mMixerThread->addTrack(this);
+    return NO_ERROR;
+}
+
+void AudioFlinger::MixerThread::Track::stop()
+{
+    LOGV("stop(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
+    Mutex::Autolock _l(mMixerThread->mLock);
+    if (mState > STOPPED) {
+        mState = STOPPED;
+        // If the track is not active (PAUSED and buffers full), flush buffers
+        if (mMixerThread->mActiveTracks.indexOf(this) < 0) {
+            reset();
+        }
+        LOGV("(> STOPPED) => STOPPED (%d)", mName);
+    }
+}
+
+void AudioFlinger::MixerThread::Track::pause()
+{
+    LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+    Mutex::Autolock _l(mMixerThread->mLock);
+    if (mState == ACTIVE || mState == RESUMING) {
+        mState = PAUSING;
+        LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
+    }
+}
+
+void AudioFlinger::MixerThread::Track::flush()
+{
+    LOGV("flush(%d)", mName);
+    Mutex::Autolock _l(mMixerThread->mLock);
+    if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
+        return;
+    }
+    // No point remaining in PAUSED state after a flush => go to
+    // STOPPED state
+    mState = STOPPED;
+
+    // NOTE: reset() will reset cblk->user and cblk->server with
+    // the risk that at the same time, the AudioMixer is trying to read
+    // data. In this case, getNextBuffer() would return a NULL pointer
+    // as audio buffer => the AudioMixer code MUST always test that pointer
+    // returned by getNextBuffer() is not NULL!
+    reset();
+}
+
+void AudioFlinger::MixerThread::Track::reset()
+{
+    // Do not reset twice to avoid discarding data written just after a flush and before
+    // the audioflinger thread detects the track is stopped.
+    if (!mResetDone) {
+        TrackBase::reset();
+        // Force underrun condition to avoid false underrun callback until first data is
+        // written to buffer
+        mCblk->flowControlFlag = 1;
+        mCblk->forceReady = 0;
+        mFillingUpStatus = FS_FILLING;        
+        mResetDone = true;
+    }
+}
+
+void AudioFlinger::MixerThread::Track::mute(bool muted)
+{
+    mMute = muted;
+}
+
+void AudioFlinger::MixerThread::Track::setVolume(float left, float right)
+{
+    mVolume[0] = left;
+    mVolume[1] = right;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::RecordTrack::RecordTrack(
+            const sp<MixerThread>& mixerThread,
+            const sp<Client>& client,
+            int streamType,
+            uint32_t sampleRate,
+            int format,
+            int channelCount,
+            int frameCount,
+            uint32_t flags)
+    :   TrackBase(mixerThread, client, streamType, sampleRate, format,
+                  channelCount, frameCount, flags, 0),
+        mOverflow(false)
+{
+}
+
+AudioFlinger::MixerThread::RecordTrack::~RecordTrack()
+{
+    mMixerThread->deleteTrackName(mName);
+}
+
+status_t AudioFlinger::MixerThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    audio_track_cblk_t* cblk = this->cblk();
+    uint32_t framesAvail;
+    uint32_t framesReq = buffer->frameCount;
+
+     // Check if last stepServer failed, try to step now
+    if (mFlags & TrackBase::STEPSERVER_FAILED) {
+        if (!step()) goto getNextBuffer_exit;
+        LOGV("stepServer recovered");
+        mFlags &= ~TrackBase::STEPSERVER_FAILED;
+    }
+
+    framesAvail = cblk->framesAvailable_l();
+
+    if (LIKELY(framesAvail)) {
+        uint32_t s = cblk->server;
+        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
+
+        if (framesReq > framesAvail) {
+            framesReq = framesAvail;
+        }
+        if (s + framesReq > bufferEnd) {
+            framesReq = bufferEnd - s;
+        }
+
+        buffer->raw = getBuffer(s, framesReq);
+        if (buffer->raw == 0) goto getNextBuffer_exit;
+
+        buffer->frameCount = framesReq;
+        return NO_ERROR;
+    }
+
+getNextBuffer_exit:
+    buffer->raw = 0;
+    buffer->frameCount = 0;
+    return NOT_ENOUGH_DATA;
+}
+
+status_t AudioFlinger::MixerThread::RecordTrack::start()
+{
+    return mMixerThread->mAudioFlinger->startRecord(this);
+}
+
+void AudioFlinger::MixerThread::RecordTrack::stop()
+{
+    mMixerThread->mAudioFlinger->stopRecord(this);
+    TrackBase::reset();
+    // Force overerrun condition to avoid false overrun callback until first data is
+    // read from buffer
+    mCblk->flowControlFlag = 1;
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::OutputTrack::OutputTrack(
+            const sp<MixerThread>& mixerThread,
+            uint32_t sampleRate,
+            int format,
+            int channelCount,
+            int frameCount)
+    :   Track(mixerThread, NULL, AudioSystem::SYSTEM, sampleRate, format, channelCount, frameCount, NULL),
+    mOutputMixerThread(mixerThread)
+{
+                
+    mCblk->out = 1;
+    mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+    mCblk->volume[0] = mCblk->volume[1] = 0x1000;
+    mOutBuffer.frameCount = 0;
+    mCblk->bufferTimeoutMs = 10;
+    
+    LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p", 
+            mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd);
+    
+}
+
+AudioFlinger::MixerThread::OutputTrack::~OutputTrack()
+{
+    stop();
+}
+
+status_t AudioFlinger::MixerThread::OutputTrack::start()
+{
+    status_t status = Track::start();
+    
+    mRetryCount = 127;
+    return status;
+}
+
+void AudioFlinger::MixerThread::OutputTrack::stop()
+{
+    Track::stop();
+    clearBufferQueue();
+    mOutBuffer.frameCount = 0;
+}
+
+void AudioFlinger::MixerThread::OutputTrack::write(int16_t* data, uint32_t frames)
+{
+    Buffer *pInBuffer;
+    Buffer inBuffer;
+    uint32_t channels = mCblk->channels;
+        
+    inBuffer.frameCount = frames;
+    inBuffer.i16 = data;
+    
+    if (mCblk->user == 0) {
+        if (mOutputMixerThread->isMusicActive()) {
+            mCblk->forceReady = 1;
+            LOGV("OutputTrack::start() force ready");
+        } else if (mCblk->frameCount > frames){
+            if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
+                uint32_t startFrames = (mCblk->frameCount - frames);
+                LOGV("OutputTrack::start() write %d frames", startFrames);
+                pInBuffer = new Buffer;
+                pInBuffer->mBuffer = new int16_t[startFrames * channels];
+                pInBuffer->frameCount = startFrames;
+                pInBuffer->i16 = pInBuffer->mBuffer;
+                memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t));
+                mBufferQueue.add(pInBuffer);                
+            } else {
+                LOGW ("OutputTrack::write() no more buffers");
+            }
+        }        
+    }
+
+    while (1) { 
+        // First write pending buffers, then new data
+        if (mBufferQueue.size()) {
+            pInBuffer = mBufferQueue.itemAt(0);
+        } else {
+            pInBuffer = &inBuffer;
+        }
+ 
+        if (pInBuffer->frameCount == 0) {
+            break;
+        }
+        
+        if (mOutBuffer.frameCount == 0) {
+            mOutBuffer.frameCount = pInBuffer->frameCount;
+            if (obtainBuffer(&mOutBuffer) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
+                break;
+            }
+        }
+            
+        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
+        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channels * sizeof(int16_t));
+        mCblk->stepUser(outFrames);
+        pInBuffer->frameCount -= outFrames;
+        pInBuffer->i16 += outFrames * channels;
+        mOutBuffer.frameCount -= outFrames;
+        mOutBuffer.i16 += outFrames * channels;            
+        
+        if (pInBuffer->frameCount == 0) {
+            if (mBufferQueue.size()) {
+                mBufferQueue.removeAt(0);
+                delete [] pInBuffer->mBuffer;
+                delete pInBuffer;
+            } else {
+                break;
+            }
+        }
+    }
+ 
+    // If we could not write all frames, allocate a buffer and queue it for next time.
+    if (inBuffer.frameCount) {
+        if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
+            pInBuffer = new Buffer;
+            pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channels];
+            pInBuffer->frameCount = inBuffer.frameCount;
+            pInBuffer->i16 = pInBuffer->mBuffer;
+            memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * sizeof(int16_t));
+            mBufferQueue.add(pInBuffer);
+        } else {
+            LOGW("OutputTrack::write() no more buffers");
+        }
+    }
+    
+    // Calling write() with a 0 length buffer, means that no more data will be written:
+    // If no more buffers are pending, fill output track buffer to make sure it is started 
+    // by output mixer.
+    if (frames == 0 && mBufferQueue.size() == 0 && mCblk->user < mCblk->frameCount) {
+        frames = mCblk->frameCount - mCblk->user;
+        pInBuffer = new Buffer;
+        pInBuffer->mBuffer = new int16_t[frames * channels];
+        pInBuffer->frameCount = frames;
+        pInBuffer->i16 = pInBuffer->mBuffer;
+        memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t));
+        mBufferQueue.add(pInBuffer);
+    }
+
+}
+
+status_t AudioFlinger::MixerThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    int active;
+    int timeout = 0;
+    status_t result;
+    audio_track_cblk_t* cblk = mCblk;
+    uint32_t framesReq = buffer->frameCount;
+
+    LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
+    buffer->frameCount  = 0;
+    
+    uint32_t framesAvail = cblk->framesAvailable();
+
+    if (framesAvail == 0) {
+        return AudioTrack::NO_MORE_BUFFERS;
+    }
+
+    if (framesReq > framesAvail) {
+        framesReq = framesAvail;
+    }
+
+    uint32_t u = cblk->user;
+    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
+
+    if (u + framesReq > bufferEnd) {
+        framesReq = bufferEnd - u;
+    }
+
+    buffer->frameCount  = framesReq;
+    buffer->raw         = (void *)cblk->buffer(u);
+    return NO_ERROR;
+}
+
+
+void AudioFlinger::MixerThread::OutputTrack::clearBufferQueue()
+{
+    size_t size = mBufferQueue.size();
+    Buffer *pBuffer;
+    
+    for (size_t i = 0; i < size; i++) {
+        pBuffer = mBufferQueue.itemAt(i);
+        delete [] pBuffer->mBuffer;
+        delete pBuffer;
+    }
+    mBufferQueue.clear();
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
+    :   RefBase(),
+        mAudioFlinger(audioFlinger),
+        mMemoryDealer(new MemoryDealer(1024*1024)),
+        mPid(pid)
+{
+    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
+}
+
+AudioFlinger::Client::~Client()
+{
+    mAudioFlinger->removeClient(mPid);
+}
+
+const sp<MemoryDealer>& AudioFlinger::Client::heap() const
+{
+    return mMemoryDealer;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::MixerThread::Track>& track)
+    : BnAudioTrack(),
+      mTrack(track)
+{
+}
+
+AudioFlinger::TrackHandle::~TrackHandle() {
+    // just stop the track on deletion, associated resources
+    // will be freed from the main thread once all pending buffers have
+    // been played. Unless it's not in the active track list, in which
+    // case we free everything now...
+    mTrack->destroy();
+}
+
+status_t AudioFlinger::TrackHandle::start() {
+    return mTrack->start();
+}
+
+void AudioFlinger::TrackHandle::stop() {
+    mTrack->stop();
+}
+
+void AudioFlinger::TrackHandle::flush() {
+    mTrack->flush();
+}
+
+void AudioFlinger::TrackHandle::mute(bool e) {
+    mTrack->mute(e);
+}
+
+void AudioFlinger::TrackHandle::pause() {
+    mTrack->pause();
+}
+
+void AudioFlinger::TrackHandle::setVolume(float left, float right) {
+    mTrack->setVolume(left, right);
+}
+
+sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
+    return mTrack->getCblk();
+}
+
+status_t AudioFlinger::TrackHandle::onTransact(
+    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    return BnAudioTrack::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+sp<IAudioRecord> AudioFlinger::openRecord(
+        pid_t pid,
+        int streamType,
+        uint32_t sampleRate,
+        int format,
+        int channelCount,
+        int frameCount,
+        uint32_t flags,
+        status_t *status)
+{
+    sp<AudioRecordThread> thread;
+    sp<MixerThread::RecordTrack> recordTrack;
+    sp<RecordHandle> recordHandle;
+    sp<Client> client;
+    wp<Client> wclient;
+    AudioStreamIn* input = 0;
+    int inFrameCount;
+    size_t inputBufferSize;
+    status_t lStatus;
+
+    // check calling permissions
+    if (!recordingAllowed()) {
+        lStatus = PERMISSION_DENIED;
+        goto Exit;
+    }
+
+    if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) {
+        LOGE("invalid stream type");
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    if (sampleRate > MAX_SAMPLE_RATE) {
+        LOGE("Sample rate out of range");
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    if (mAudioRecordThread == 0) {
+        LOGE("Audio record thread not started");
+        lStatus = NO_INIT;
+        goto Exit;
+    }
+
+
+    // Check that audio input stream accepts requested audio parameters 
+    inputBufferSize = mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
+    if (inputBufferSize == 0) {
+        lStatus = BAD_VALUE;
+        LOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d",  sampleRate, format, channelCount);
+        goto Exit;
+    }
+
+    // add client to list
+    {
+        Mutex::Autolock _l(mLock);
+        wclient = mClients.valueFor(pid);
+        if (wclient != NULL) {
+            client = wclient.promote();
+        } else {
+            client = new Client(this, pid);
+            mClients.add(pid, client);
+        }
+    }
+
+    // frameCount must be a multiple of input buffer size
+    inFrameCount = inputBufferSize/channelCount/sizeof(short);
+    frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
+
+    // create new record track and pass to record thread
+    recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, streamType, sampleRate,
+                                               format, channelCount, frameCount, flags);
+    if (recordTrack->getCblk() == NULL) {
+        recordTrack.clear();
+        lStatus = NO_MEMORY;
+        goto Exit;
+    }
+
+    // return to handle to client
+    recordHandle = new RecordHandle(recordTrack);
+    lStatus = NO_ERROR;
+
+Exit:
+    if (status) {
+        *status = lStatus;
+    }
+    return recordHandle;
+}
+
+status_t AudioFlinger::startRecord(MixerThread::RecordTrack* recordTrack) {
+    if (mAudioRecordThread != 0) {
+        return mAudioRecordThread->start(recordTrack);        
+    }
+    return NO_INIT;
+}
+
+void AudioFlinger::stopRecord(MixerThread::RecordTrack* recordTrack) {
+    if (mAudioRecordThread != 0) {
+        mAudioRecordThread->stop(recordTrack);
+    }
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::MixerThread::RecordTrack>& recordTrack)
+    : BnAudioRecord(),
+    mRecordTrack(recordTrack)
+{
+}
+
+AudioFlinger::RecordHandle::~RecordHandle() {
+    stop();
+}
+
+status_t AudioFlinger::RecordHandle::start() {
+    LOGV("RecordHandle::start()");
+    return mRecordTrack->start();
+}
+
+void AudioFlinger::RecordHandle::stop() {
+    LOGV("RecordHandle::stop()");
+    mRecordTrack->stop();
+}
+
+sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
+    return mRecordTrack->getCblk();
+}
+
+status_t AudioFlinger::RecordHandle::onTransact(
+    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    return BnAudioRecord::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::AudioRecordThread::AudioRecordThread(AudioHardwareInterface* audioHardware) :
+    mAudioHardware(audioHardware),
+    mActive(false)
+{
+}
+
+AudioFlinger::AudioRecordThread::~AudioRecordThread()
+{
+}
+
+bool AudioFlinger::AudioRecordThread::threadLoop()
+{
+    LOGV("AudioRecordThread: start record loop");
+    AudioBufferProvider::Buffer buffer;
+    int inBufferSize = 0;
+    int inFrameCount = 0;
+    AudioStreamIn* input = 0;
+
+    mActive = 0;
+    
+    // start recording
+    while (!exitPending()) {
+        if (!mActive) {
+            mLock.lock();
+            if (!mActive && !exitPending()) {
+                LOGV("AudioRecordThread: loop stopping");
+                if (input) {
+                    delete input;
+                    input = 0;
+                }
+                mRecordTrack.clear();
+                mStopped.signal();
+
+                mWaitWorkCV.wait(mLock);
+               
+                LOGV("AudioRecordThread: loop starting");
+                if (mRecordTrack != 0) {
+                    input = mAudioHardware->openInputStream(mRecordTrack->format(), 
+                                    mRecordTrack->channelCount(), 
+                                    mRecordTrack->sampleRate(), 
+                                    &mStartStatus,
+                                    (AudioSystem::audio_in_acoustics)(mRecordTrack->mFlags >> 16));
+                    if (input != 0) {
+                        inBufferSize = input->bufferSize();
+                        inFrameCount = inBufferSize/input->frameSize();                        
+                    }
+                } else {
+                    mStartStatus = NO_INIT;
+                }
+                if (mStartStatus !=NO_ERROR) {
+                    LOGW("record start failed, status %d", mStartStatus);
+                    mActive = false;
+                    mRecordTrack.clear();                    
+                }
+                mWaitWorkCV.signal();
+            }
+            mLock.unlock();
+        } else if (mRecordTrack != 0) {
+
+            buffer.frameCount = inFrameCount;
+            if (LIKELY(mRecordTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+                LOGV("AudioRecordThread read: %d frames", buffer.frameCount);
+                ssize_t bytesRead = input->read(buffer.raw, inBufferSize);
+                if (bytesRead < 0) {
+                    LOGE("Error reading audio input");
+                    sleep(1);
+                }
+                mRecordTrack->releaseBuffer(&buffer);
+                mRecordTrack->overflow();
+            }
+
+            // client isn't retrieving buffers fast enough
+            else {
+                if (!mRecordTrack->setOverflow())
+                    LOGW("AudioRecordThread: buffer overflow");
+                // Release the processor for a while before asking for a new buffer.
+                // This will give the application more chance to read from the buffer and
+                // clear the overflow.
+                usleep(5000);
+            }
+        }
+    }
+
+
+    if (input) {
+        delete input;
+    }
+    mRecordTrack.clear();
+    
+    return false;
+}
+
+status_t AudioFlinger::AudioRecordThread::start(MixerThread::RecordTrack* recordTrack)
+{
+    LOGV("AudioRecordThread::start");
+    AutoMutex lock(&mLock);
+    mActive = true;
+    // If starting the active track, just reset mActive in case a stop
+    // was pending and exit
+    if (recordTrack == mRecordTrack.get()) return NO_ERROR;
+
+    if (mRecordTrack != 0) return -EBUSY;
+
+    mRecordTrack = recordTrack;
+
+    // signal thread to start
+    LOGV("Signal record thread");
+    mWaitWorkCV.signal();
+    mWaitWorkCV.wait(mLock);
+    LOGV("Record started, status %d", mStartStatus);
+    return mStartStatus;
+}
+
+void AudioFlinger::AudioRecordThread::stop(MixerThread::RecordTrack* recordTrack) {
+    LOGV("AudioRecordThread::stop");
+    AutoMutex lock(&mLock);
+    if (mActive && (recordTrack == mRecordTrack.get())) {
+        mActive = false;
+        mStopped.wait(mLock);
+    }
+}
+
+void AudioFlinger::AudioRecordThread::exit()
+{
+    LOGV("AudioRecordThread::exit");
+    {
+        AutoMutex lock(&mLock);
+        requestExit();
+        mWaitWorkCV.signal();
+    }
+    requestExitAndWait();
+}
+
+status_t AudioFlinger::AudioRecordThread::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    pid_t pid = 0;
+
+    if (mRecordTrack != 0 && mRecordTrack->mClient != 0) {
+        snprintf(buffer, SIZE, "Record client pid: %d\n", mRecordTrack->mClient->pid());
+        result.append(buffer);
+    } else {
+        result.append("No record client\n");
+    }
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::onTransact(
+        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    return BnAudioFlinger::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+void AudioFlinger::instantiate() {
+    defaultServiceManager()->addService(
+            String16("media.audio_flinger"), new AudioFlinger());
+}
+
+}; // namespace android