Merge "move native services under services/" into gingerbread

1 files changed, 6078 insertions, 0 deletions

diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
new file mode 100644
index 0000000..97eb6c0
--- /dev/null
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -0,0 +1,6078 @@
+/* //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 <binder/IServiceManager.h>
+#include <utils/Log.h>
+#include <binder/Parcel.h>
+#include <binder/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 <private/media/AudioEffectShared.h>
+#include <hardware_legacy/AudioHardwareInterface.h>
+
+#include "AudioMixer.h"
+#include "AudioFlinger.h"
+
+#ifdef WITH_A2DP
+#include "A2dpAudioInterface.h"
+#endif
+
+#ifdef LVMX
+#include "lifevibes.h"
+#endif
+
+#include <media/EffectsFactoryApi.h>
+#include <media/EffectVisualizerApi.h>
+
+// ----------------------------------------------------------------------------
+// the sim build doesn't have gettid
+
+#ifndef HAVE_GETTID
+# define gettid getpid
+#endif
+
+// ----------------------------------------------------------------------------
+
+namespace android {
+
+static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
+static const char* kHardwareLockedString = "Hardware lock is taken\n";
+
+//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
+static const float MAX_GAIN = 4096.0f;
+static const float MAX_GAIN_INT = 0x1000;
+
+// retry counts for buffer fill timeout
+// 50 * ~20msecs = 1 second
+static const int8_t kMaxTrackRetries = 50;
+static const int8_t kMaxTrackStartupRetries = 50;
+// allow less retry attempts on direct output thread.
+// direct outputs can be a scarce resource in audio hardware and should
+// be released as quickly as possible.
+static const int8_t kMaxTrackRetriesDirect = 2;
+
+static const int kDumpLockRetries = 50;
+static const int kDumpLockSleep = 20000;
+
+static const nsecs_t kWarningThrottle = seconds(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), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
+        mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0)
+{
+    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
+        mMode = AudioSystem::MODE_NORMAL;
+        setMode(mMode);
+
+        setMasterVolume(1.0f);
+        setMasterMute(false);
+    } else {
+        LOGE("Couldn't even initialize the stubbed audio hardware!");
+    }
+#ifdef LVMX
+    LifeVibes::init();
+    mLifeVibesClientPid = -1;
+#endif
+}
+
+AudioFlinger::~AudioFlinger()
+{
+    while (!mRecordThreads.isEmpty()) {
+        // closeInput() will remove first entry from mRecordThreads
+        closeInput(mRecordThreads.keyAt(0));
+    }
+    while (!mPlaybackThreads.isEmpty()) {
+        // closeOutput() will remove first entry from mPlaybackThreads
+        closeOutput(mPlaybackThreads.keyAt(0));
+    }
+    if (mAudioHardware) {
+        delete mAudioHardware;
+    }
+}
+
+
+
+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;
+    int hardwareStatus = mHardwareStatus;
+
+    snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
+    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;
+}
+
+static bool tryLock(Mutex& mutex)
+{
+    bool locked = false;
+    for (int i = 0; i < kDumpLockRetries; ++i) {
+        if (mutex.tryLock() == NO_ERROR) {
+            locked = true;
+            break;
+        }
+        usleep(kDumpLockSleep);
+    }
+    return locked;
+}
+
+status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
+{
+    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+        dumpPermissionDenial(fd, args);
+    } else {
+        // get state of hardware lock
+        bool hardwareLocked = tryLock(mHardwareLock);
+        if (!hardwareLocked) {
+            String8 result(kHardwareLockedString);
+            write(fd, result.string(), result.size());
+        } else {
+            mHardwareLock.unlock();
+        }
+
+        bool locked = tryLock(mLock);
+
+        // failed to lock - AudioFlinger is probably deadlocked
+        if (!locked) {
+            String8 result(kDeadlockedString);
+            write(fd, result.string(), result.size());
+        }
+
+        dumpClients(fd, args);
+        dumpInternals(fd, args);
+
+        // dump playback threads
+        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+            mPlaybackThreads.valueAt(i)->dump(fd, args);
+        }
+
+        // dump record threads
+        for (size_t i = 0; i < mRecordThreads.size(); i++) {
+            mRecordThreads.valueAt(i)->dump(fd, args);
+        }
+
+        if (mAudioHardware) {
+            mAudioHardware->dumpState(fd, args);
+        }
+        if (locked) mLock.unlock();
+    }
+    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,
+        int output,
+        int *sessionId,
+        status_t *status)
+{
+    sp<PlaybackThread::Track> track;
+    sp<TrackHandle> trackHandle;
+    sp<Client> client;
+    wp<Client> wclient;
+    status_t lStatus;
+    int lSessionId;
+
+    if (streamType >= AudioSystem::NUM_STREAM_TYPES) {
+        LOGE("invalid stream type");
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    {
+        Mutex::Autolock _l(mLock);
+        PlaybackThread *thread = checkPlaybackThread_l(output);
+        if (thread == NULL) {
+            LOGE("unknown output thread");
+            lStatus = BAD_VALUE;
+            goto Exit;
+        }
+
+        wclient = mClients.valueFor(pid);
+
+        if (wclient != NULL) {
+            client = wclient.promote();
+        } else {
+            client = new Client(this, pid);
+            mClients.add(pid, client);
+        }
+
+        // If no audio session id is provided, create one here
+        // TODO: enforce same stream type for all tracks in same audio session?
+        // TODO: prevent same audio session on different output threads
+        LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
+        if (sessionId != NULL && *sessionId != 0) {
+            lSessionId = *sessionId;
+        } else {
+            lSessionId = nextUniqueId();
+            if (sessionId != NULL) {
+                *sessionId = lSessionId;
+            }
+        }
+        LOGV("createTrack() lSessionId: %d", lSessionId);
+
+        track = thread->createTrack_l(client, streamType, sampleRate, format,
+                channelCount, frameCount, sharedBuffer, lSessionId, &lStatus);
+    }
+    if (lStatus == NO_ERROR) {
+        trackHandle = new TrackHandle(track);
+    } else {
+        // remove local strong reference to Client before deleting the Track so that the Client
+        // destructor is called by the TrackBase destructor with mLock held
+        client.clear();
+        track.clear();
+    }
+
+Exit:
+    if(status) {
+        *status = lStatus;
+    }
+    return trackHandle;
+}
+
+uint32_t AudioFlinger::sampleRate(int output) const
+{
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread == NULL) {
+        LOGW("sampleRate() unknown thread %d", output);
+        return 0;
+    }
+    return thread->sampleRate();
+}
+
+int AudioFlinger::channelCount(int output) const
+{
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread == NULL) {
+        LOGW("channelCount() unknown thread %d", output);
+        return 0;
+    }
+    return thread->channelCount();
+}
+
+int AudioFlinger::format(int output) const
+{
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread == NULL) {
+        LOGW("format() unknown thread %d", output);
+        return 0;
+    }
+    return thread->format();
+}
+
+size_t AudioFlinger::frameCount(int output) const
+{
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread == NULL) {
+        LOGW("frameCount() unknown thread %d", output);
+        return 0;
+    }
+    return thread->frameCount();
+}
+
+uint32_t AudioFlinger::latency(int output) const
+{
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread == NULL) {
+        LOGW("latency() unknown thread %d", output);
+        return 0;
+    }
+    return thread->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;
+
+    mMasterVolume = value;
+    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+       mPlaybackThreads.valueAt(i)->setMasterVolume(value);
+
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::setMode(int mode)
+{
+    status_t ret;
+
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
+        LOGW("Illegal value: setMode(%d)", mode);
+        return BAD_VALUE;
+    }
+
+    { // scope for the lock
+        AutoMutex lock(mHardwareLock);
+        mHardwareStatus = AUDIO_HW_SET_MODE;
+        ret = mAudioHardware->setMode(mode);
+        mHardwareStatus = AUDIO_HW_IDLE;
+    }
+
+    if (NO_ERROR == ret) {
+        Mutex::Autolock _l(mLock);
+        mMode = mode;
+        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+           mPlaybackThreads.valueAt(i)->setMode(mode);
+#ifdef LVMX
+        LifeVibes::setMode(mode);
+#endif
+    }
+
+    return ret;
+}
+
+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;
+    }
+
+    mMasterMute = muted;
+    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+       mPlaybackThreads.valueAt(i)->setMasterMute(muted);
+
+    return NO_ERROR;
+}
+
+float AudioFlinger::masterVolume() const
+{
+    return mMasterVolume;
+}
+
+bool AudioFlinger::masterMute() const
+{
+    return mMasterMute;
+}
+
+status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+        return BAD_VALUE;
+    }
+
+    AutoMutex lock(mLock);
+    PlaybackThread *thread = NULL;
+    if (output) {
+        thread = checkPlaybackThread_l(output);
+        if (thread == NULL) {
+            return BAD_VALUE;
+        }
+    }
+
+    mStreamTypes[stream].volume = value;
+
+    if (thread == NULL) {
+        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
+           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
+        }
+    } else {
+        thread->setStreamVolume(stream, value);
+    }
+
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::setStreamMute(int stream, bool muted)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
+        uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
+        return BAD_VALUE;
+    }
+
+    mStreamTypes[stream].mute = muted;
+    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
+
+    return NO_ERROR;
+}
+
+float AudioFlinger::streamVolume(int stream, int output) const
+{
+    if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+        return 0.0f;
+    }
+
+    AutoMutex lock(mLock);
+    float volume;
+    if (output) {
+        PlaybackThread *thread = checkPlaybackThread_l(output);
+        if (thread == NULL) {
+            return 0.0f;
+        }
+        volume = thread->streamVolume(stream);
+    } else {
+        volume = mStreamTypes[stream].volume;
+    }
+
+    return volume;
+}
+
+bool AudioFlinger::streamMute(int stream) const
+{
+    if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) {
+        return true;
+    }
+
+    return mStreamTypes[stream].mute;
+}
+
+bool AudioFlinger::isStreamActive(int stream) const
+{
+    Mutex::Autolock _l(mLock);
+    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
+        if (mPlaybackThreads.valueAt(i)->isStreamActive(stream)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
+{
+    status_t result;
+
+    LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
+            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+#ifdef LVMX
+    AudioParameter param = AudioParameter(keyValuePairs);
+    LifeVibes::setParameters(ioHandle,keyValuePairs);
+    String8 key = String8(AudioParameter::keyRouting);
+    int device;
+    if (NO_ERROR != param.getInt(key, device)) {
+        device = -1;
+    }
+
+    key = String8(LifevibesTag);
+    String8 value;
+    int musicEnabled = -1;
+    if (NO_ERROR == param.get(key, value)) {
+        if (value == LifevibesEnable) {
+            mLifeVibesClientPid = IPCThreadState::self()->getCallingPid();
+            musicEnabled = 1;
+        } else if (value == LifevibesDisable) {
+            mLifeVibesClientPid = -1;
+            musicEnabled = 0;
+        }
+    }
+#endif
+
+    // ioHandle == 0 means the parameters are global to the audio hardware interface
+    if (ioHandle == 0) {
+        AutoMutex lock(mHardwareLock);
+        mHardwareStatus = AUDIO_SET_PARAMETER;
+        result = mAudioHardware->setParameters(keyValuePairs);
+#ifdef LVMX
+        if (musicEnabled != -1) {
+            LifeVibes::enableMusic((bool) musicEnabled);
+        }
+#endif
+        mHardwareStatus = AUDIO_HW_IDLE;
+        return result;
+    }
+
+    // hold a strong ref on thread in case closeOutput() or closeInput() is called
+    // and the thread is exited once the lock is released
+    sp<ThreadBase> thread;
+    {
+        Mutex::Autolock _l(mLock);
+        thread = checkPlaybackThread_l(ioHandle);
+        if (thread == NULL) {
+            thread = checkRecordThread_l(ioHandle);
+        }
+    }
+    if (thread != NULL) {
+        result = thread->setParameters(keyValuePairs);
+#ifdef LVMX
+        if ((NO_ERROR == result) && (device != -1)) {
+            LifeVibes::setDevice(LifeVibes::threadIdToAudioOutputType(thread->id()), device);
+        }
+#endif
+        return result;
+    }
+    return BAD_VALUE;
+}
+
+String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
+{
+//    LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
+//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
+
+    if (ioHandle == 0) {
+        return mAudioHardware->getParameters(keys);
+    }
+
+    Mutex::Autolock _l(mLock);
+
+    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
+    if (playbackThread != NULL) {
+        return playbackThread->getParameters(keys);
+    }
+    RecordThread *recordThread = checkRecordThread_l(ioHandle);
+    if (recordThread != NULL) {
+        return recordThread->getParameters(keys);
+    }
+    return String8("");
+}
+
+size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+{
+    return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
+}
+
+unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
+{
+    if (ioHandle == 0) {
+        return 0;
+    }
+
+    Mutex::Autolock _l(mLock);
+
+    RecordThread *recordThread = checkRecordThread_l(ioHandle);
+    if (recordThread != NULL) {
+        return recordThread->getInputFramesLost();
+    }
+    return 0;
+}
+
+status_t AudioFlinger::setVoiceVolume(float value)
+{
+    // check calling permissions
+    if (!settingsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+
+    AutoMutex lock(mHardwareLock);
+    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
+    status_t ret = mAudioHardware->setVoiceVolume(value);
+    mHardwareStatus = AUDIO_HW_IDLE;
+
+    return ret;
+}
+
+status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
+{
+    status_t status;
+
+    Mutex::Autolock _l(mLock);
+
+    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
+    if (playbackThread != NULL) {
+        return playbackThread->getRenderPosition(halFrames, dspFrames);
+    }
+
+    return BAD_VALUE;
+}
+
+void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
+{
+
+    Mutex::Autolock _l(mLock);
+
+    int pid = IPCThreadState::self()->getCallingPid();
+    if (mNotificationClients.indexOfKey(pid) < 0) {
+        sp<NotificationClient> notificationClient = new NotificationClient(this,
+                                                                            client,
+                                                                            pid);
+        LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
+
+        mNotificationClients.add(pid, notificationClient);
+
+        sp<IBinder> binder = client->asBinder();
+        binder->linkToDeath(notificationClient);
+
+        // the config change is always sent from playback or record threads to avoid deadlock
+        // with AudioSystem::gLock
+        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
+        }
+
+        for (size_t i = 0; i < mRecordThreads.size(); i++) {
+            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
+        }
+    }
+}
+
+void AudioFlinger::removeNotificationClient(pid_t pid)
+{
+    Mutex::Autolock _l(mLock);
+
+    int index = mNotificationClients.indexOfKey(pid);
+    if (index >= 0) {
+        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
+        LOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
+#ifdef LVMX
+        if (pid == mLifeVibesClientPid) {
+            LOGV("Disabling lifevibes");
+            LifeVibes::enableMusic(false);
+            mLifeVibesClientPid = -1;
+        }
+#endif
+        mNotificationClients.removeItem(pid);
+    }
+}
+
+// audioConfigChanged_l() must be called with AudioFlinger::mLock held
+void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
+{
+    size_t size = mNotificationClients.size();
+    for (size_t i = 0; i < size; i++) {
+        mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
+    }
+}
+
+// removeClient_l() must be called with AudioFlinger::mLock held
+void AudioFlinger::removeClient_l(pid_t pid)
+{
+    LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
+    mClients.removeItem(pid);
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id)
+    :   Thread(false),
+        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
+        mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false)
+{
+}
+
+AudioFlinger::ThreadBase::~ThreadBase()
+{
+    mParamCond.broadcast();
+    mNewParameters.clear();
+}
+
+void AudioFlinger::ThreadBase::exit()
+{
+    // keep a strong ref on ourself so that we wont get
+    // destroyed in the middle of requestExitAndWait()
+    sp <ThreadBase> strongMe = this;
+
+    LOGV("ThreadBase::exit");
+    {
+        AutoMutex lock(&mLock);
+        mExiting = true;
+        requestExit();
+        mWaitWorkCV.signal();
+    }
+    requestExitAndWait();
+}
+
+uint32_t AudioFlinger::ThreadBase::sampleRate() const
+{
+    return mSampleRate;
+}
+
+int AudioFlinger::ThreadBase::channelCount() const
+{
+    return (int)mChannelCount;
+}
+
+int AudioFlinger::ThreadBase::format() const
+{
+    return mFormat;
+}
+
+size_t AudioFlinger::ThreadBase::frameCount() const
+{
+    return mFrameCount;
+}
+
+status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
+{
+    status_t status;
+
+    LOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
+    Mutex::Autolock _l(mLock);
+
+    mNewParameters.add(keyValuePairs);
+    mWaitWorkCV.signal();
+    // wait condition with timeout in case the thread loop has exited
+    // before the request could be processed
+    if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) {
+        status = mParamStatus;
+        mWaitWorkCV.signal();
+    } else {
+        status = TIMED_OUT;
+    }
+    return status;
+}
+
+void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
+{
+    Mutex::Autolock _l(mLock);
+    sendConfigEvent_l(event, param);
+}
+
+// sendConfigEvent_l() must be called with ThreadBase::mLock held
+void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
+{
+    ConfigEvent *configEvent = new ConfigEvent();
+    configEvent->mEvent = event;
+    configEvent->mParam = param;
+    mConfigEvents.add(configEvent);
+    LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
+    mWaitWorkCV.signal();
+}
+
+void AudioFlinger::ThreadBase::processConfigEvents()
+{
+    mLock.lock();
+    while(!mConfigEvents.isEmpty()) {
+        LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
+        ConfigEvent *configEvent = mConfigEvents[0];
+        mConfigEvents.removeAt(0);
+        // release mLock before locking AudioFlinger mLock: lock order is always
+        // AudioFlinger then ThreadBase to avoid cross deadlock
+        mLock.unlock();
+        mAudioFlinger->mLock.lock();
+        audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
+        mAudioFlinger->mLock.unlock();
+        delete configEvent;
+        mLock.lock();
+    }
+    mLock.unlock();
+}
+
+status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    bool locked = tryLock(mLock);
+    if (!locked) {
+        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
+        write(fd, buffer, strlen(buffer));
+    }
+
+    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
+    result.append(buffer);
+
+    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
+    result.append(buffer);
+    result.append(" Index Command");
+    for (size_t i = 0; i < mNewParameters.size(); ++i) {
+        snprintf(buffer, SIZE, "\n %02d    ", i);
+        result.append(buffer);
+        result.append(mNewParameters[i]);
+    }
+
+    snprintf(buffer, SIZE, "\n\nPending config events: \n");
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Index event param\n");
+    result.append(buffer);
+    for (size_t i = 0; i < mConfigEvents.size(); i++) {
+        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
+        result.append(buffer);
+    }
+    result.append("\n");
+
+    write(fd, result.string(), result.size());
+
+    if (locked) {
+        mLock.unlock();
+    }
+    return NO_ERROR;
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
+    :   ThreadBase(audioFlinger, id),
+        mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
+        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
+        mDevice(device)
+{
+    readOutputParameters();
+
+    mMasterVolume = mAudioFlinger->masterVolume();
+    mMasterMute = mAudioFlinger->masterMute();
+
+    for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
+        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
+    }
+}
+
+AudioFlinger::PlaybackThread::~PlaybackThread()
+{
+    delete [] mMixBuffer;
+}
+
+status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
+{
+    dumpInternals(fd, args);
+    dumpTracks(fd, args);
+    dumpEffectChains(fd, args);
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
+    result.append(buffer);
+    result.append("   Name  Clien Typ Fmt Chn Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<Track> track = mTracks[i];
+        if (track != 0) {
+            track->dump(buffer, SIZE);
+            result.append(buffer);
+        }
+    }
+
+    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
+    result.append(buffer);
+    result.append("   Name  Clien Typ Fmt Chn Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
+    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
+        wp<Track> wTrack = mActiveTracks[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::PlaybackThread::dumpEffectChains(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
+    write(fd, buffer, strlen(buffer));
+
+    for (size_t i = 0; i < mEffectChains.size(); ++i) {
+        sp<EffectChain> chain = mEffectChains[i];
+        if (chain != 0) {
+            chain->dump(fd, args);
+        }
+    }
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
+    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, "suspend count: %d\n", mSuspended);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    dumpBase(fd, args);
+
+    return NO_ERROR;
+}
+
+// Thread virtuals
+status_t AudioFlinger::PlaybackThread::readyToRun()
+{
+    if (mSampleRate == 0) {
+        LOGE("No working audio driver found.");
+        return NO_INIT;
+    }
+    LOGI("AudioFlinger's thread %p ready to run", this);
+    return NO_ERROR;
+}
+
+void AudioFlinger::PlaybackThread::onFirstRef()
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+
+    snprintf(buffer, SIZE, "Playback Thread %p", this);
+
+    run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
+}
+
+// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
+        const sp<AudioFlinger::Client>& client,
+        int streamType,
+        uint32_t sampleRate,
+        int format,
+        int channelCount,
+        int frameCount,
+        const sp<IMemory>& sharedBuffer,
+        int sessionId,
+        status_t *status)
+{
+    sp<Track> track;
+    status_t lStatus;
+
+    if (mType == DIRECT) {
+        if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) {
+            LOGE("createTrack_l() Bad parameter:  sampleRate %d format %d, channelCount %d for output %p",
+                 sampleRate, format, channelCount, mOutput);
+            lStatus = BAD_VALUE;
+            goto Exit;
+        }
+    } else {
+        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
+        if (sampleRate > mSampleRate*2) {
+            LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
+            lStatus = BAD_VALUE;
+            goto Exit;
+        }
+    }
+
+    if (mOutput == 0) {
+        LOGE("Audio driver not initialized.");
+        lStatus = NO_INIT;
+        goto Exit;
+    }
+
+    { // scope for mLock
+        Mutex::Autolock _l(mLock);
+        track = new Track(this, client, streamType, sampleRate, format,
+                channelCount, frameCount, sharedBuffer, sessionId);
+        if (track->getCblk() == NULL || track->name() < 0) {
+            lStatus = NO_MEMORY;
+            goto Exit;
+        }
+        mTracks.add(track);
+
+        sp<EffectChain> chain = getEffectChain_l(sessionId);
+        if (chain != 0) {
+            LOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
+            track->setMainBuffer(chain->inBuffer());
+        }
+    }
+    lStatus = NO_ERROR;
+
+Exit:
+    if(status) {
+        *status = lStatus;
+    }
+    return track;
+}
+
+uint32_t AudioFlinger::PlaybackThread::latency() const
+{
+    if (mOutput) {
+        return mOutput->latency();
+    }
+    else {
+        return 0;
+    }
+}
+
+status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
+{
+#ifdef LVMX
+    int audioOutputType = LifeVibes::getMixerType(mId, mType);
+    if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+        LifeVibes::setMasterVolume(audioOutputType, value);
+    }
+#endif
+    mMasterVolume = value;
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
+{
+#ifdef LVMX
+    int audioOutputType = LifeVibes::getMixerType(mId, mType);
+    if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+        LifeVibes::setMasterMute(audioOutputType, muted);
+    }
+#endif
+    mMasterMute = muted;
+    return NO_ERROR;
+}
+
+float AudioFlinger::PlaybackThread::masterVolume() const
+{
+    return mMasterVolume;
+}
+
+bool AudioFlinger::PlaybackThread::masterMute() const
+{
+    return mMasterMute;
+}
+
+status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
+{
+#ifdef LVMX
+    int audioOutputType = LifeVibes::getMixerType(mId, mType);
+    if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+        LifeVibes::setStreamVolume(audioOutputType, stream, value);
+    }
+#endif
+    mStreamTypes[stream].volume = value;
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
+{
+#ifdef LVMX
+    int audioOutputType = LifeVibes::getMixerType(mId, mType);
+    if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+        LifeVibes::setStreamMute(audioOutputType, stream, muted);
+    }
+#endif
+    mStreamTypes[stream].mute = muted;
+    return NO_ERROR;
+}
+
+float AudioFlinger::PlaybackThread::streamVolume(int stream) const
+{
+    return mStreamTypes[stream].volume;
+}
+
+bool AudioFlinger::PlaybackThread::streamMute(int stream) const
+{
+    return mStreamTypes[stream].mute;
+}
+
+bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const
+{
+    Mutex::Autolock _l(mLock);
+    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->type() == stream)
+            return true;
+    }
+    return false;
+}
+
+// addTrack_l() must be called with ThreadBase::mLock held
+status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
+{
+    status_t status = ALREADY_EXISTS;
+
+    // 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;
+        mActiveTracks.add(track);
+        if (track->mainBuffer() != mMixBuffer) {
+            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
+            if (chain != 0) {
+                LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
+                chain->startTrack();
+            }
+        }
+
+        status = NO_ERROR;
+    }
+
+    LOGV("mWaitWorkCV.broadcast");
+    mWaitWorkCV.broadcast();
+
+    return status;
+}
+
+// destroyTrack_l() must be called with ThreadBase::mLock held
+void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
+{
+    track->mState = TrackBase::TERMINATED;
+    if (mActiveTracks.indexOf(track) < 0) {
+        mTracks.remove(track);
+        deleteTrackName_l(track->name());
+    }
+}
+
+String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
+{
+    return mOutput->getParameters(keys);
+}
+
+// destroyTrack_l() must be called with AudioFlinger::mLock held
+void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
+    AudioSystem::OutputDescriptor desc;
+    void *param2 = 0;
+
+    LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
+
+    switch (event) {
+    case AudioSystem::OUTPUT_OPENED:
+    case AudioSystem::OUTPUT_CONFIG_CHANGED:
+        desc.channels = mChannels;
+        desc.samplingRate = mSampleRate;
+        desc.format = mFormat;
+        desc.frameCount = mFrameCount;
+        desc.latency = latency();
+        param2 = &desc;
+        break;
+
+    case AudioSystem::STREAM_CONFIG_CHANGED:
+        param2 = &param;
+    case AudioSystem::OUTPUT_CLOSED:
+    default:
+        break;
+    }
+    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
+}
+
+void AudioFlinger::PlaybackThread::readOutputParameters()
+{
+    mSampleRate = mOutput->sampleRate();
+    mChannels = mOutput->channels();
+    mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);
+    mFormat = mOutput->format();
+    mFrameSize = (uint16_t)mOutput->frameSize();
+    mFrameCount = mOutput->bufferSize() / mFrameSize;
+
+    // FIXME - Current mixer implementation only supports stereo output: Always
+    // Allocate a stereo buffer even if HW output is mono.
+    if (mMixBuffer != NULL) delete[] mMixBuffer;
+    mMixBuffer = new int16_t[mFrameCount * 2];
+    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
+
+    //TODO handle effects reconfig
+}
+
+status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
+{
+    if (halFrames == 0 || dspFrames == 0) {
+        return BAD_VALUE;
+    }
+    if (mOutput == 0) {
+        return INVALID_OPERATION;
+    }
+    *halFrames = mBytesWritten/mOutput->frameSize();
+
+    return mOutput->getRenderPosition(dspFrames);
+}
+
+bool AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
+{
+    Mutex::Autolock _l(mLock);
+    if (getEffectChain_l(sessionId) != 0) {
+        return true;
+    }
+
+    for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<Track> track = mTracks[i];
+        if (sessionId == track->sessionId()) {
+            return true;
+        }
+    }
+
+    return false;
+}
+
+sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId)
+{
+    Mutex::Autolock _l(mLock);
+    return getEffectChain_l(sessionId);
+}
+
+sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId)
+{
+    sp<EffectChain> chain;
+
+    size_t size = mEffectChains.size();
+    for (size_t i = 0; i < size; i++) {
+        if (mEffectChains[i]->sessionId() == sessionId) {
+            chain = mEffectChains[i];
+            break;
+        }
+    }
+    return chain;
+}
+
+void AudioFlinger::PlaybackThread::setMode(uint32_t mode)
+{
+    Mutex::Autolock _l(mLock);
+    size_t size = mEffectChains.size();
+    for (size_t i = 0; i < size; i++) {
+        mEffectChains[i]->setMode(mode);
+    }
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
+    :   PlaybackThread(audioFlinger, output, id, device),
+        mAudioMixer(0)
+{
+    mType = PlaybackThread::MIXER;
+    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
+
+    // FIXME - Current mixer implementation only supports stereo output
+    if (mChannelCount == 1) {
+        LOGE("Invalid audio hardware channel count");
+    }
+}
+
+AudioFlinger::MixerThread::~MixerThread()
+{
+    delete mAudioMixer;
+}
+
+bool AudioFlinger::MixerThread::threadLoop()
+{
+    Vector< sp<Track> > tracksToRemove;
+    uint32_t mixerStatus = MIXER_IDLE;
+    nsecs_t standbyTime = systemTime();
+    size_t mixBufferSize = mFrameCount * mFrameSize;
+    // FIXME: Relaxed timing because of a certain device that can't meet latency
+    // Should be reduced to 2x after the vendor fixes the driver issue
+    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
+    nsecs_t lastWarning = 0;
+    bool longStandbyExit = false;
+    uint32_t activeSleepTime = activeSleepTimeUs();
+    uint32_t idleSleepTime = idleSleepTimeUs();
+    uint32_t sleepTime = idleSleepTime;
+    Vector< sp<EffectChain> > effectChains;
+
+    while (!exitPending())
+    {
+        processConfigEvents();
+
+        mixerStatus = MIXER_IDLE;
+        { // scope for mLock
+
+            Mutex::Autolock _l(mLock);
+
+            if (checkForNewParameters_l()) {
+                mixBufferSize = mFrameCount * mFrameSize;
+                // FIXME: Relaxed timing because of a certain device that can't meet latency
+                // Should be reduced to 2x after the vendor fixes the driver issue
+                maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
+                activeSleepTime = activeSleepTimeUs();
+                idleSleepTime = idleSleepTimeUs();
+            }
+
+            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+
+            // put audio hardware into standby after short delay
+            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+                        mSuspended) {
+                if (!mStandby) {
+                    LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
+                    mOutput->standby();
+                    mStandby = true;
+                    mBytesWritten = 0;
+                }
+
+                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
+                    // we're about to wait, flush the binder command buffer
+                    IPCThreadState::self()->flushCommands();
+
+                    if (exitPending()) break;
+
+                    // wait until we have something to do...
+                    LOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
+                    mWaitWorkCV.wait(mLock);
+                    LOGV("MixerThread %p TID %d waking up\n", this, gettid());
+
+                    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;
+                    sleepTime = idleSleepTime;
+                    continue;
+                }
+            }
+
+            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
+
+            // prevent any changes in effect chain list and in each effect chain
+            // during mixing and effect process as the audio buffers could be deleted
+            // or modified if an effect is created or deleted
+            effectChains = mEffectChains;
+            lockEffectChains_l();
+       }
+
+        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+            // mix buffers...
+            mAudioMixer->process();
+            sleepTime = 0;
+            standbyTime = systemTime() + kStandbyTimeInNsecs;
+            //TODO: delay standby when effects have a tail
+        } else {
+            // If no tracks are ready, sleep once for the duration of an output
+            // buffer size, then write 0s to the output
+            if (sleepTime == 0) {
+                if (mixerStatus == MIXER_TRACKS_ENABLED) {
+                    sleepTime = activeSleepTime;
+                } else {
+                    sleepTime = idleSleepTime;
+                }
+            } else if (mBytesWritten != 0 ||
+                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
+                memset (mMixBuffer, 0, mixBufferSize);
+                sleepTime = 0;
+                LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
+            }
+            // TODO add standby time extension fct of effect tail
+        }
+
+        if (mSuspended) {
+            sleepTime = idleSleepTime;
+        }
+        // sleepTime == 0 means we must write to audio hardware
+        if (sleepTime == 0) {
+             for (size_t i = 0; i < effectChains.size(); i ++) {
+                 effectChains[i]->process_l();
+             }
+             // enable changes in effect chain
+             unlockEffectChains();
+#ifdef LVMX
+            int audioOutputType = LifeVibes::getMixerType(mId, mType);
+            if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+               LifeVibes::process(audioOutputType, mMixBuffer, mixBufferSize);
+            }
+#endif
+            mLastWriteTime = systemTime();
+            mInWrite = true;
+            mBytesWritten += mixBufferSize;
+
+            int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
+            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
+            mNumWrites++;
+            mInWrite = false;
+            nsecs_t now = systemTime();
+            nsecs_t delta = now - mLastWriteTime;
+            if (delta > maxPeriod) {
+                mNumDelayedWrites++;
+                if ((now - lastWarning) > kWarningThrottle) {
+                    LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
+                            ns2ms(delta), mNumDelayedWrites, this);
+                    lastWarning = now;
+                }
+                if (mStandby) {
+                    longStandbyExit = true;
+                }
+            }
+            mStandby = false;
+        } else {
+            // enable changes in effect chain
+            unlockEffectChains();
+            usleep(sleepTime);
+        }
+
+        // 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();
+
+        // Effect chains will be actually deleted here if they were removed from
+        // mEffectChains list during mixing or effects processing
+        effectChains.clear();
+    }
+
+    if (!mStandby) {
+        mOutput->standby();
+    }
+
+    LOGV("MixerThread %p exiting", this);
+    return false;
+}
+
+// prepareTracks_l() must be called with ThreadBase::mLock held
+uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
+{
+
+    uint32_t mixerStatus = MIXER_IDLE;
+    // find out which tracks need to be processed
+    size_t count = activeTracks.size();
+    size_t mixedTracks = 0;
+    size_t tracksWithEffect = 0;
+
+    float masterVolume = mMasterVolume;
+    bool  masterMute = mMasterMute;
+
+#ifdef LVMX
+    bool tracksConnectedChanged = false;
+    bool stateChanged = false;
+
+    int audioOutputType = LifeVibes::getMixerType(mId, mType);
+    if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType))
+    {
+        int activeTypes = 0;
+        for (size_t i=0 ; i<count ; i++) {
+            sp<Track> t = activeTracks[i].promote();
+            if (t == 0) continue;
+            Track* const track = t.get();
+            int iTracktype=track->type();
+            activeTypes |= 1<<track->type();
+        }
+        LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute);
+    }
+#endif
+    // Delegate master volume control to effect in output mix effect chain if needed
+    sp<EffectChain> chain = getEffectChain_l(0);
+    if (chain != 0) {
+        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
+        chain->setVolume(&v, &v);
+        masterVolume = (float)((v + (1 << 23)) >> 24);
+        chain.clear();
+    }
+
+    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() && !track->isTerminated())
+        {
+            //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
+
+            mixedTracks++;
+
+            // track->mainBuffer() != mMixBuffer means there is an effect chain
+            // connected to the track
+            chain.clear();
+            if (track->mainBuffer() != mMixBuffer) {
+                chain = getEffectChain_l(track->sessionId());
+                // Delegate volume control to effect in track effect chain if needed
+                if (chain != 0) {
+                    tracksWithEffect++;
+                } else {
+                    LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
+                            track->name(), track->sessionId());
+                }
+            }
+
+
+            int param = AudioMixer::VOLUME;
+            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 if (cblk->server != 0) {
+                // If the track is stopped before the first frame was mixed,
+                // do not apply ramp
+                param = AudioMixer::RAMP_VOLUME;
+            }
+
+            // compute volume for this track
+            int16_t left, right, aux;
+            if (track->isMuted() || masterMute || track->isPausing() ||
+                mStreamTypes[track->type()].mute) {
+                left = right = aux = 0;
+                if (track->isPausing()) {
+                    track->setPaused();
+                }
+            } else {
+                // read original volumes with volume control
+                float typeVolume = mStreamTypes[track->type()].volume;
+#ifdef LVMX
+                bool streamMute=false;
+                // read the volume from the LivesVibes audio engine.
+                if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType))
+                {
+                    LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute);
+                    if (streamMute) {
+                        typeVolume = 0;
+                    }
+                }
+#endif
+                float v = masterVolume * typeVolume;
+                uint32_t vl = (uint32_t)(v * cblk->volume[0]) << 12;
+                uint32_t vr = (uint32_t)(v * cblk->volume[1]) << 12;
+
+                // Delegate volume control to effect in track effect chain if needed
+                if (chain != 0 && chain->setVolume(&vl, &vr)) {
+                    // Do not ramp volume is volume is controlled by effect
+                    param = AudioMixer::VOLUME;
+                }
+
+                // Convert volumes from 8.24 to 4.12 format
+                uint32_t v_clamped = (vl + (1 << 11)) >> 12;
+                if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
+                left = int16_t(v_clamped);
+                v_clamped = (vr + (1 << 11)) >> 12;
+                if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
+                right = int16_t(v_clamped);
+
+                v_clamped = (uint32_t)(v * cblk->sendLevel);
+                if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
+                aux = int16_t(v_clamped);
+            }
+
+#ifdef LVMX
+            if ( tracksConnectedChanged || stateChanged )
+            {
+                 // only do the ramp when the volume is changed by the user / application
+                 param = AudioMixer::VOLUME;
+            }
+#endif
+
+            // XXX: these things DON'T need to be done each time
+            mAudioMixer->setBufferProvider(track);
+            mAudioMixer->enable(AudioMixer::MIXING);
+
+            mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
+            mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
+            mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
+            mAudioMixer->setParameter(
+                AudioMixer::TRACK,
+                AudioMixer::FORMAT, (void *)track->format());
+            mAudioMixer->setParameter(
+                AudioMixer::TRACK,
+                AudioMixer::CHANNEL_COUNT, (void *)track->channelCount());
+            mAudioMixer->setParameter(
+                AudioMixer::RESAMPLE,
+                AudioMixer::SAMPLE_RATE,
+                (void *)(cblk->sampleRate));
+            mAudioMixer->setParameter(
+                AudioMixer::TRACK,
+                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
+            mAudioMixer->setParameter(
+                AudioMixer::TRACK,
+                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
+
+            // reset retry count
+            track->mRetryCount = kMaxTrackRetries;
+            mixerStatus = MIXER_TRACKS_READY;
+        } else {
+            //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
+            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.
+                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 on thread %p", track->name(), this);
+                    tracksToRemove->add(track);
+                } else if (mixerStatus != MIXER_TRACKS_READY) {
+                    mixerStatus = MIXER_TRACKS_ENABLED;
+                }
+            }
+            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->itemAt(i);
+            mActiveTracks.remove(track);
+            if (track->mainBuffer() != mMixBuffer) {
+                chain = getEffectChain_l(track->sessionId());
+                if (chain != 0) {
+                    LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
+                    chain->stopTrack();
+                }
+            }
+            if (track->isTerminated()) {
+                mTracks.remove(track);
+                deleteTrackName_l(track->mName);
+            }
+        }
+    }
+
+    // mix buffer must be cleared if all tracks are connected to an
+    // effect chain as in this case the mixer will not write to
+    // mix buffer and track effects will accumulate into it
+    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
+        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
+    }
+
+    return mixerStatus;
+}
+
+void AudioFlinger::MixerThread::invalidateTracks(int streamType)
+{
+    LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", this,  streamType, mTracks.size());
+    Mutex::Autolock _l(mLock);
+    size_t size = mTracks.size();
+    for (size_t i = 0; i < size; i++) {
+        sp<Track> t = mTracks[i];
+        if (t->type() == streamType) {
+            t->mCblk->lock.lock();
+            t->mCblk->flags |= CBLK_INVALID_ON;
+            t->mCblk->cv.signal();
+            t->mCblk->lock.unlock();
+        }
+    }
+}
+
+
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::MixerThread::getTrackName_l()
+{
+    return mAudioMixer->getTrackName();
+}
+
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::MixerThread::deleteTrackName_l(int name)
+{
+    LOGV("remove track (%d) and delete from mixer", name);
+    mAudioMixer->deleteTrackName(name);
+}
+
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::MixerThread::checkForNewParameters_l()
+{
+    bool reconfig = false;
+
+    while (!mNewParameters.isEmpty()) {
+        status_t status = NO_ERROR;
+        String8 keyValuePair = mNewParameters[0];
+        AudioParameter param = AudioParameter(keyValuePair);
+        int value;
+
+        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+            if (value != AudioSystem::PCM_16_BIT) {
+                status = BAD_VALUE;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+            if (value != AudioSystem::CHANNEL_OUT_STEREO) {
+                status = BAD_VALUE;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be garantied
+            // if frame count is changed after track creation
+            if (!mTracks.isEmpty()) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
+            // forward device change to effects that have requested to be
+            // aware of attached audio device.
+            mDevice = (uint32_t)value;
+            for (size_t i = 0; i < mEffectChains.size(); i++) {
+                mEffectChains[i]->setDevice(mDevice);
+            }
+        }
+
+        if (status == NO_ERROR) {
+            status = mOutput->setParameters(keyValuePair);
+            if (!mStandby && status == INVALID_OPERATION) {
+               mOutput->standby();
+               mStandby = true;
+               mBytesWritten = 0;
+               status = mOutput->setParameters(keyValuePair);
+            }
+            if (status == NO_ERROR && reconfig) {
+                delete mAudioMixer;
+                readOutputParameters();
+                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
+                for (size_t i = 0; i < mTracks.size() ; i++) {
+                    int name = getTrackName_l();
+                    if (name < 0) break;
+                    mTracks[i]->mName = name;
+                    // limit track sample rate to 2 x new output sample rate
+                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
+                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
+                    }
+                }
+                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
+            }
+        }
+
+        mNewParameters.removeAt(0);
+
+        mParamStatus = status;
+        mParamCond.signal();
+        mWaitWorkCV.wait(mLock);
+    }
+    return reconfig;
+}
+
+status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    PlaybackThread::dumpInternals(fd, args);
+
+    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+uint32_t AudioFlinger::MixerThread::activeSleepTimeUs()
+{
+    return (uint32_t)(mOutput->latency() * 1000) / 2;
+}
+
+uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
+{
+    return (uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000;
+}
+
+// ----------------------------------------------------------------------------
+AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
+    :   PlaybackThread(audioFlinger, output, id, device)
+{
+    mType = PlaybackThread::DIRECT;
+}
+
+AudioFlinger::DirectOutputThread::~DirectOutputThread()
+{
+}
+
+
+static inline int16_t clamp16(int32_t sample)
+{
+    if ((sample>>15) ^ (sample>>31))
+        sample = 0x7FFF ^ (sample>>31);
+    return sample;
+}
+
+static inline
+int32_t mul(int16_t in, int16_t v)
+{
+#if defined(__arm__) && !defined(__thumb__)
+    int32_t out;
+    asm( "smulbb %[out], %[in], %[v] \n"
+         : [out]"=r"(out)
+         : [in]"%r"(in), [v]"r"(v)
+         : );
+    return out;
+#else
+    return in * int32_t(v);
+#endif
+}
+
+void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
+{
+    // Do not apply volume on compressed audio
+    if (!AudioSystem::isLinearPCM(mFormat)) {
+        return;
+    }
+
+    // convert to signed 16 bit before volume calculation
+    if (mFormat == AudioSystem::PCM_8_BIT) {
+        size_t count = mFrameCount * mChannelCount;
+        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
+        int16_t *dst = mMixBuffer + count-1;
+        while(count--) {
+            *dst-- = (int16_t)(*src--^0x80) << 8;
+        }
+    }
+
+    size_t frameCount = mFrameCount;
+    int16_t *out = mMixBuffer;
+    if (ramp) {
+        if (mChannelCount == 1) {
+            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
+            int32_t vlInc = d / (int32_t)frameCount;
+            int32_t vl = ((int32_t)mLeftVolShort << 16);
+            do {
+                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
+                out++;
+                vl += vlInc;
+            } while (--frameCount);
+
+        } else {
+            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
+            int32_t vlInc = d / (int32_t)frameCount;
+            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
+            int32_t vrInc = d / (int32_t)frameCount;
+            int32_t vl = ((int32_t)mLeftVolShort << 16);
+            int32_t vr = ((int32_t)mRightVolShort << 16);
+            do {
+                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
+                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
+                out += 2;
+                vl += vlInc;
+                vr += vrInc;
+            } while (--frameCount);
+        }
+    } else {
+        if (mChannelCount == 1) {
+            do {
+                out[0] = clamp16(mul(out[0], leftVol) >> 12);
+                out++;
+            } while (--frameCount);
+        } else {
+            do {
+                out[0] = clamp16(mul(out[0], leftVol) >> 12);
+                out[1] = clamp16(mul(out[1], rightVol) >> 12);
+                out += 2;
+            } while (--frameCount);
+        }
+    }
+
+    // convert back to unsigned 8 bit after volume calculation
+    if (mFormat == AudioSystem::PCM_8_BIT) {
+        size_t count = mFrameCount * mChannelCount;
+        int16_t *src = mMixBuffer;
+        uint8_t *dst = (uint8_t *)mMixBuffer;
+        while(count--) {
+            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
+        }
+    }
+
+    mLeftVolShort = leftVol;
+    mRightVolShort = rightVol;
+}
+
+bool AudioFlinger::DirectOutputThread::threadLoop()
+{
+    uint32_t mixerStatus = MIXER_IDLE;
+    sp<Track> trackToRemove;
+    sp<Track> activeTrack;
+    nsecs_t standbyTime = systemTime();
+    int8_t *curBuf;
+    size_t mixBufferSize = mFrameCount*mFrameSize;
+    uint32_t activeSleepTime = activeSleepTimeUs();
+    uint32_t idleSleepTime = idleSleepTimeUs();
+    uint32_t sleepTime = idleSleepTime;
+    // use shorter standby delay as on normal output to release
+    // hardware resources as soon as possible
+    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
+
+
+    while (!exitPending())
+    {
+        bool rampVolume;
+        uint16_t leftVol;
+        uint16_t rightVol;
+        Vector< sp<EffectChain> > effectChains;
+
+        processConfigEvents();
+
+        mixerStatus = MIXER_IDLE;
+
+        { // scope for the mLock
+
+            Mutex::Autolock _l(mLock);
+
+            if (checkForNewParameters_l()) {
+                mixBufferSize = mFrameCount*mFrameSize;
+                activeSleepTime = activeSleepTimeUs();
+                idleSleepTime = idleSleepTimeUs();
+                standbyDelay = microseconds(activeSleepTime*2);
+            }
+
+            // put audio hardware into standby after short delay
+            if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
+                        mSuspended) {
+                // wait until we have something to do...
+                if (!mStandby) {
+                    LOGV("Audio hardware entering standby, mixer %p\n", this);
+                    mOutput->standby();
+                    mStandby = true;
+                    mBytesWritten = 0;
+                }
+
+                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
+                    // we're about to wait, flush the binder command buffer
+                    IPCThreadState::self()->flushCommands();
+
+                    if (exitPending()) break;
+
+                    LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
+                    mWaitWorkCV.wait(mLock);
+                    LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
+
+                    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() + standbyDelay;
+                    sleepTime = idleSleepTime;
+                    continue;
+                }
+            }
+
+            effectChains = mEffectChains;
+
+            // find out which tracks need to be processed
+            if (mActiveTracks.size() != 0) {
+                sp<Track> t = mActiveTracks[0].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
+                if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
+                        !track->isPaused() && !track->isTerminated())
+                {
+                    //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+                    if (track->mFillingUpStatus == Track::FS_FILLED) {
+                        track->mFillingUpStatus = Track::FS_ACTIVE;
+                        mLeftVolFloat = mRightVolFloat = 0;
+                        mLeftVolShort = mRightVolShort = 0;
+                        if (track->mState == TrackBase::RESUMING) {
+                            track->mState = TrackBase::ACTIVE;
+                            rampVolume = true;
+                        }
+                    } else if (cblk->server != 0) {
+                        // If the track is stopped before the first frame was mixed,
+                        // do not apply ramp
+                        rampVolume = true;
+                    }
+                    // compute volume for this track
+                    float left, right;
+                    if (track->isMuted() || mMasterMute || track->isPausing() ||
+                        mStreamTypes[track->type()].mute) {
+                        left = right = 0;
+                        if (track->isPausing()) {
+                            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 = v_clamped/MAX_GAIN;
+                        v_clamped = v * cblk->volume[1];
+                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+                        right = v_clamped/MAX_GAIN;
+                    }
+
+                    if (left != mLeftVolFloat || right != mRightVolFloat) {
+                        mLeftVolFloat = left;
+                        mRightVolFloat = right;
+
+                        // If audio HAL implements volume control,
+                        // force software volume to nominal value
+                        if (mOutput->setVolume(left, right) == NO_ERROR) {
+                            left = 1.0f;
+                            right = 1.0f;
+                        }
+
+                        // Convert volumes from float to 8.24
+                        uint32_t vl = (uint32_t)(left * (1 << 24));
+                        uint32_t vr = (uint32_t)(right * (1 << 24));
+
+                        // Delegate volume control to effect in track effect chain if needed
+                        // only one effect chain can be present on DirectOutputThread, so if
+                        // there is one, the track is connected to it
+                        if (!effectChains.isEmpty()) {
+                            // Do not ramp volume is volume is controlled by effect
+                            if(effectChains[0]->setVolume(&vl, &vr)) {
+                                rampVolume = false;
+                            }
+                        }
+
+                        // Convert volumes from 8.24 to 4.12 format
+                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
+                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
+                        leftVol = (uint16_t)v_clamped;
+                        v_clamped = (vr + (1 << 11)) >> 12;
+                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
+                        rightVol = (uint16_t)v_clamped;
+                    } else {
+                        leftVol = mLeftVolShort;
+                        rightVol = mRightVolShort;
+                        rampVolume = false;
+                    }
+
+                    // reset retry count
+                    track->mRetryCount = kMaxTrackRetriesDirect;
+                    activeTrack = t;
+                    mixerStatus = MIXER_TRACKS_READY;
+                } 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.
+                        trackToRemove = 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());
+                            trackToRemove = track;
+                        } else {
+                            mixerStatus = MIXER_TRACKS_ENABLED;
+                        }
+                    }
+                }
+            }
+
+            // remove all the tracks that need to be...
+            if (UNLIKELY(trackToRemove != 0)) {
+                mActiveTracks.remove(trackToRemove);
+                if (!effectChains.isEmpty()) {
+                    LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), trackToRemove->sessionId());
+                    effectChains[0]->stopTrack();
+                }
+                if (trackToRemove->isTerminated()) {
+                    mTracks.remove(trackToRemove);
+                    deleteTrackName_l(trackToRemove->mName);
+                }
+            }
+
+            lockEffectChains_l();
+       }
+
+        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+            AudioBufferProvider::Buffer buffer;
+            size_t frameCount = mFrameCount;
+            curBuf = (int8_t *)mMixBuffer;
+            // output audio to hardware
+            while (frameCount) {
+                buffer.frameCount = frameCount;
+                activeTrack->getNextBuffer(&buffer);
+                if (UNLIKELY(buffer.raw == 0)) {
+                    memset(curBuf, 0, frameCount * mFrameSize);
+                    break;
+                }
+                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
+                frameCount -= buffer.frameCount;
+                curBuf += buffer.frameCount * mFrameSize;
+                activeTrack->releaseBuffer(&buffer);
+            }
+            sleepTime = 0;
+            standbyTime = systemTime() + standbyDelay;
+        } else {
+            if (sleepTime == 0) {
+                if (mixerStatus == MIXER_TRACKS_ENABLED) {
+                    sleepTime = activeSleepTime;
+                } else {
+                    sleepTime = idleSleepTime;
+                }
+            } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) {
+                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
+                sleepTime = 0;
+            }
+        }
+
+        if (mSuspended) {
+            sleepTime = idleSleepTime;
+        }
+        // sleepTime == 0 means we must write to audio hardware
+        if (sleepTime == 0) {
+            if (mixerStatus == MIXER_TRACKS_READY) {
+                applyVolume(leftVol, rightVol, rampVolume);
+            }
+            for (size_t i = 0; i < effectChains.size(); i ++) {
+                effectChains[i]->process_l();
+            }
+            unlockEffectChains();
+
+            mLastWriteTime = systemTime();
+            mInWrite = true;
+            mBytesWritten += mixBufferSize;
+            int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
+            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
+            mNumWrites++;
+            mInWrite = false;
+            mStandby = false;
+        } else {
+            unlockEffectChains();
+            usleep(sleepTime);
+        }
+
+        // finally let go of removed track, without the lock held
+        // since we can't guarantee the destructors won't acquire that
+        // same lock.
+        trackToRemove.clear();
+        activeTrack.clear();
+
+        // Effect chains will be actually deleted here if they were removed from
+        // mEffectChains list during mixing or effects processing
+        effectChains.clear();
+    }
+
+    if (!mStandby) {
+        mOutput->standby();
+    }
+
+    LOGV("DirectOutputThread %p exiting", this);
+    return false;
+}
+
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::DirectOutputThread::getTrackName_l()
+{
+    return 0;
+}
+
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
+{
+}
+
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
+{
+    bool reconfig = false;
+
+    while (!mNewParameters.isEmpty()) {
+        status_t status = NO_ERROR;
+        String8 keyValuePair = mNewParameters[0];
+        AudioParameter param = AudioParameter(keyValuePair);
+        int value;
+
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be garantied
+            // if frame count is changed after track creation
+            if (!mTracks.isEmpty()) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (status == NO_ERROR) {
+            status = mOutput->setParameters(keyValuePair);
+            if (!mStandby && status == INVALID_OPERATION) {
+               mOutput->standby();
+               mStandby = true;
+               mBytesWritten = 0;
+               status = mOutput->setParameters(keyValuePair);
+            }
+            if (status == NO_ERROR && reconfig) {
+                readOutputParameters();
+                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
+            }
+        }
+
+        mNewParameters.removeAt(0);
+
+        mParamStatus = status;
+        mParamCond.signal();
+        mWaitWorkCV.wait(mLock);
+    }
+    return reconfig;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
+{
+    uint32_t time;
+    if (AudioSystem::isLinearPCM(mFormat)) {
+        time = (uint32_t)(mOutput->latency() * 1000) / 2;
+    } else {
+        time = 10000;
+    }
+    return time;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
+{
+    uint32_t time;
+    if (AudioSystem::isLinearPCM(mFormat)) {
+        time = (uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000;
+    } else {
+        time = 10000;
+    }
+    return time;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
+    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
+{
+    mType = PlaybackThread::DUPLICATING;
+    addOutputTrack(mainThread);
+}
+
+AudioFlinger::DuplicatingThread::~DuplicatingThread()
+{
+    for (size_t i = 0; i < mOutputTracks.size(); i++) {
+        mOutputTracks[i]->destroy();
+    }
+    mOutputTracks.clear();
+}
+
+bool AudioFlinger::DuplicatingThread::threadLoop()
+{
+    Vector< sp<Track> > tracksToRemove;
+    uint32_t mixerStatus = MIXER_IDLE;
+    nsecs_t standbyTime = systemTime();
+    size_t mixBufferSize = mFrameCount*mFrameSize;
+    SortedVector< sp<OutputTrack> > outputTracks;
+    uint32_t writeFrames = 0;
+    uint32_t activeSleepTime = activeSleepTimeUs();
+    uint32_t idleSleepTime = idleSleepTimeUs();
+    uint32_t sleepTime = idleSleepTime;
+    Vector< sp<EffectChain> > effectChains;
+
+    while (!exitPending())
+    {
+        processConfigEvents();
+
+        mixerStatus = MIXER_IDLE;
+        { // scope for the mLock
+
+            Mutex::Autolock _l(mLock);
+
+            if (checkForNewParameters_l()) {
+                mixBufferSize = mFrameCount*mFrameSize;
+                updateWaitTime();
+                activeSleepTime = activeSleepTimeUs();
+                idleSleepTime = idleSleepTimeUs();
+            }
+
+            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+
+            for (size_t i = 0; i < mOutputTracks.size(); i++) {
+                outputTracks.add(mOutputTracks[i]);
+            }
+
+            // put audio hardware into standby after short delay
+            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+                         mSuspended) {
+                if (!mStandby) {
+                    for (size_t i = 0; i < outputTracks.size(); i++) {
+                        outputTracks[i]->stop();
+                    }
+                    mStandby = true;
+                    mBytesWritten = 0;
+                }
+
+                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
+                    // we're about to wait, flush the binder command buffer
+                    IPCThreadState::self()->flushCommands();
+                    outputTracks.clear();
+
+                    if (exitPending()) break;
+
+                    LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
+                    mWaitWorkCV.wait(mLock);
+                    LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
+                    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;
+                    sleepTime = idleSleepTime;
+                    continue;
+                }
+            }
+
+            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
+
+            // prevent any changes in effect chain list and in each effect chain
+            // during mixing and effect process as the audio buffers could be deleted
+            // or modified if an effect is created or deleted
+            effectChains = mEffectChains;
+            lockEffectChains_l();
+        }
+
+        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+            // mix buffers...
+            if (outputsReady(outputTracks)) {
+                mAudioMixer->process();
+            } else {
+                memset(mMixBuffer, 0, mixBufferSize);
+            }
+            sleepTime = 0;
+            writeFrames = mFrameCount;
+        } else {
+            if (sleepTime == 0) {
+                if (mixerStatus == MIXER_TRACKS_ENABLED) {
+                    sleepTime = activeSleepTime;
+                } else {
+                    sleepTime = idleSleepTime;
+                }
+            } else if (mBytesWritten != 0) {
+                // flush remaining overflow buffers in output tracks
+                for (size_t i = 0; i < outputTracks.size(); i++) {
+                    if (outputTracks[i]->isActive()) {
+                        sleepTime = 0;
+                        writeFrames = 0;
+                        memset(mMixBuffer, 0, mixBufferSize);
+                        break;
+                    }
+                }
+            }
+        }
+
+        if (mSuspended) {
+            sleepTime = idleSleepTime;
+        }
+        // sleepTime == 0 means we must write to audio hardware
+        if (sleepTime == 0) {
+            for (size_t i = 0; i < effectChains.size(); i ++) {
+                effectChains[i]->process_l();
+            }
+            // enable changes in effect chain
+            unlockEffectChains();
+
+            standbyTime = systemTime() + kStandbyTimeInNsecs;
+            for (size_t i = 0; i < outputTracks.size(); i++) {
+                outputTracks[i]->write(mMixBuffer, writeFrames);
+            }
+            mStandby = false;
+            mBytesWritten += mixBufferSize;
+        } else {
+            // enable changes in effect chain
+            unlockEffectChains();
+            usleep(sleepTime);
+        }
+
+        // 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();
+        outputTracks.clear();
+
+        // Effect chains will be actually deleted here if they were removed from
+        // mEffectChains list during mixing or effects processing
+        effectChains.clear();
+    }
+
+    return false;
+}
+
+void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
+{
+    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
+    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
+                                            this,
+                                            mSampleRate,
+                                            mFormat,
+                                            mChannelCount,
+                                            frameCount);
+    if (outputTrack->cblk() != NULL) {
+        thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f);
+        mOutputTracks.add(outputTrack);
+        LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
+        updateWaitTime();
+    }
+}
+
+void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
+{
+    Mutex::Autolock _l(mLock);
+    for (size_t i = 0; i < mOutputTracks.size(); i++) {
+        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
+            mOutputTracks[i]->destroy();
+            mOutputTracks.removeAt(i);
+            updateWaitTime();
+            return;
+        }
+    }
+    LOGV("removeOutputTrack(): unkonwn thread: %p", thread);
+}
+
+void AudioFlinger::DuplicatingThread::updateWaitTime()
+{
+    mWaitTimeMs = UINT_MAX;
+    for (size_t i = 0; i < mOutputTracks.size(); i++) {
+        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
+        if (strong != NULL) {
+            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
+            if (waitTimeMs < mWaitTimeMs) {
+                mWaitTimeMs = waitTimeMs;
+            }
+        }
+    }
+}
+
+
+bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
+{
+    for (size_t i = 0; i < outputTracks.size(); i++) {
+        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
+        if (thread == 0) {
+            LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
+            return false;
+        }
+        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        if (playbackThread->standby() && !playbackThread->isSuspended()) {
+            LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
+            return false;
+        }
+    }
+    return true;
+}
+
+uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
+{
+    return (mWaitTimeMs * 1000) / 2;
+}
+
+// ----------------------------------------------------------------------------
+
+// TrackBase constructor must be called with AudioFlinger::mLock held
+AudioFlinger::ThreadBase::TrackBase::TrackBase(
+            const wp<ThreadBase>& thread,
+            const sp<Client>& client,
+            uint32_t sampleRate,
+            int format,
+            int channelCount,
+            int frameCount,
+            uint32_t flags,
+            const sp<IMemory>& sharedBuffer,
+            int sessionId)
+    :   RefBase(),
+        mThread(thread),
+        mClient(client),
+        mCblk(0),
+        mFrameCount(0),
+        mState(IDLE),
+        mClientTid(-1),
+        mFormat(format),
+        mFlags(flags & ~SYSTEM_FLAGS_MASK),
+        mSessionId(sessionId)
+{
+    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->channelCount = (uint8_t)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->flags = CBLK_UNDERRUN_ON;
+                } 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->channelCount = (uint8_t)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->flags = CBLK_UNDERRUN_ON;
+           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
+       }
+   }
+}
+
+AudioFlinger::ThreadBase::TrackBase::~TrackBase()
+{
+    if (mCblk) {
+        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
+        if (mClient == NULL) {
+            delete mCblk;
+        }
+    }
+    mCblkMemory.clear();            // and free the shared memory
+    if (mClient != NULL) {
+        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
+        mClient.clear();
+    }
+}
+
+void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    buffer->raw = 0;
+    mFrameCount = buffer->frameCount;
+    step();
+    buffer->frameCount = 0;
+}
+
+bool AudioFlinger::ThreadBase::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::ThreadBase::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::ThreadBase::TrackBase::getCblk() const
+{
+    return mCblkMemory;
+}
+
+int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
+    return (int)mCblk->sampleRate;
+}
+
+int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
+    return (int)mCblk->channelCount;
+}
+
+void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
+    audio_track_cblk_t* cblk = this->cblk();
+    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
+    int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
+
+    // Check validity of returned pointer in case the track control block would have been corrupted.
+    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
+        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
+        LOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
+                server %d, serverBase %d, user %d, userBase %d, channelCount %d",
+                bufferStart, bufferEnd, mBuffer, mBufferEnd,
+                cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount);
+        return 0;
+    }
+
+    return bufferStart;
+}
+
+// ----------------------------------------------------------------------------
+
+// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
+AudioFlinger::PlaybackThread::Track::Track(
+            const wp<ThreadBase>& thread,
+            const sp<Client>& client,
+            int streamType,
+            uint32_t sampleRate,
+            int format,
+            int channelCount,
+            int frameCount,
+            const sp<IMemory>& sharedBuffer,
+            int sessionId)
+    :   TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId),
+    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), mAuxEffectId(0)
+{
+    if (mCblk != NULL) {
+        sp<ThreadBase> baseThread = thread.promote();
+        if (baseThread != 0) {
+            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
+            mName = playbackThread->getTrackName_l();
+            mMainBuffer = playbackThread->mixBuffer();
+        }
+        LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+        if (mName < 0) {
+            LOGE("no more track names available");
+        }
+        mVolume[0] = 1.0f;
+        mVolume[1] = 1.0f;
+        mStreamType = streamType;
+        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
+        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
+        mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
+    }
+}
+
+AudioFlinger::PlaybackThread::Track::~Track()
+{
+    LOGV("PlaybackThread::Track destructor");
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        mState = TERMINATED;
+    }
+}
+
+void AudioFlinger::PlaybackThread::Track::destroy()
+{
+    // NOTE: destroyTrack_l() can remove a strong reference to this Track
+    // by removing it from mTracks vector, so there is a risk that this Tracks's
+    // desctructor is called. As the destructor needs to lock mLock,
+    // we must acquire a strong reference on this Track before locking mLock
+    // here so that the destructor is called only when exiting this function.
+    // On the other hand, as long as Track::destroy() is only called by
+    // TrackHandle destructor, the TrackHandle still holds a strong ref on
+    // this Track with its member mTrack.
+    sp<Track> keep(this);
+    { // scope for mLock
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            if (!isOutputTrack()) {
+                if (mState == ACTIVE || mState == RESUMING) {
+                    AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType);
+                }
+                AudioSystem::releaseOutput(thread->id());
+            }
+            Mutex::Autolock _l(thread->mLock);
+            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            playbackThread->destroyTrack_l(this);
+        }
+    }
+}
+
+void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
+{
+    snprintf(buffer, size, "   %05d %05d %03u %03u %03u %05u   %04u %1d %1d %1d %05u %05u %05u  0x%08x 0x%08x 0x%08x 0x%08x\n",
+            mName - AudioMixer::TRACK0,
+            (mClient == NULL) ? getpid() : mClient->pid(),
+            mStreamType,
+            mFormat,
+            mCblk->channelCount,
+            mSessionId,
+            mFrameCount,
+            mState,
+            mMute,
+            mFillingUpStatus,
+            mCblk->sampleRate,
+            mCblk->volume[0],
+            mCblk->volume[1],
+            mCblk->server,
+            mCblk->user,
+            (int)mMainBuffer,
+            (int)mAuxBuffer);
+}
+
+status_t AudioFlinger::PlaybackThread::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;
+     LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
+     return NOT_ENOUGH_DATA;
+}
+
+bool AudioFlinger::PlaybackThread::Track::isReady() const {
+    if (mFillingUpStatus != FS_FILLING) return true;
+
+    if (mCblk->framesReady() >= mCblk->frameCount ||
+            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
+        mFillingUpStatus = FS_FILLED;
+        mCblk->flags &= ~CBLK_FORCEREADY_MSK;
+        return true;
+    }
+    return false;
+}
+
+status_t AudioFlinger::PlaybackThread::Track::start()
+{
+    status_t status = NO_ERROR;
+    LOGV("start(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        int state = mState;
+        // here the track could be either new, or restarted
+        // in both cases "unstop" the track
+        if (mState == PAUSED) {
+            mState = TrackBase::RESUMING;
+            LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
+        } else {
+            mState = TrackBase::ACTIVE;
+            LOGV("? => ACTIVE (%d) on thread %p", mName, this);
+        }
+
+        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
+            thread->mLock.unlock();
+            status = AudioSystem::startOutput(thread->id(), (AudioSystem::stream_type)mStreamType);
+            thread->mLock.lock();
+        }
+        if (status == NO_ERROR) {
+            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            playbackThread->addTrack_l(this);
+        } else {
+            mState = state;
+        }
+    } else {
+        status = BAD_VALUE;
+    }
+    return status;
+}
+
+void AudioFlinger::PlaybackThread::Track::stop()
+{
+    LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        int state = mState;
+        if (mState > STOPPED) {
+            mState = STOPPED;
+            // If the track is not active (PAUSED and buffers full), flush buffers
+            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+                reset();
+            }
+            LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
+        }
+        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
+            thread->mLock.unlock();
+            AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType);
+            thread->mLock.lock();
+        }
+    }
+}
+
+void AudioFlinger::PlaybackThread::Track::pause()
+{
+    LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        if (mState == ACTIVE || mState == RESUMING) {
+            mState = PAUSING;
+            LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
+            if (!isOutputTrack()) {
+                thread->mLock.unlock();
+                AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType);
+                thread->mLock.lock();
+            }
+        }
+    }
+}
+
+void AudioFlinger::PlaybackThread::Track::flush()
+{
+    LOGV("flush(%d)", mName);
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        Mutex::Autolock _l(thread->mLock);
+        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
+            return;
+        }
+        // No point remaining in PAUSED state after a flush => go to
+        // STOPPED state
+        mState = STOPPED;
+
+        mCblk->lock.lock();
+        // 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();
+        mCblk->lock.unlock();
+    }
+}
+
+void AudioFlinger::PlaybackThread::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->flags |= CBLK_UNDERRUN_ON;
+        mCblk->flags &= ~CBLK_FORCEREADY_MSK;
+        mFillingUpStatus = FS_FILLING;
+        mResetDone = true;
+    }
+}
+
+void AudioFlinger::PlaybackThread::Track::mute(bool muted)
+{
+    mMute = muted;
+}
+
+void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
+{
+    mVolume[0] = left;
+    mVolume[1] = right;
+}
+
+status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
+{
+    status_t status = DEAD_OBJECT;
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+       status = playbackThread->attachAuxEffect(this, EffectId);
+    }
+    return status;
+}
+
+void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
+{
+    mAuxEffectId = EffectId;
+    mAuxBuffer = buffer;
+}
+
+// ----------------------------------------------------------------------------
+
+// RecordTrack constructor must be called with AudioFlinger::mLock held
+AudioFlinger::RecordThread::RecordTrack::RecordTrack(
+            const wp<ThreadBase>& thread,
+            const sp<Client>& client,
+            uint32_t sampleRate,
+            int format,
+            int channelCount,
+            int frameCount,
+            uint32_t flags,
+            int sessionId)
+    :   TrackBase(thread, client, sampleRate, format,
+                  channelCount, frameCount, flags, 0, sessionId),
+        mOverflow(false)
+{
+    if (mCblk != NULL) {
+       LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+       if (format == AudioSystem::PCM_16_BIT) {
+           mCblk->frameSize = channelCount * sizeof(int16_t);
+       } else if (format == AudioSystem::PCM_8_BIT) {
+           mCblk->frameSize = channelCount * sizeof(int8_t);
+       } else {
+           mCblk->frameSize = sizeof(int8_t);
+       }
+    }
+}
+
+AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
+{
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        AudioSystem::releaseInput(thread->id());
+    }
+}
+
+status_t AudioFlinger::RecordThread::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::RecordThread::RecordTrack::start()
+{
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        RecordThread *recordThread = (RecordThread *)thread.get();
+        return recordThread->start(this);
+    } else {
+        return BAD_VALUE;
+    }
+}
+
+void AudioFlinger::RecordThread::RecordTrack::stop()
+{
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread != 0) {
+        RecordThread *recordThread = (RecordThread *)thread.get();
+        recordThread->stop(this);
+        TrackBase::reset();
+        // Force overerrun condition to avoid false overrun callback until first data is
+        // read from buffer
+        mCblk->flags |= CBLK_UNDERRUN_ON;
+    }
+}
+
+void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
+{
+    snprintf(buffer, size, "   %05d %03u %03u %05d   %04u %01d %05u  %08x %08x\n",
+            (mClient == NULL) ? getpid() : mClient->pid(),
+            mFormat,
+            mCblk->channelCount,
+            mSessionId,
+            mFrameCount,
+            mState,
+            mCblk->sampleRate,
+            mCblk->server,
+            mCblk->user);
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
+            const wp<ThreadBase>& thread,
+            DuplicatingThread *sourceThread,
+            uint32_t sampleRate,
+            int format,
+            int channelCount,
+            int frameCount)
+    :   Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0),
+    mActive(false), mSourceThread(sourceThread)
+{
+
+    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
+    if (mCblk != NULL) {
+        mCblk->flags |= CBLK_DIRECTION_OUT;
+        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+        mCblk->volume[0] = mCblk->volume[1] = 0x1000;
+        mOutBuffer.frameCount = 0;
+        playbackThread->mTracks.add(this);
+        LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p",
+                mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd);
+    } else {
+        LOGW("Error creating output track on thread %p", playbackThread);
+    }
+}
+
+AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
+{
+    clearBufferQueue();
+}
+
+status_t AudioFlinger::PlaybackThread::OutputTrack::start()
+{
+    status_t status = Track::start();
+    if (status != NO_ERROR) {
+        return status;
+    }
+
+    mActive = true;
+    mRetryCount = 127;
+    return status;
+}
+
+void AudioFlinger::PlaybackThread::OutputTrack::stop()
+{
+    Track::stop();
+    clearBufferQueue();
+    mOutBuffer.frameCount = 0;
+    mActive = false;
+}
+
+bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
+{
+    Buffer *pInBuffer;
+    Buffer inBuffer;
+    uint32_t channelCount = mCblk->channelCount;
+    bool outputBufferFull = false;
+    inBuffer.frameCount = frames;
+    inBuffer.i16 = data;
+
+    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
+
+    if (!mActive && frames != 0) {
+        start();
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            MixerThread *mixerThread = (MixerThread *)thread.get();
+            if (mCblk->frameCount > frames){
+                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
+                    uint32_t startFrames = (mCblk->frameCount - frames);
+                    pInBuffer = new Buffer;
+                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
+                    pInBuffer->frameCount = startFrames;
+                    pInBuffer->i16 = pInBuffer->mBuffer;
+                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
+                    mBufferQueue.add(pInBuffer);
+                } else {
+                    LOGW ("OutputTrack::write() %p no more buffers in queue", this);
+                }
+            }
+        }
+    }
+
+    while (waitTimeLeftMs) {
+        // 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;
+            nsecs_t startTime = systemTime();
+            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
+                LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
+                outputBufferFull = true;
+                break;
+            }
+            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
+            if (waitTimeLeftMs >= waitTimeMs) {
+                waitTimeLeftMs -= waitTimeMs;
+            } else {
+                waitTimeLeftMs = 0;
+            }
+        }
+
+        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
+        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
+        mCblk->stepUser(outFrames);
+        pInBuffer->frameCount -= outFrames;
+        pInBuffer->i16 += outFrames * channelCount;
+        mOutBuffer.frameCount -= outFrames;
+        mOutBuffer.i16 += outFrames * channelCount;
+
+        if (pInBuffer->frameCount == 0) {
+            if (mBufferQueue.size()) {
+                mBufferQueue.removeAt(0);
+                delete [] pInBuffer->mBuffer;
+                delete pInBuffer;
+                LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
+            } else {
+                break;
+            }
+        }
+    }
+
+    // If we could not write all frames, allocate a buffer and queue it for next time.
+    if (inBuffer.frameCount) {
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0 && !thread->standby()) {
+            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
+                pInBuffer = new Buffer;
+                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
+                pInBuffer->frameCount = inBuffer.frameCount;
+                pInBuffer->i16 = pInBuffer->mBuffer;
+                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
+                mBufferQueue.add(pInBuffer);
+                LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
+            } else {
+                LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
+            }
+        }
+    }
+
+    // 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) {
+        if (mCblk->user < mCblk->frameCount) {
+            frames = mCblk->frameCount - mCblk->user;
+            pInBuffer = new Buffer;
+            pInBuffer->mBuffer = new int16_t[frames * channelCount];
+            pInBuffer->frameCount = frames;
+            pInBuffer->i16 = pInBuffer->mBuffer;
+            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
+            mBufferQueue.add(pInBuffer);
+        } else if (mActive) {
+            stop();
+        }
+    }
+
+    return outputBufferFull;
+}
+
+status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
+{
+    int active;
+    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) {
+        Mutex::Autolock _l(cblk->lock);
+        goto start_loop_here;
+        while (framesAvail == 0) {
+            active = mActive;
+            if (UNLIKELY(!active)) {
+                LOGV("Not active and NO_MORE_BUFFERS");
+                return AudioTrack::NO_MORE_BUFFERS;
+            }
+            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
+            if (result != NO_ERROR) {
+                return AudioTrack::NO_MORE_BUFFERS;
+            }
+            // read the server count again
+        start_loop_here:
+            framesAvail = cblk->framesAvailable_l();
+        }
+    }
+
+//    if (framesAvail < framesReq) {
+//        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::PlaybackThread::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, "AudioFlinger::Client")),
+        mPid(pid)
+{
+    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
+}
+
+// Client destructor must be called with AudioFlinger::mLock held
+AudioFlinger::Client::~Client()
+{
+    mAudioFlinger->removeClient_l(mPid);
+}
+
+const sp<MemoryDealer>& AudioFlinger::Client::heap() const
+{
+    return mMemoryDealer;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
+                                                     const sp<IAudioFlingerClient>& client,
+                                                     pid_t pid)
+    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
+{
+}
+
+AudioFlinger::NotificationClient::~NotificationClient()
+{
+    mClient.clear();
+}
+
+void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
+{
+    sp<NotificationClient> keep(this);
+    {
+        mAudioFlinger->removeNotificationClient(mPid);
+    }
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::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::attachAuxEffect(int EffectId)
+{
+    return mTrack->attachAuxEffect(EffectId);
+}
+
+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 input,
+        uint32_t sampleRate,
+        int format,
+        int channelCount,
+        int frameCount,
+        uint32_t flags,
+        int *sessionId,
+        status_t *status)
+{
+    sp<RecordThread::RecordTrack> recordTrack;
+    sp<RecordHandle> recordHandle;
+    sp<Client> client;
+    wp<Client> wclient;
+    status_t lStatus;
+    RecordThread *thread;
+    size_t inFrameCount;
+    int lSessionId;
+
+    // check calling permissions
+    if (!recordingAllowed()) {
+        lStatus = PERMISSION_DENIED;
+        goto Exit;
+    }
+
+    // add client to list
+    { // scope for mLock
+        Mutex::Autolock _l(mLock);
+        thread = checkRecordThread_l(input);
+        if (thread == NULL) {
+            lStatus = BAD_VALUE;
+            goto Exit;
+        }
+
+        wclient = mClients.valueFor(pid);
+        if (wclient != NULL) {
+            client = wclient.promote();
+        } else {
+            client = new Client(this, pid);
+            mClients.add(pid, client);
+        }
+
+        // If no audio session id is provided, create one here
+        if (sessionId != NULL && *sessionId != 0) {
+            lSessionId = *sessionId;
+        } else {
+            lSessionId = nextUniqueId();
+            if (sessionId != NULL) {
+                *sessionId = lSessionId;
+            }
+        }
+        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
+        recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate,
+                                                   format, channelCount, frameCount, flags, lSessionId);
+    }
+    if (recordTrack->getCblk() == NULL) {
+        // remove local strong reference to Client before deleting the RecordTrack so that the Client
+        // destructor is called by the TrackBase destructor with mLock held
+        client.clear();
+        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;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::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::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) :
+    ThreadBase(audioFlinger, id),
+    mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
+{
+    mReqChannelCount = AudioSystem::popCount(channels);
+    mReqSampleRate = sampleRate;
+    readInputParameters();
+}
+
+
+AudioFlinger::RecordThread::~RecordThread()
+{
+    delete[] mRsmpInBuffer;
+    if (mResampler != 0) {
+        delete mResampler;
+        delete[] mRsmpOutBuffer;
+    }
+}
+
+void AudioFlinger::RecordThread::onFirstRef()
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+
+    snprintf(buffer, SIZE, "Record Thread %p", this);
+
+    run(buffer, PRIORITY_URGENT_AUDIO);
+}
+
+bool AudioFlinger::RecordThread::threadLoop()
+{
+    AudioBufferProvider::Buffer buffer;
+    sp<RecordTrack> activeTrack;
+
+    // start recording
+    while (!exitPending()) {
+
+        processConfigEvents();
+
+        { // scope for mLock
+            Mutex::Autolock _l(mLock);
+            checkForNewParameters_l();
+            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
+                if (!mStandby) {
+                    mInput->standby();
+                    mStandby = true;
+                }
+
+                if (exitPending()) break;
+
+                LOGV("RecordThread: loop stopping");
+                // go to sleep
+                mWaitWorkCV.wait(mLock);
+                LOGV("RecordThread: loop starting");
+                continue;
+            }
+            if (mActiveTrack != 0) {
+                if (mActiveTrack->mState == TrackBase::PAUSING) {
+                    if (!mStandby) {
+                        mInput->standby();
+                        mStandby = true;
+                    }
+                    mActiveTrack.clear();
+                    mStartStopCond.broadcast();
+                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
+                    if (mReqChannelCount != mActiveTrack->channelCount()) {
+                        mActiveTrack.clear();
+                        mStartStopCond.broadcast();
+                    } else if (mBytesRead != 0) {
+                        // record start succeeds only if first read from audio input
+                        // succeeds
+                        if (mBytesRead > 0) {
+                            mActiveTrack->mState = TrackBase::ACTIVE;
+                        } else {
+                            mActiveTrack.clear();
+                        }
+                        mStartStopCond.broadcast();
+                    }
+                    mStandby = false;
+                }
+            }
+        }
+
+        if (mActiveTrack != 0) {
+            if (mActiveTrack->mState != TrackBase::ACTIVE &&
+                mActiveTrack->mState != TrackBase::RESUMING) {
+                usleep(5000);
+                continue;
+            }
+            buffer.frameCount = mFrameCount;
+            if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+                size_t framesOut = buffer.frameCount;
+                if (mResampler == 0) {
+                    // no resampling
+                    while (framesOut) {
+                        size_t framesIn = mFrameCount - mRsmpInIndex;
+                        if (framesIn) {
+                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
+                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
+                            if (framesIn > framesOut)
+                                framesIn = framesOut;
+                            mRsmpInIndex += framesIn;
+                            framesOut -= framesIn;
+                            if ((int)mChannelCount == mReqChannelCount ||
+                                mFormat != AudioSystem::PCM_16_BIT) {
+                                memcpy(dst, src, framesIn * mFrameSize);
+                            } else {
+                                int16_t *src16 = (int16_t *)src;
+                                int16_t *dst16 = (int16_t *)dst;
+                                if (mChannelCount == 1) {
+                                    while (framesIn--) {
+                                        *dst16++ = *src16;
+                                        *dst16++ = *src16++;
+                                    }
+                                } else {
+                                    while (framesIn--) {
+                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
+                                        src16 += 2;
+                                    }
+                                }
+                            }
+                        }
+                        if (framesOut && mFrameCount == mRsmpInIndex) {
+                            if (framesOut == mFrameCount &&
+                                ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) {
+                                mBytesRead = mInput->read(buffer.raw, mInputBytes);
+                                framesOut = 0;
+                            } else {
+                                mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
+                                mRsmpInIndex = 0;
+                            }
+                            if (mBytesRead < 0) {
+                                LOGE("Error reading audio input");
+                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
+                                    // Force input into standby so that it tries to
+                                    // recover at next read attempt
+                                    mInput->standby();
+                                    usleep(5000);
+                                }
+                                mRsmpInIndex = mFrameCount;
+                                framesOut = 0;
+                                buffer.frameCount = 0;
+                            }
+                        }
+                    }
+                } else {
+                    // resampling
+
+                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
+                    // alter output frame count as if we were expecting stereo samples
+                    if (mChannelCount == 1 && mReqChannelCount == 1) {
+                        framesOut >>= 1;
+                    }
+                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
+                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
+                    // are 32 bit aligned which should be always true.
+                    if (mChannelCount == 2 && mReqChannelCount == 1) {
+                        AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
+                        // the resampler always outputs stereo samples: do post stereo to mono conversion
+                        int16_t *src = (int16_t *)mRsmpOutBuffer;
+                        int16_t *dst = buffer.i16;
+                        while (framesOut--) {
+                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
+                            src += 2;
+                        }
+                    } else {
+                        AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+                    }
+
+                }
+                mActiveTrack->releaseBuffer(&buffer);
+                mActiveTrack->overflow();
+            }
+            // client isn't retrieving buffers fast enough
+            else {
+                if (!mActiveTrack->setOverflow())
+                    LOGW("RecordThread: 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 (!mStandby) {
+        mInput->standby();
+    }
+    mActiveTrack.clear();
+
+    mStartStopCond.broadcast();
+
+    LOGV("RecordThread %p exiting", this);
+    return false;
+}
+
+status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
+{
+    LOGV("RecordThread::start");
+    sp <ThreadBase> strongMe = this;
+    status_t status = NO_ERROR;
+    {
+        AutoMutex lock(&mLock);
+        if (mActiveTrack != 0) {
+            if (recordTrack != mActiveTrack.get()) {
+                status = -EBUSY;
+            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
+                mActiveTrack->mState = TrackBase::ACTIVE;
+            }
+            return status;
+        }
+
+        recordTrack->mState = TrackBase::IDLE;
+        mActiveTrack = recordTrack;
+        mLock.unlock();
+        status_t status = AudioSystem::startInput(mId);
+        mLock.lock();
+        if (status != NO_ERROR) {
+            mActiveTrack.clear();
+            return status;
+        }
+        mActiveTrack->mState = TrackBase::RESUMING;
+        mRsmpInIndex = mFrameCount;
+        mBytesRead = 0;
+        // signal thread to start
+        LOGV("Signal record thread");
+        mWaitWorkCV.signal();
+        // do not wait for mStartStopCond if exiting
+        if (mExiting) {
+            mActiveTrack.clear();
+            status = INVALID_OPERATION;
+            goto startError;
+        }
+        mStartStopCond.wait(mLock);
+        if (mActiveTrack == 0) {
+            LOGV("Record failed to start");
+            status = BAD_VALUE;
+            goto startError;
+        }
+        LOGV("Record started OK");
+        return status;
+    }
+startError:
+    AudioSystem::stopInput(mId);
+    return status;
+}
+
+void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
+    LOGV("RecordThread::stop");
+    sp <ThreadBase> strongMe = this;
+    {
+        AutoMutex lock(&mLock);
+        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
+            mActiveTrack->mState = TrackBase::PAUSING;
+            // do not wait for mStartStopCond if exiting
+            if (mExiting) {
+                return;
+            }
+            mStartStopCond.wait(mLock);
+            // if we have been restarted, recordTrack == mActiveTrack.get() here
+            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
+                mLock.unlock();
+                AudioSystem::stopInput(mId);
+                mLock.lock();
+                LOGV("Record stopped OK");
+            }
+        }
+    }
+}
+
+status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    pid_t pid = 0;
+
+    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
+    result.append(buffer);
+
+    if (mActiveTrack != 0) {
+        result.append("Active Track:\n");
+        result.append("   Clien Fmt Chn Session Buf  S SRate  Serv     User\n");
+        mActiveTrack->dump(buffer, SIZE);
+        result.append(buffer);
+
+        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
+        result.append(buffer);
+        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
+        result.append(buffer);
+        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
+        result.append(buffer);
+        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
+        result.append(buffer);
+        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
+        result.append(buffer);
+
+
+    } else {
+        result.append("No record client\n");
+    }
+    write(fd, result.string(), result.size());
+
+    dumpBase(fd, args);
+
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    size_t framesReq = buffer->frameCount;
+    size_t framesReady = mFrameCount - mRsmpInIndex;
+    int channelCount;
+
+    if (framesReady == 0) {
+        mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
+        if (mBytesRead < 0) {
+            LOGE("RecordThread::getNextBuffer() Error reading audio input");
+            if (mActiveTrack->mState == TrackBase::ACTIVE) {
+                // Force input into standby so that it tries to
+                // recover at next read attempt
+                mInput->standby();
+                usleep(5000);
+            }
+            buffer->raw = 0;
+            buffer->frameCount = 0;
+            return NOT_ENOUGH_DATA;
+        }
+        mRsmpInIndex = 0;
+        framesReady = mFrameCount;
+    }
+
+    if (framesReq > framesReady) {
+        framesReq = framesReady;
+    }
+
+    if (mChannelCount == 1 && mReqChannelCount == 2) {
+        channelCount = 1;
+    } else {
+        channelCount = 2;
+    }
+    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
+    buffer->frameCount = framesReq;
+    return NO_ERROR;
+}
+
+void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+    mRsmpInIndex += buffer->frameCount;
+    buffer->frameCount = 0;
+}
+
+bool AudioFlinger::RecordThread::checkForNewParameters_l()
+{
+    bool reconfig = false;
+
+    while (!mNewParameters.isEmpty()) {
+        status_t status = NO_ERROR;
+        String8 keyValuePair = mNewParameters[0];
+        AudioParameter param = AudioParameter(keyValuePair);
+        int value;
+        int reqFormat = mFormat;
+        int reqSamplingRate = mReqSampleRate;
+        int reqChannelCount = mReqChannelCount;
+
+        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+            reqSamplingRate = value;
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+            reqFormat = value;
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+            reqChannelCount = AudioSystem::popCount(value);
+            reconfig = true;
+        }
+        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+            // do not accept frame count changes if tracks are open as the track buffer
+            // size depends on frame count and correct behavior would not be garantied
+            // if frame count is changed after track creation
+            if (mActiveTrack != 0) {
+                status = INVALID_OPERATION;
+            } else {
+                reconfig = true;
+            }
+        }
+        if (status == NO_ERROR) {
+            status = mInput->setParameters(keyValuePair);
+            if (status == INVALID_OPERATION) {
+               mInput->standby();
+               status = mInput->setParameters(keyValuePair);
+            }
+            if (reconfig) {
+                if (status == BAD_VALUE &&
+                    reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT &&
+                    ((int)mInput->sampleRate() <= 2 * reqSamplingRate) &&
+                    (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) {
+                    status = NO_ERROR;
+                }
+                if (status == NO_ERROR) {
+                    readInputParameters();
+                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
+                }
+            }
+        }
+
+        mNewParameters.removeAt(0);
+
+        mParamStatus = status;
+        mParamCond.signal();
+        mWaitWorkCV.wait(mLock);
+    }
+    return reconfig;
+}
+
+String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
+{
+    return mInput->getParameters(keys);
+}
+
+void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
+    AudioSystem::OutputDescriptor desc;
+    void *param2 = 0;
+
+    switch (event) {
+    case AudioSystem::INPUT_OPENED:
+    case AudioSystem::INPUT_CONFIG_CHANGED:
+        desc.channels = mChannels;
+        desc.samplingRate = mSampleRate;
+        desc.format = mFormat;
+        desc.frameCount = mFrameCount;
+        desc.latency = 0;
+        param2 = &desc;
+        break;
+
+    case AudioSystem::INPUT_CLOSED:
+    default:
+        break;
+    }
+    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
+}
+
+void AudioFlinger::RecordThread::readInputParameters()
+{
+    if (mRsmpInBuffer) delete mRsmpInBuffer;
+    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
+    if (mResampler) delete mResampler;
+    mResampler = 0;
+
+    mSampleRate = mInput->sampleRate();
+    mChannels = mInput->channels();
+    mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);
+    mFormat = mInput->format();
+    mFrameSize = (uint16_t)mInput->frameSize();
+    mInputBytes = mInput->bufferSize();
+    mFrameCount = mInputBytes / mFrameSize;
+    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
+
+    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
+    {
+        int channelCount;
+         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
+         // stereo to mono post process as the resampler always outputs stereo.
+        if (mChannelCount == 1 && mReqChannelCount == 2) {
+            channelCount = 1;
+        } else {
+            channelCount = 2;
+        }
+        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
+        mResampler->setSampleRate(mSampleRate);
+        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
+
+        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
+        if (mChannelCount == 1 && mReqChannelCount == 1) {
+            mFrameCount >>= 1;
+        }
+
+    }
+    mRsmpInIndex = mFrameCount;
+}
+
+unsigned int AudioFlinger::RecordThread::getInputFramesLost()
+{
+    return mInput->getInputFramesLost();
+}
+
+// ----------------------------------------------------------------------------
+
+int AudioFlinger::openOutput(uint32_t *pDevices,
+                                uint32_t *pSamplingRate,
+                                uint32_t *pFormat,
+                                uint32_t *pChannels,
+                                uint32_t *pLatencyMs,
+                                uint32_t flags)
+{
+    status_t status;
+    PlaybackThread *thread = NULL;
+    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
+    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
+    uint32_t format = pFormat ? *pFormat : 0;
+    uint32_t channels = pChannels ? *pChannels : 0;
+    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
+
+    LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
+            pDevices ? *pDevices : 0,
+            samplingRate,
+            format,
+            channels,
+            flags);
+
+    if (pDevices == NULL || *pDevices == 0) {
+        return 0;
+    }
+    Mutex::Autolock _l(mLock);
+
+    AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices,
+                                                             (int *)&format,
+                                                             &channels,
+                                                             &samplingRate,
+                                                             &status);
+    LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
+            output,
+            samplingRate,
+            format,
+            channels,
+            status);
+
+    mHardwareStatus = AUDIO_HW_IDLE;
+    if (output != 0) {
+        int id = nextUniqueId();
+        if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+            (format != AudioSystem::PCM_16_BIT) ||
+            (channels != AudioSystem::CHANNEL_OUT_STEREO)) {
+            thread = new DirectOutputThread(this, output, id, *pDevices);
+            LOGV("openOutput() created direct output: ID %d thread %p", id, thread);
+        } else {
+            thread = new MixerThread(this, output, id, *pDevices);
+            LOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
+
+#ifdef LVMX
+            unsigned bitsPerSample =
+                (format == AudioSystem::PCM_16_BIT) ? 16 :
+                    ((format == AudioSystem::PCM_8_BIT) ? 8 : 0);
+            unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1;
+            int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id());
+
+            LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount);
+            LifeVibes::setDevice(audioOutputType, *pDevices);
+#endif
+
+        }
+        mPlaybackThreads.add(id, thread);
+
+        if (pSamplingRate) *pSamplingRate = samplingRate;
+        if (pFormat) *pFormat = format;
+        if (pChannels) *pChannels = channels;
+        if (pLatencyMs) *pLatencyMs = thread->latency();
+
+        // notify client processes of the new output creation
+        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
+        return id;
+    }
+
+    return 0;
+}
+
+int AudioFlinger::openDuplicateOutput(int output1, int output2)
+{
+    Mutex::Autolock _l(mLock);
+    MixerThread *thread1 = checkMixerThread_l(output1);
+    MixerThread *thread2 = checkMixerThread_l(output2);
+
+    if (thread1 == NULL || thread2 == NULL) {
+        LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
+        return 0;
+    }
+
+    int id = nextUniqueId();
+    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
+    thread->addOutputTrack(thread2);
+    mPlaybackThreads.add(id, thread);
+    // notify client processes of the new output creation
+    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
+    return id;
+}
+
+status_t AudioFlinger::closeOutput(int output)
+{
+    // keep strong reference on the playback thread so that
+    // it is not destroyed while exit() is executed
+    sp <PlaybackThread> thread;
+    {
+        Mutex::Autolock _l(mLock);
+        thread = checkPlaybackThread_l(output);
+        if (thread == NULL) {
+            return BAD_VALUE;
+        }
+
+        LOGV("closeOutput() %d", output);
+
+        if (thread->type() == PlaybackThread::MIXER) {
+            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+                if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) {
+                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
+                    dupThread->removeOutputTrack((MixerThread *)thread.get());
+                }
+            }
+        }
+        void *param2 = 0;
+        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
+        mPlaybackThreads.removeItem(output);
+    }
+    thread->exit();
+
+    if (thread->type() != PlaybackThread::DUPLICATING) {
+        mAudioHardware->closeOutputStream(thread->getOutput());
+    }
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::suspendOutput(int output)
+{
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+
+    if (thread == NULL) {
+        return BAD_VALUE;
+    }
+
+    LOGV("suspendOutput() %d", output);
+    thread->suspend();
+
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::restoreOutput(int output)
+{
+    Mutex::Autolock _l(mLock);
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+
+    if (thread == NULL) {
+        return BAD_VALUE;
+    }
+
+    LOGV("restoreOutput() %d", output);
+
+    thread->restore();
+
+    return NO_ERROR;
+}
+
+int AudioFlinger::openInput(uint32_t *pDevices,
+                                uint32_t *pSamplingRate,
+                                uint32_t *pFormat,
+                                uint32_t *pChannels,
+                                uint32_t acoustics)
+{
+    status_t status;
+    RecordThread *thread = NULL;
+    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
+    uint32_t format = pFormat ? *pFormat : 0;
+    uint32_t channels = pChannels ? *pChannels : 0;
+    uint32_t reqSamplingRate = samplingRate;
+    uint32_t reqFormat = format;
+    uint32_t reqChannels = channels;
+
+    if (pDevices == NULL || *pDevices == 0) {
+        return 0;
+    }
+    Mutex::Autolock _l(mLock);
+
+    AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices,
+                                                             (int *)&format,
+                                                             &channels,
+                                                             &samplingRate,
+                                                             &status,
+                                                             (AudioSystem::audio_in_acoustics)acoustics);
+    LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
+            input,
+            samplingRate,
+            format,
+            channels,
+            acoustics,
+            status);
+
+    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
+    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
+    // or stereo to mono conversions on 16 bit PCM inputs.
+    if (input == 0 && status == BAD_VALUE &&
+        reqFormat == format && format == AudioSystem::PCM_16_BIT &&
+        (samplingRate <= 2 * reqSamplingRate) &&
+        (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) {
+        LOGV("openInput() reopening with proposed sampling rate and channels");
+        input = mAudioHardware->openInputStream(*pDevices,
+                                                 (int *)&format,
+                                                 &channels,
+                                                 &samplingRate,
+                                                 &status,
+                                                 (AudioSystem::audio_in_acoustics)acoustics);
+    }
+
+    if (input != 0) {
+        int id = nextUniqueId();
+         // Start record thread
+        thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id);
+        mRecordThreads.add(id, thread);
+        LOGV("openInput() created record thread: ID %d thread %p", id, thread);
+        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
+        if (pFormat) *pFormat = format;
+        if (pChannels) *pChannels = reqChannels;
+
+        input->standby();
+
+        // notify client processes of the new input creation
+        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
+        return id;
+    }
+
+    return 0;
+}
+
+status_t AudioFlinger::closeInput(int input)
+{
+    // keep strong reference on the record thread so that
+    // it is not destroyed while exit() is executed
+    sp <RecordThread> thread;
+    {
+        Mutex::Autolock _l(mLock);
+        thread = checkRecordThread_l(input);
+        if (thread == NULL) {
+            return BAD_VALUE;
+        }
+
+        LOGV("closeInput() %d", input);
+        void *param2 = 0;
+        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
+        mRecordThreads.removeItem(input);
+    }
+    thread->exit();
+
+    mAudioHardware->closeInputStream(thread->getInput());
+
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
+{
+    Mutex::Autolock _l(mLock);
+    MixerThread *dstThread = checkMixerThread_l(output);
+    if (dstThread == NULL) {
+        LOGW("setStreamOutput() bad output id %d", output);
+        return BAD_VALUE;
+    }
+
+    LOGV("setStreamOutput() stream %d to output %d", stream, output);
+    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
+
+    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+        if (thread != dstThread &&
+            thread->type() != PlaybackThread::DIRECT) {
+            MixerThread *srcThread = (MixerThread *)thread;
+            srcThread->invalidateTracks(stream);
+            }
+        }
+
+    return NO_ERROR;
+}
+
+
+int AudioFlinger::newAudioSessionId()
+{
+    return nextUniqueId();
+}
+
+// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
+{
+    PlaybackThread *thread = NULL;
+    if (mPlaybackThreads.indexOfKey(output) >= 0) {
+        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
+    }
+    return thread;
+}
+
+// checkMixerThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
+{
+    PlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread != NULL) {
+        if (thread->type() == PlaybackThread::DIRECT) {
+            thread = NULL;
+        }
+    }
+    return (MixerThread *)thread;
+}
+
+// checkRecordThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
+{
+    RecordThread *thread = NULL;
+    if (mRecordThreads.indexOfKey(input) >= 0) {
+        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
+    }
+    return thread;
+}
+
+int AudioFlinger::nextUniqueId()
+{
+    return android_atomic_inc(&mNextUniqueId);
+}
+
+// ----------------------------------------------------------------------------
+//  Effect management
+// ----------------------------------------------------------------------------
+
+
+status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle)
+{
+    Mutex::Autolock _l(mLock);
+    return EffectLoadLibrary(libPath, handle);
+}
+
+status_t AudioFlinger::unloadEffectLibrary(int handle)
+{
+    Mutex::Autolock _l(mLock);
+    return EffectUnloadLibrary(handle);
+}
+
+status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
+{
+    Mutex::Autolock _l(mLock);
+    return EffectQueryNumberEffects(numEffects);
+}
+
+status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
+{
+    Mutex::Autolock _l(mLock);
+    return EffectQueryEffect(index, descriptor);
+}
+
+status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
+{
+    Mutex::Autolock _l(mLock);
+    return EffectGetDescriptor(pUuid, descriptor);
+}
+
+
+// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp
+static const effect_uuid_t VISUALIZATION_UUID_ =
+    {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
+
+sp<IEffect> AudioFlinger::createEffect(pid_t pid,
+        effect_descriptor_t *pDesc,
+        const sp<IEffectClient>& effectClient,
+        int32_t priority,
+        int output,
+        int sessionId,
+        status_t *status,
+        int *id,
+        int *enabled)
+{
+    status_t lStatus = NO_ERROR;
+    sp<EffectHandle> handle;
+    effect_interface_t itfe;
+    effect_descriptor_t desc;
+    sp<Client> client;
+    wp<Client> wclient;
+
+    LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", pid, effectClient.get(), priority, sessionId, output);
+
+    if (pDesc == NULL) {
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    {
+        Mutex::Autolock _l(mLock);
+
+        // check recording permission for visualizer
+        if (memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 ||
+            memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) {
+            if (!recordingAllowed()) {
+                lStatus = PERMISSION_DENIED;
+                goto Exit;
+            }
+        }
+
+        if (!EffectIsNullUuid(&pDesc->uuid)) {
+            // if uuid is specified, request effect descriptor
+            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
+            if (lStatus < 0) {
+                LOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
+                goto Exit;
+            }
+        } else {
+            // if uuid is not specified, look for an available implementation
+            // of the required type in effect factory
+            if (EffectIsNullUuid(&pDesc->type)) {
+                LOGW("createEffect() no effect type");
+                lStatus = BAD_VALUE;
+                goto Exit;
+            }
+            uint32_t numEffects = 0;
+            effect_descriptor_t d;
+            bool found = false;
+
+            lStatus = EffectQueryNumberEffects(&numEffects);
+            if (lStatus < 0) {
+                LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
+                goto Exit;
+            }
+            for (uint32_t i = 0; i < numEffects; i++) {
+                lStatus = EffectQueryEffect(i, &desc);
+                if (lStatus < 0) {
+                    LOGW("createEffect() error %d from EffectQueryEffect", lStatus);
+                    continue;
+                }
+                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
+                    // If matching type found save effect descriptor. If the session is
+                    // 0 and the effect is not auxiliary, continue enumeration in case
+                    // an auxiliary version of this effect type is available
+                    found = true;
+                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
+                    if (sessionId != 0 ||
+                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+                        break;
+                    }
+                }
+            }
+            if (!found) {
+                lStatus = BAD_VALUE;
+                LOGW("createEffect() effect not found");
+                goto Exit;
+            }
+            // For same effect type, chose auxiliary version over insert version if
+            // connect to output mix (Compliance to OpenSL ES)
+            if (sessionId == 0 &&
+                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
+                memcpy(&desc, &d, sizeof(effect_descriptor_t));
+            }
+        }
+
+        // Do not allow auxiliary effects on a session different from 0 (output mix)
+        if (sessionId != 0 &&
+             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+            lStatus = INVALID_OPERATION;
+            goto Exit;
+        }
+
+        // Session -1 is reserved for output stage effects that can only be created
+        // by audio policy manager (running in same process)
+        if (sessionId == -1 && getpid() != IPCThreadState::self()->getCallingPid()) {
+            lStatus = INVALID_OPERATION;
+            goto Exit;
+        }
+
+        // return effect descriptor
+        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
+
+        // If output is not specified try to find a matching audio session ID in one of the
+        // output threads.
+        // TODO: allow attachment of effect to inputs
+        if (output == 0) {
+            if (sessionId <= 0) {
+                // default to first output
+                // TODO: define criteria to choose output when not specified. Or
+                // receive output from audio policy manager
+                if (mPlaybackThreads.size() != 0) {
+                    output = mPlaybackThreads.keyAt(0);
+                }
+            } else {
+                 // look for the thread where the specified audio session is present
+                for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+                    if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId)) {
+                        output = mPlaybackThreads.keyAt(i);
+                        break;
+                    }
+                }
+            }
+        }
+        PlaybackThread *thread = checkPlaybackThread_l(output);
+        if (thread == NULL) {
+            LOGE("unknown output thread");
+            lStatus = BAD_VALUE;
+            goto Exit;
+        }
+
+        wclient = mClients.valueFor(pid);
+
+        if (wclient != NULL) {
+            client = wclient.promote();
+        } else {
+            client = new Client(this, pid);
+            mClients.add(pid, client);
+        }
+
+        // create effect on selected output trhead
+        handle = thread->createEffect_l(client, effectClient, priority, sessionId, &desc, enabled, &lStatus);
+        if (handle != 0 && id != NULL) {
+            *id = handle->id();
+        }
+    }
+
+Exit:
+    if(status) {
+        *status = lStatus;
+    }
+    return handle;
+}
+
+status_t AudioFlinger::registerEffectResource_l(effect_descriptor_t *desc) {
+    if (mTotalEffectsCpuLoad + desc->cpuLoad > MAX_EFFECTS_CPU_LOAD) {
+        LOGW("registerEffectResource() CPU Load limit exceeded for Fx %s, CPU %f MIPS",
+                desc->name, (float)desc->cpuLoad/10);
+        return INVALID_OPERATION;
+    }
+    if (mTotalEffectsMemory + desc->memoryUsage > MAX_EFFECTS_MEMORY) {
+        LOGW("registerEffectResource() memory limit exceeded for Fx %s, Memory %d KB",
+                desc->name, desc->memoryUsage);
+        return INVALID_OPERATION;
+    }
+    mTotalEffectsCpuLoad += desc->cpuLoad;
+    mTotalEffectsMemory += desc->memoryUsage;
+    LOGV("registerEffectResource_l() effect %s, CPU %d, memory %d",
+            desc->name, desc->cpuLoad, desc->memoryUsage);
+    LOGV("  total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory);
+    return NO_ERROR;
+}
+
+void AudioFlinger::unregisterEffectResource_l(effect_descriptor_t *desc) {
+    mTotalEffectsCpuLoad -= desc->cpuLoad;
+    mTotalEffectsMemory -= desc->memoryUsage;
+    LOGV("unregisterEffectResource_l() effect %s, CPU %d, memory %d",
+            desc->name, desc->cpuLoad, desc->memoryUsage);
+    LOGV("  total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory);
+}
+
+// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l(
+        const sp<AudioFlinger::Client>& client,
+        const sp<IEffectClient>& effectClient,
+        int32_t priority,
+        int sessionId,
+        effect_descriptor_t *desc,
+        int *enabled,
+        status_t *status
+        )
+{
+    sp<EffectModule> effect;
+    sp<EffectHandle> handle;
+    status_t lStatus;
+    sp<Track> track;
+    sp<EffectChain> chain;
+    bool effectCreated = false;
+    bool effectRegistered = false;
+
+    if (mOutput == 0) {
+        LOGW("createEffect_l() Audio driver not initialized.");
+        lStatus = NO_INIT;
+        goto Exit;
+    }
+
+    // Do not allow auxiliary effect on session other than 0
+    if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY &&
+        sessionId != 0) {
+        LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", desc->name, sessionId);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    // Do not allow effects with session ID 0 on direct output or duplicating threads
+    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
+    if (sessionId == 0 && mType != MIXER) {
+        LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", desc->name, sessionId);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
+    LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
+
+    { // scope for mLock
+        Mutex::Autolock _l(mLock);
+
+        // check for existing effect chain with the requested audio session
+        chain = getEffectChain_l(sessionId);
+        if (chain == 0) {
+            // create a new chain for this session
+            LOGV("createEffect_l() new effect chain for session %d", sessionId);
+            chain = new EffectChain(this, sessionId);
+            addEffectChain_l(chain);
+        } else {
+            effect = chain->getEffectFromDesc(desc);
+        }
+
+        LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
+
+        if (effect == 0) {
+            // Check CPU and memory usage
+            lStatus = mAudioFlinger->registerEffectResource_l(desc);
+            if (lStatus != NO_ERROR) {
+                goto Exit;
+            }
+            effectRegistered = true;
+            // create a new effect module if none present in the chain
+            effect = new EffectModule(this, chain, desc, mAudioFlinger->nextUniqueId(), sessionId);
+            lStatus = effect->status();
+            if (lStatus != NO_ERROR) {
+                goto Exit;
+            }
+            lStatus = chain->addEffect(effect);
+            if (lStatus != NO_ERROR) {
+                goto Exit;
+            }
+            effectCreated = true;
+
+            effect->setDevice(mDevice);
+            effect->setMode(mAudioFlinger->getMode());
+        }
+        // create effect handle and connect it to effect module
+        handle = new EffectHandle(effect, client, effectClient, priority);
+        lStatus = effect->addHandle(handle);
+        if (enabled) {
+            *enabled = (int)effect->isEnabled();
+        }
+    }
+
+Exit:
+    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
+        if (effectCreated) {
+            if (chain->removeEffect(effect) == 0) {
+                removeEffectChain_l(chain);
+            }
+        }
+        if (effectRegistered) {
+            mAudioFlinger->unregisterEffectResource_l(desc);
+        }
+        handle.clear();
+    }
+
+    if(status) {
+        *status = lStatus;
+    }
+    return handle;
+}
+
+void AudioFlinger::PlaybackThread::disconnectEffect(const sp< EffectModule>& effect,
+                                                    const wp<EffectHandle>& handle) {
+    effect_descriptor_t desc = effect->desc();
+    Mutex::Autolock _l(mLock);
+    // delete the effect module if removing last handle on it
+    if (effect->removeHandle(handle) == 0) {
+        if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+            detachAuxEffect_l(effect->id());
+        }
+        sp<EffectChain> chain = effect->chain().promote();
+        if (chain != 0) {
+            // remove effect chain if remove last effect
+            if (chain->removeEffect(effect) == 0) {
+                removeEffectChain_l(chain);
+            }
+        }
+        mLock.unlock();
+        mAudioFlinger->mLock.lock();
+        mAudioFlinger->unregisterEffectResource_l(&desc);
+        mAudioFlinger->mLock.unlock();
+    }
+}
+
+status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
+{
+    int session = chain->sessionId();
+    int16_t *buffer = mMixBuffer;
+    bool ownsBuffer = false;
+
+    LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
+    if (session > 0) {
+        // Only one effect chain can be present in direct output thread and it uses
+        // the mix buffer as input
+        if (mType != DIRECT) {
+            size_t numSamples = mFrameCount * mChannelCount;
+            buffer = new int16_t[numSamples];
+            memset(buffer, 0, numSamples * sizeof(int16_t));
+            LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
+            ownsBuffer = true;
+        }
+
+        // Attach all tracks with same session ID to this chain.
+        for (size_t i = 0; i < mTracks.size(); ++i) {
+            sp<Track> track = mTracks[i];
+            if (session == track->sessionId()) {
+                LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
+                track->setMainBuffer(buffer);
+            }
+        }
+
+        // indicate all active tracks in the chain
+        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
+            sp<Track> track = mActiveTracks[i].promote();
+            if (track == 0) continue;
+            if (session == track->sessionId()) {
+                LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
+                chain->startTrack();
+            }
+        }
+    }
+
+    chain->setInBuffer(buffer, ownsBuffer);
+    chain->setOutBuffer(mMixBuffer);
+    // Effect chain for session -1 is inserted at end of effect chains list
+    // in order to be processed last as it contains output stage effects
+    // Effect chain for session 0 is inserted before session -1 to be processed
+    // after track specific effects and before output stage
+    // Effect chain for session other than 0 is inserted at beginning of effect
+    // chains list to be processed before output mix effects. Relative order between
+    // sessions other than 0 is not important
+    size_t size = mEffectChains.size();
+    size_t i = 0;
+    for (i = 0; i < size; i++) {
+        if (mEffectChains[i]->sessionId() < session) break;
+    }
+    mEffectChains.insertAt(chain, i);
+
+    return NO_ERROR;
+}
+
+size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
+{
+    int session = chain->sessionId();
+
+    LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
+
+    for (size_t i = 0; i < mEffectChains.size(); i++) {
+        if (chain == mEffectChains[i]) {
+            mEffectChains.removeAt(i);
+            // detach all tracks with same session ID from this chain
+            for (size_t i = 0; i < mTracks.size(); ++i) {
+                sp<Track> track = mTracks[i];
+                if (session == track->sessionId()) {
+                    track->setMainBuffer(mMixBuffer);
+                }
+            }
+        }
+    }
+    return mEffectChains.size();
+}
+
+void AudioFlinger::PlaybackThread::lockEffectChains_l()
+{
+    for (size_t i = 0; i < mEffectChains.size(); i++) {
+        mEffectChains[i]->lock();
+    }
+}
+
+void AudioFlinger::PlaybackThread::unlockEffectChains()
+{
+    Mutex::Autolock _l(mLock);
+    for (size_t i = 0; i < mEffectChains.size(); i++) {
+        mEffectChains[i]->unlock();
+    }
+}
+
+sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId)
+{
+    sp<EffectModule> effect;
+
+    sp<EffectChain> chain = getEffectChain_l(sessionId);
+    if (chain != 0) {
+        effect = chain->getEffectFromId(effectId);
+    }
+    return effect;
+}
+
+status_t AudioFlinger::PlaybackThread::attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
+{
+    Mutex::Autolock _l(mLock);
+    return attachAuxEffect_l(track, EffectId);
+}
+
+status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
+{
+    status_t status = NO_ERROR;
+
+    if (EffectId == 0) {
+        track->setAuxBuffer(0, NULL);
+    } else {
+        // Auxiliary effects are always in audio session 0
+        sp<EffectModule> effect = getEffect_l(0, EffectId);
+        if (effect != 0) {
+            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
+            } else {
+                status = INVALID_OPERATION;
+            }
+        } else {
+            status = BAD_VALUE;
+        }
+    }
+    return status;
+}
+
+void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
+{
+     for (size_t i = 0; i < mTracks.size(); ++i) {
+        sp<Track> track = mTracks[i];
+        if (track->auxEffectId() == effectId) {
+            attachAuxEffect_l(track, 0);
+        }
+    }
+}
+
+// ----------------------------------------------------------------------------
+//  EffectModule implementation
+// ----------------------------------------------------------------------------
+
+#undef LOG_TAG
+#define LOG_TAG "AudioFlinger::EffectModule"
+
+AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
+                                        const wp<AudioFlinger::EffectChain>& chain,
+                                        effect_descriptor_t *desc,
+                                        int id,
+                                        int sessionId)
+    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
+      mStatus(NO_INIT), mState(IDLE)
+{
+    LOGV("Constructor %p", this);
+    int lStatus;
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread == 0) {
+        return;
+    }
+    PlaybackThread *p = (PlaybackThread *)thread.get();
+
+    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
+
+    // create effect engine from effect factory
+    mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface);
+
+    if (mStatus != NO_ERROR) {
+        return;
+    }
+    lStatus = init();
+    if (lStatus < 0) {
+        mStatus = lStatus;
+        goto Error;
+    }
+
+    LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
+    return;
+Error:
+    EffectRelease(mEffectInterface);
+    mEffectInterface = NULL;
+    LOGV("Constructor Error %d", mStatus);
+}
+
+AudioFlinger::EffectModule::~EffectModule()
+{
+    LOGV("Destructor %p", this);
+    if (mEffectInterface != NULL) {
+        // release effect engine
+        EffectRelease(mEffectInterface);
+    }
+}
+
+status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
+{
+    status_t status;
+
+    Mutex::Autolock _l(mLock);
+    // First handle in mHandles has highest priority and controls the effect module
+    int priority = handle->priority();
+    size_t size = mHandles.size();
+    sp<EffectHandle> h;
+    size_t i;
+    for (i = 0; i < size; i++) {
+        h = mHandles[i].promote();
+        if (h == 0) continue;
+        if (h->priority() <= priority) break;
+    }
+    // if inserted in first place, move effect control from previous owner to this handle
+    if (i == 0) {
+        if (h != 0) {
+            h->setControl(false, true);
+        }
+        handle->setControl(true, false);
+        status = NO_ERROR;
+    } else {
+        status = ALREADY_EXISTS;
+    }
+    mHandles.insertAt(handle, i);
+    return status;
+}
+
+size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
+{
+    Mutex::Autolock _l(mLock);
+    size_t size = mHandles.size();
+    size_t i;
+    for (i = 0; i < size; i++) {
+        if (mHandles[i] == handle) break;
+    }
+    if (i == size) {
+        return size;
+    }
+    mHandles.removeAt(i);
+    size = mHandles.size();
+    // if removed from first place, move effect control from this handle to next in line
+    if (i == 0 && size != 0) {
+        sp<EffectHandle> h = mHandles[0].promote();
+        if (h != 0) {
+            h->setControl(true, true);
+        }
+    }
+
+    return size;
+}
+
+void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle)
+{
+    // keep a strong reference on this EffectModule to avoid calling the
+    // destructor before we exit
+    sp<EffectModule> keep(this);
+    {
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+            playbackThread->disconnectEffect(keep, handle);
+        }
+    }
+}
+
+void AudioFlinger::EffectModule::updateState() {
+    Mutex::Autolock _l(mLock);
+
+    switch (mState) {
+    case RESTART:
+        reset_l();
+        // FALL THROUGH
+
+    case STARTING:
+        // clear auxiliary effect input buffer for next accumulation
+        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+            memset(mConfig.inputCfg.buffer.raw,
+                   0,
+                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
+        }
+        start_l();
+        mState = ACTIVE;
+        break;
+    case STOPPING:
+        stop_l();
+        mDisableWaitCnt = mMaxDisableWaitCnt;
+        mState = STOPPED;
+        break;
+    case STOPPED:
+        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
+        // turn off sequence.
+        if (--mDisableWaitCnt == 0) {
+            reset_l();
+            mState = IDLE;
+        }
+        break;
+    default: //IDLE , ACTIVE
+        break;
+    }
+}
+
+void AudioFlinger::EffectModule::process()
+{
+    Mutex::Autolock _l(mLock);
+
+    if (mEffectInterface == NULL ||
+            mConfig.inputCfg.buffer.raw == NULL ||
+            mConfig.outputCfg.buffer.raw == NULL) {
+        return;
+    }
+
+    if (mState == ACTIVE || mState == STOPPING || mState == STOPPED) {
+        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
+        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+            AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32,
+                                        mConfig.inputCfg.buffer.s32,
+                                        mConfig.inputCfg.buffer.frameCount);
+        }
+
+        // do the actual processing in the effect engine
+        int ret = (*mEffectInterface)->process(mEffectInterface,
+                                               &mConfig.inputCfg.buffer,
+                                               &mConfig.outputCfg.buffer);
+
+        // force transition to IDLE state when engine is ready
+        if (mState == STOPPED && ret == -ENODATA) {
+            mDisableWaitCnt = 1;
+        }
+
+        // clear auxiliary effect input buffer for next accumulation
+        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+            memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
+        }
+    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
+                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw){
+        // If an insert effect is idle and input buffer is different from output buffer, copy input to
+        // output
+        sp<EffectChain> chain = mChain.promote();
+        if (chain != 0 && chain->activeTracks() != 0) {
+            size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t);
+            if (mConfig.inputCfg.channels == CHANNEL_STEREO) {
+                size *= 2;
+            }
+            memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size);
+        }
+    }
+}
+
+void AudioFlinger::EffectModule::reset_l()
+{
+    if (mEffectInterface == NULL) {
+        return;
+    }
+    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
+}
+
+status_t AudioFlinger::EffectModule::configure()
+{
+    uint32_t channels;
+    if (mEffectInterface == NULL) {
+        return NO_INIT;
+    }
+
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread == 0) {
+        return DEAD_OBJECT;
+    }
+
+    // TODO: handle configuration of effects replacing track process
+    if (thread->channelCount() == 1) {
+        channels = CHANNEL_MONO;
+    } else {
+        channels = CHANNEL_STEREO;
+    }
+
+    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+        mConfig.inputCfg.channels = CHANNEL_MONO;
+    } else {
+        mConfig.inputCfg.channels = channels;
+    }
+    mConfig.outputCfg.channels = channels;
+    mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15;
+    mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15;
+    mConfig.inputCfg.samplingRate = thread->sampleRate();
+    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
+    mConfig.inputCfg.bufferProvider.cookie = NULL;
+    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
+    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
+    mConfig.outputCfg.bufferProvider.cookie = NULL;
+    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
+    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
+    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
+    // Insert effect:
+    // - in session 0 or -1, always overwrites output buffer: input buffer == output buffer
+    // - in other sessions:
+    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
+    //      other effect: overwrites output buffer: input buffer == output buffer
+    // Auxiliary effect:
+    //      accumulates in output buffer: input buffer != output buffer
+    // Therefore: accumulate <=> input buffer != output buffer
+    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
+        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
+    } else {
+        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
+    }
+    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
+    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
+    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
+    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
+
+    status_t cmdStatus;
+    int size = sizeof(int);
+    status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_CONFIGURE, sizeof(effect_config_t), &mConfig, &size, &cmdStatus);
+    if (status == 0) {
+        status = cmdStatus;
+    }
+
+    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
+            (1000 * mConfig.outputCfg.buffer.frameCount);
+
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::init()
+{
+    Mutex::Autolock _l(mLock);
+    if (mEffectInterface == NULL) {
+        return NO_INIT;
+    }
+    status_t cmdStatus;
+    int size = sizeof(status_t);
+    status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_INIT, 0, NULL, &size, &cmdStatus);
+    if (status == 0) {
+        status = cmdStatus;
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::start_l()
+{
+    if (mEffectInterface == NULL) {
+        return NO_INIT;
+    }
+    status_t cmdStatus;
+    int size = sizeof(status_t);
+    status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_ENABLE, 0, NULL, &size, &cmdStatus);
+    if (status == 0) {
+        status = cmdStatus;
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::stop_l()
+{
+    if (mEffectInterface == NULL) {
+        return NO_INIT;
+    }
+    status_t cmdStatus;
+    int size = sizeof(status_t);
+    status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_DISABLE, 0, NULL, &size, &cmdStatus);
+    if (status == 0) {
+        status = cmdStatus;
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData)
+{
+    Mutex::Autolock _l(mLock);
+//    LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
+
+    if (mEffectInterface == NULL) {
+        return NO_INIT;
+    }
+    status_t status = (*mEffectInterface)->command(mEffectInterface, cmdCode, cmdSize, pCmdData, replySize, pReplyData);
+    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
+        int size = (replySize == NULL) ? 0 : *replySize;
+        for (size_t i = 1; i < mHandles.size(); i++) {
+            sp<EffectHandle> h = mHandles[i].promote();
+            if (h != 0) {
+                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
+            }
+        }
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
+{
+    Mutex::Autolock _l(mLock);
+    LOGV("setEnabled %p enabled %d", this, enabled);
+
+    if (enabled != isEnabled()) {
+        switch (mState) {
+        // going from disabled to enabled
+        case IDLE:
+            mState = STARTING;
+            break;
+        case STOPPED:
+            mState = RESTART;
+            break;
+        case STOPPING:
+            mState = ACTIVE;
+            break;
+
+        // going from enabled to disabled
+        case RESTART:
+        case STARTING:
+            mState = IDLE;
+            break;
+        case ACTIVE:
+            mState = STOPPING;
+            break;
+        }
+        for (size_t i = 1; i < mHandles.size(); i++) {
+            sp<EffectHandle> h = mHandles[i].promote();
+            if (h != 0) {
+                h->setEnabled(enabled);
+            }
+        }
+    }
+    return NO_ERROR;
+}
+
+bool AudioFlinger::EffectModule::isEnabled()
+{
+    switch (mState) {
+    case RESTART:
+    case STARTING:
+    case ACTIVE:
+        return true;
+    case IDLE:
+    case STOPPING:
+    case STOPPED:
+    default:
+        return false;
+    }
+}
+
+status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
+{
+    Mutex::Autolock _l(mLock);
+    status_t status = NO_ERROR;
+
+    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
+    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
+    if ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) & (EFFECT_FLAG_VOLUME_CTRL|EFFECT_FLAG_VOLUME_IND)) {
+        status_t cmdStatus;
+        uint32_t volume[2];
+        uint32_t *pVolume = NULL;
+        int size = sizeof(volume);
+        volume[0] = *left;
+        volume[1] = *right;
+        if (controller) {
+            pVolume = volume;
+        }
+        status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_VOLUME, size, volume, &size, pVolume);
+        if (controller && status == NO_ERROR && size == sizeof(volume)) {
+            *left = volume[0];
+            *right = volume[1];
+        }
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
+{
+    Mutex::Autolock _l(mLock);
+    status_t status = NO_ERROR;
+    if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
+        // convert device bit field from AudioSystem to EffectApi format.
+        device = deviceAudioSystemToEffectApi(device);
+        if (device == 0) {
+            return BAD_VALUE;
+        }
+        status_t cmdStatus;
+        int size = sizeof(status_t);
+        status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_DEVICE, sizeof(uint32_t), &device, &size, &cmdStatus);
+        if (status == NO_ERROR) {
+            status = cmdStatus;
+        }
+    }
+    return status;
+}
+
+status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
+{
+    Mutex::Autolock _l(mLock);
+    status_t status = NO_ERROR;
+    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
+        // convert audio mode from AudioSystem to EffectApi format.
+        int effectMode = modeAudioSystemToEffectApi(mode);
+        if (effectMode < 0) {
+            return BAD_VALUE;
+        }
+        status_t cmdStatus;
+        int size = sizeof(status_t);
+        status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_AUDIO_MODE, sizeof(int), &effectMode, &size, &cmdStatus);
+        if (status == NO_ERROR) {
+            status = cmdStatus;
+        }
+    }
+    return status;
+}
+
+// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified
+const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = {
+    DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE
+    DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER
+    DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET
+    DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE
+    DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO
+    DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET
+    DEVICE_BLUETOOTH_SCO_CARKIT, //  AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT
+    DEVICE_BLUETOOTH_A2DP, //  AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP
+    DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES
+    DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
+    DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL
+};
+
+uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device)
+{
+    uint32_t deviceOut = 0;
+    while (device) {
+        const uint32_t i = 31 - __builtin_clz(device);
+        device &= ~(1 << i);
+        if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) {
+            LOGE("device convertion error for AudioSystem device 0x%08x", device);
+            return 0;
+        }
+        deviceOut |= (uint32_t)sDeviceConvTable[i];
+    }
+    return deviceOut;
+}
+
+// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified
+const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = {
+    AUDIO_MODE_NORMAL,   // AudioSystem::MODE_NORMAL
+    AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE
+    AUDIO_MODE_IN_CALL   // AudioSystem::MODE_IN_CALL
+};
+
+int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode)
+{
+    int modeOut = -1;
+    if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) {
+        modeOut = (int)sModeConvTable[mode];
+    }
+    return modeOut;
+}
+
+status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
+    result.append(buffer);
+
+    bool locked = tryLock(mLock);
+    // failed to lock - AudioFlinger is probably deadlocked
+    if (!locked) {
+        result.append("\t\tCould not lock Fx mutex:\n");
+    }
+
+    result.append("\t\tSession Status State Engine:\n");
+    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
+            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
+    result.append(buffer);
+
+    result.append("\t\tDescriptor:\n");
+    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
+            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
+            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
+            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
+                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
+                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
+                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n",
+            mDescriptor.apiVersion,
+            mDescriptor.flags);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\t- name: %s\n",
+            mDescriptor.name);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
+            mDescriptor.implementor);
+    result.append(buffer);
+
+    result.append("\t\t- Input configuration:\n");
+    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
+    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
+            (uint32_t)mConfig.inputCfg.buffer.raw,
+            mConfig.inputCfg.buffer.frameCount,
+            mConfig.inputCfg.samplingRate,
+            mConfig.inputCfg.channels,
+            mConfig.inputCfg.format);
+    result.append(buffer);
+
+    result.append("\t\t- Output configuration:\n");
+    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
+    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
+            (uint32_t)mConfig.outputCfg.buffer.raw,
+            mConfig.outputCfg.buffer.frameCount,
+            mConfig.outputCfg.samplingRate,
+            mConfig.outputCfg.channels,
+            mConfig.outputCfg.format);
+    result.append(buffer);
+
+    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
+    result.append(buffer);
+    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
+    for (size_t i = 0; i < mHandles.size(); ++i) {
+        sp<EffectHandle> handle = mHandles[i].promote();
+        if (handle != 0) {
+            handle->dump(buffer, SIZE);
+            result.append(buffer);
+        }
+    }
+
+    result.append("\n");
+
+    write(fd, result.string(), result.length());
+
+    if (locked) {
+        mLock.unlock();
+    }
+
+    return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+//  EffectHandle implementation
+// ----------------------------------------------------------------------------
+
+#undef LOG_TAG
+#define LOG_TAG "AudioFlinger::EffectHandle"
+
+AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
+                                        const sp<AudioFlinger::Client>& client,
+                                        const sp<IEffectClient>& effectClient,
+                                        int32_t priority)
+    : BnEffect(),
+    mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false)
+{
+    LOGV("constructor %p", this);
+
+    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
+    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
+    if (mCblkMemory != 0) {
+        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
+
+        if (mCblk) {
+            new(mCblk) effect_param_cblk_t();
+            mBuffer = (uint8_t *)mCblk + bufOffset;
+         }
+    } else {
+        LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
+        return;
+    }
+}
+
+AudioFlinger::EffectHandle::~EffectHandle()
+{
+    LOGV("Destructor %p", this);
+    disconnect();
+}
+
+status_t AudioFlinger::EffectHandle::enable()
+{
+    if (!mHasControl) return INVALID_OPERATION;
+    if (mEffect == 0) return DEAD_OBJECT;
+
+    return mEffect->setEnabled(true);
+}
+
+status_t AudioFlinger::EffectHandle::disable()
+{
+    if (!mHasControl) return INVALID_OPERATION;
+    if (mEffect == NULL) return DEAD_OBJECT;
+
+    return mEffect->setEnabled(false);
+}
+
+void AudioFlinger::EffectHandle::disconnect()
+{
+    if (mEffect == 0) {
+        return;
+    }
+    mEffect->disconnect(this);
+    // release sp on module => module destructor can be called now
+    mEffect.clear();
+    if (mCblk) {
+        mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
+    }
+    mCblkMemory.clear();            // and free the shared memory
+    if (mClient != 0) {
+        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
+        mClient.clear();
+    }
+}
+
+status_t AudioFlinger::EffectHandle::command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData)
+{
+//    LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
+
+    // only get parameter command is permitted for applications not controlling the effect
+    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
+        return INVALID_OPERATION;
+    }
+    if (mEffect == 0) return DEAD_OBJECT;
+
+    // handle commands that are not forwarded transparently to effect engine
+    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
+        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
+        // no risk to block the whole media server process or mixer threads is we are stuck here
+        Mutex::Autolock _l(mCblk->lock);
+        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
+            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
+            mCblk->serverIndex = 0;
+            mCblk->clientIndex = 0;
+            return BAD_VALUE;
+        }
+        status_t status = NO_ERROR;
+        while (mCblk->serverIndex < mCblk->clientIndex) {
+            int reply;
+            int rsize = sizeof(int);
+            int *p = (int *)(mBuffer + mCblk->serverIndex);
+            int size = *p++;
+            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
+                LOGW("command(): invalid parameter block size");
+                break;
+            }
+            effect_param_t *param = (effect_param_t *)p;
+            if (param->psize == 0 || param->vsize == 0) {
+                LOGW("command(): null parameter or value size");
+                mCblk->serverIndex += size;
+                continue;
+            }
+            int psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize;
+            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, psize, p, &rsize, &reply);
+            if (ret == NO_ERROR) {
+                if (reply != NO_ERROR) {
+                    status = reply;
+                }
+            } else {
+                status = ret;
+            }
+            mCblk->serverIndex += size;
+        }
+        mCblk->serverIndex = 0;
+        mCblk->clientIndex = 0;
+        return status;
+    } else if (cmdCode == EFFECT_CMD_ENABLE) {
+        return enable();
+    } else if (cmdCode == EFFECT_CMD_DISABLE) {
+        return disable();
+    }
+
+    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
+}
+
+sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
+    return mCblkMemory;
+}
+
+void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal)
+{
+    LOGV("setControl %p control %d", this, hasControl);
+
+    mHasControl = hasControl;
+    if (signal && mEffectClient != 0) {
+        mEffectClient->controlStatusChanged(hasControl);
+    }
+}
+
+void AudioFlinger::EffectHandle::commandExecuted(int cmdCode, int cmdSize, void *pCmdData, int replySize, void *pReplyData)
+{
+    if (mEffectClient != 0) {
+        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
+    }
+}
+
+
+
+void AudioFlinger::EffectHandle::setEnabled(bool enabled)
+{
+    if (mEffectClient != 0) {
+        mEffectClient->enableStatusChanged(enabled);
+    }
+}
+
+status_t AudioFlinger::EffectHandle::onTransact(
+    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+    return BnEffect::onTransact(code, data, reply, flags);
+}
+
+
+void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
+{
+    bool locked = tryLock(mCblk->lock);
+
+    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
+            (mClient == NULL) ? getpid() : mClient->pid(),
+            mPriority,
+            mHasControl,
+            !locked,
+            mCblk->clientIndex,
+            mCblk->serverIndex
+            );
+
+    if (locked) {
+        mCblk->lock.unlock();
+    }
+}
+
+#undef LOG_TAG
+#define LOG_TAG "AudioFlinger::EffectChain"
+
+AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
+                                        int sessionId)
+    : mThread(wThread), mSessionId(sessionId), mVolumeCtrlIdx(-1), mActiveTrackCnt(0), mOwnInBuffer(false)
+{
+
+}
+
+AudioFlinger::EffectChain::~EffectChain()
+{
+    if (mOwnInBuffer) {
+        delete mInBuffer;
+    }
+
+}
+
+sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc(effect_descriptor_t *descriptor)
+{
+    sp<EffectModule> effect;
+    size_t size = mEffects.size();
+
+    for (size_t i = 0; i < size; i++) {
+        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
+            effect = mEffects[i];
+            break;
+        }
+    }
+    return effect;
+}
+
+sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId(int id)
+{
+    sp<EffectModule> effect;
+    size_t size = mEffects.size();
+
+    for (size_t i = 0; i < size; i++) {
+        if (mEffects[i]->id() == id) {
+            effect = mEffects[i];
+            break;
+        }
+    }
+    return effect;
+}
+
+// Must be called with EffectChain::mLock locked
+void AudioFlinger::EffectChain::process_l()
+{
+    size_t size = mEffects.size();
+    for (size_t i = 0; i < size; i++) {
+        mEffects[i]->process();
+    }
+    for (size_t i = 0; i < size; i++) {
+        mEffects[i]->updateState();
+    }
+    // if no track is active, input buffer must be cleared here as the mixer process
+    // will not do it
+    if (mSessionId > 0 && activeTracks() == 0) {
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread != 0) {
+            size_t numSamples = thread->frameCount() * thread->channelCount();
+            memset(mInBuffer, 0, numSamples * sizeof(int16_t));
+        }
+    }
+}
+
+status_t AudioFlinger::EffectChain::addEffect(sp<EffectModule>& effect)
+{
+    effect_descriptor_t desc = effect->desc();
+    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
+
+    Mutex::Autolock _l(mLock);
+
+    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+        // Auxiliary effects are inserted at the beginning of mEffects vector as
+        // they are processed first and accumulated in chain input buffer
+        mEffects.insertAt(effect, 0);
+        sp<ThreadBase> thread = mThread.promote();
+        if (thread == 0) {
+            return NO_INIT;
+        }
+        // the input buffer for auxiliary effect contains mono samples in
+        // 32 bit format. This is to avoid saturation in AudoMixer
+        // accumulation stage. Saturation is done in EffectModule::process() before
+        // calling the process in effect engine
+        size_t numSamples = thread->frameCount();
+        int32_t *buffer = new int32_t[numSamples];
+        memset(buffer, 0, numSamples * sizeof(int32_t));
+        effect->setInBuffer((int16_t *)buffer);
+        // auxiliary effects output samples to chain input buffer for further processing
+        // by insert effects
+        effect->setOutBuffer(mInBuffer);
+    } else {
+        // Insert effects are inserted at the end of mEffects vector as they are processed
+        //  after track and auxiliary effects.
+        // Insert effect order as a function of indicated preference:
+        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
+        //  another effect is present
+        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
+        //  last effect claiming first position
+        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
+        //  first effect claiming last position
+        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
+        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
+        // already present
+
+        int size = (int)mEffects.size();
+        int idx_insert = size;
+        int idx_insert_first = -1;
+        int idx_insert_last = -1;
+
+        for (int i = 0; i < size; i++) {
+            effect_descriptor_t d = mEffects[i]->desc();
+            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
+            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
+            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
+                // check invalid effect chaining combinations
+                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
+                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
+                    LOGW("addEffect() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
+                    return INVALID_OPERATION;
+                }
+                // remember position of first insert effect and by default
+                // select this as insert position for new effect
+                if (idx_insert == size) {
+                    idx_insert = i;
+                }
+                // remember position of last insert effect claiming
+                // first position
+                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
+                    idx_insert_first = i;
+                }
+                // remember position of first insert effect claiming
+                // last position
+                if (iPref == EFFECT_FLAG_INSERT_LAST &&
+                    idx_insert_last == -1) {
+                    idx_insert_last = i;
+                }
+            }
+        }
+
+        // modify idx_insert from first position if needed
+        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
+            if (idx_insert_last != -1) {
+                idx_insert = idx_insert_last;
+            } else {
+                idx_insert = size;
+            }
+        } else {
+            if (idx_insert_first != -1) {
+                idx_insert = idx_insert_first + 1;
+            }
+        }
+
+        // always read samples from chain input buffer
+        effect->setInBuffer(mInBuffer);
+
+        // if last effect in the chain, output samples to chain
+        // output buffer, otherwise to chain input buffer
+        if (idx_insert == size) {
+            if (idx_insert != 0) {
+                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
+                mEffects[idx_insert-1]->configure();
+            }
+            effect->setOutBuffer(mOutBuffer);
+        } else {
+            effect->setOutBuffer(mInBuffer);
+        }
+        mEffects.insertAt(effect, idx_insert);
+        // Always give volume control to last effect in chain with volume control capability
+        if (((desc.flags & EFFECT_FLAG_VOLUME_MASK) & EFFECT_FLAG_VOLUME_CTRL) &&
+                mVolumeCtrlIdx < idx_insert) {
+            mVolumeCtrlIdx = idx_insert;
+        }
+
+        LOGV("addEffect() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
+    }
+    effect->configure();
+    return NO_ERROR;
+}
+
+size_t AudioFlinger::EffectChain::removeEffect(const sp<EffectModule>& effect)
+{
+    Mutex::Autolock _l(mLock);
+
+    int size = (int)mEffects.size();
+    int i;
+    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
+
+    for (i = 0; i < size; i++) {
+        if (effect == mEffects[i]) {
+            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
+                delete[] effect->inBuffer();
+            } else {
+                if (i == size - 1 && i != 0) {
+                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
+                    mEffects[i - 1]->configure();
+                }
+            }
+            mEffects.removeAt(i);
+            LOGV("removeEffect() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
+            break;
+        }
+    }
+    // Return volume control to last effect in chain with volume control capability
+    if (mVolumeCtrlIdx == i) {
+        size = (int)mEffects.size();
+        for (i = size; i > 0; i--) {
+            if ((mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) & EFFECT_FLAG_VOLUME_CTRL) {
+                break;
+            }
+        }
+        // mVolumeCtrlIdx reset to -1 if no effect found with volume control flag set
+        mVolumeCtrlIdx = i - 1;
+    }
+
+    return mEffects.size();
+}
+
+void AudioFlinger::EffectChain::setDevice(uint32_t device)
+{
+    size_t size = mEffects.size();
+    for (size_t i = 0; i < size; i++) {
+        mEffects[i]->setDevice(device);
+    }
+}
+
+void AudioFlinger::EffectChain::setMode(uint32_t mode)
+{
+    size_t size = mEffects.size();
+    for (size_t i = 0; i < size; i++) {
+        mEffects[i]->setMode(mode);
+    }
+}
+
+bool AudioFlinger::EffectChain::setVolume(uint32_t *left, uint32_t *right)
+{
+    uint32_t newLeft = *left;
+    uint32_t newRight = *right;
+    bool hasControl = false;
+
+    // first get volume update from volume controller
+    if (mVolumeCtrlIdx >= 0) {
+        mEffects[mVolumeCtrlIdx]->setVolume(&newLeft, &newRight, true);
+        hasControl = true;
+    }
+    // then indicate volume to all other effects in chain.
+    // Pass altered volume to effects before volume controller
+    // and requested volume to effects after controller
+    uint32_t lVol = newLeft;
+    uint32_t rVol = newRight;
+    size_t size = mEffects.size();
+    for (size_t i = 0; i < size; i++) {
+        if ((int)i == mVolumeCtrlIdx) continue;
+        // this also works for mVolumeCtrlIdx == -1 when there is no volume controller
+        if ((int)i > mVolumeCtrlIdx) {
+            lVol = *left;
+            rVol = *right;
+        }
+        mEffects[i]->setVolume(&lVol, &rVol, false);
+    }
+    *left = newLeft;
+    *right = newRight;
+
+    return hasControl;
+}
+
+sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getVolumeController()
+{
+    sp<EffectModule> effect;
+    if (mVolumeCtrlIdx >= 0) {
+        effect = mEffects[mVolumeCtrlIdx];
+    }
+    return effect;
+}
+
+
+status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
+    result.append(buffer);
+
+    bool locked = tryLock(mLock);
+    // failed to lock - AudioFlinger is probably deadlocked
+    if (!locked) {
+        result.append("\tCould not lock mutex:\n");
+    }
+
+    result.append("\tNum fx In buffer   Out buffer   Vol ctrl Active tracks:\n");
+    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %02d       %d\n",
+            mEffects.size(),
+            (uint32_t)mInBuffer,
+            (uint32_t)mOutBuffer,
+            (mVolumeCtrlIdx == -1) ? 0 : mEffects[mVolumeCtrlIdx]->id(),
+            mActiveTrackCnt);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    for (size_t i = 0; i < mEffects.size(); ++i) {
+        sp<EffectModule> effect = mEffects[i];
+        if (effect != 0) {
+            effect->dump(fd, args);
+        }
+    }
+
+    if (locked) {
+        mLock.unlock();
+    }
+
+    return NO_ERROR;
+}
+
+#undef LOG_TAG
+#define LOG_TAG "AudioFlinger"
+
+// ----------------------------------------------------------------------------
+
+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