diff --git a/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp b/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp
new file mode 100644
index 0000000..a8e1d1f
--- /dev/null
+++ b/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp
@@ -0,0 +1,727 @@
+/*
+**
+** Copyright 2018, 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_NDEBUG 0
+#define LOG_TAG "MediaPlayer2AudioOutput"
+#include <mediaplayer2/MediaPlayer2AudioOutput.h>
+
+#include <cutils/properties.h> // for property_get
+#include <utils/Log.h>
+
+#include <media/AudioPolicyHelper.h>
+#include <media/AudioTrack.h>
+#include <media/stagefright/foundation/ADebug.h>
+
+namespace {
+
+const float kMaxRequiredSpeed = 8.0f; // for PCM tracks allow up to 8x speedup.
+
+} // anonymous namespace
+
+namespace android {
+
+// TODO: Find real cause of Audio/Video delay in PV framework and remove this workaround
+/* static */ int MediaPlayer2AudioOutput::mMinBufferCount = 4;
+/* static */ bool MediaPlayer2AudioOutput::mIsOnEmulator = false;
+
+status_t MediaPlayer2AudioOutput::dump(int fd, const Vector<String16>& args) const {
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    result.append(" MediaPlayer2AudioOutput\n");
+    snprintf(buffer, 255, "  stream type(%d), left - right volume(%f, %f)\n",
+            mStreamType, mLeftVolume, mRightVolume);
+    result.append(buffer);
+    snprintf(buffer, 255, "  msec per frame(%f), latency (%d)\n",
+            mMsecsPerFrame, (mTrack != 0) ? mTrack->latency() : -1);
+    result.append(buffer);
+    snprintf(buffer, 255, "  aux effect id(%d), send level (%f)\n",
+            mAuxEffectId, mSendLevel);
+    result.append(buffer);
+
+    ::write(fd, result.string(), result.size());
+    if (mTrack != 0) {
+        mTrack->dump(fd, args);
+    }
+    return NO_ERROR;
+}
+
+MediaPlayer2AudioOutput::MediaPlayer2AudioOutput(audio_session_t sessionId, uid_t uid, int pid,
+        const audio_attributes_t* attr, const sp<AudioSystem::AudioDeviceCallback>& deviceCallback)
+    : mCallback(NULL),
+      mCallbackCookie(NULL),
+      mCallbackData(NULL),
+      mStreamType(AUDIO_STREAM_MUSIC),
+      mLeftVolume(1.0),
+      mRightVolume(1.0),
+      mPlaybackRate(AUDIO_PLAYBACK_RATE_DEFAULT),
+      mSampleRateHz(0),
+      mMsecsPerFrame(0),
+      mFrameSize(0),
+      mSessionId(sessionId),
+      mUid(uid),
+      mPid(pid),
+      mSendLevel(0.0),
+      mAuxEffectId(0),
+      mFlags(AUDIO_OUTPUT_FLAG_NONE),
+      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
+      mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
+      mDeviceCallbackEnabled(false),
+      mDeviceCallback(deviceCallback) {
+    ALOGV("MediaPlayer2AudioOutput(%d)", sessionId);
+    if (attr != NULL) {
+        mAttributes = (audio_attributes_t *) calloc(1, sizeof(audio_attributes_t));
+        if (mAttributes != NULL) {
+            memcpy(mAttributes, attr, sizeof(audio_attributes_t));
+            mStreamType = audio_attributes_to_stream_type(attr);
+        }
+    } else {
+        mAttributes = NULL;
+    }
+
+    setMinBufferCount();
+}
+
+MediaPlayer2AudioOutput::~MediaPlayer2AudioOutput() {
+    close();
+    free(mAttributes);
+    delete mCallbackData;
+}
+
+//static
+void MediaPlayer2AudioOutput::setMinBufferCount() {
+    char value[PROPERTY_VALUE_MAX];
+    if (property_get("ro.kernel.qemu", value, 0)) {
+        mIsOnEmulator = true;
+        mMinBufferCount = 12;  // to prevent systematic buffer underrun for emulator
+    }
+}
+
+// static
+bool MediaPlayer2AudioOutput::isOnEmulator() {
+    setMinBufferCount();  // benign race wrt other threads
+    return mIsOnEmulator;
+}
+
+// static
+int MediaPlayer2AudioOutput::getMinBufferCount() {
+    setMinBufferCount();  // benign race wrt other threads
+    return mMinBufferCount;
+}
+
+ssize_t MediaPlayer2AudioOutput::bufferSize() const {
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return NO_INIT;
+    }
+    return mTrack->frameCount() * mFrameSize;
+}
+
+ssize_t MediaPlayer2AudioOutput::frameCount() const {
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return NO_INIT;
+    }
+    return mTrack->frameCount();
+}
+
+ssize_t MediaPlayer2AudioOutput::channelCount() const {
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return NO_INIT;
+    }
+    return mTrack->channelCount();
+}
+
+ssize_t MediaPlayer2AudioOutput::frameSize() const {
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return NO_INIT;
+    }
+    return mFrameSize;
+}
+
+uint32_t MediaPlayer2AudioOutput::latency () const {
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return 0;
+    }
+    return mTrack->latency();
+}
+
+float MediaPlayer2AudioOutput::msecsPerFrame() const {
+    Mutex::Autolock lock(mLock);
+    return mMsecsPerFrame;
+}
+
+status_t MediaPlayer2AudioOutput::getPosition(uint32_t *position) const {
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return NO_INIT;
+    }
+    return mTrack->getPosition(position);
+}
+
+status_t MediaPlayer2AudioOutput::getTimestamp(AudioTimestamp &ts) const {
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return NO_INIT;
+    }
+    return mTrack->getTimestamp(ts);
+}
+
+// TODO: Remove unnecessary calls to getPlayedOutDurationUs()
+// as it acquires locks and may query the audio driver.
+//
+// Some calls could conceivably retrieve extrapolated data instead of
+// accessing getTimestamp() or getPosition() every time a data buffer with
+// a media time is received.
+//
+// Calculate duration of played samples if played at normal rate (i.e., 1.0).
+int64_t MediaPlayer2AudioOutput::getPlayedOutDurationUs(int64_t nowUs) const {
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0 || mSampleRateHz == 0) {
+        return 0;
+    }
+
+    uint32_t numFramesPlayed;
+    int64_t numFramesPlayedAtUs;
+    AudioTimestamp ts;
+
+    status_t res = mTrack->getTimestamp(ts);
+    if (res == OK) {                 // case 1: mixing audio tracks and offloaded tracks.
+        numFramesPlayed = ts.mPosition;
+        numFramesPlayedAtUs = ts.mTime.tv_sec * 1000000LL + ts.mTime.tv_nsec / 1000;
+        //ALOGD("getTimestamp: OK %d %lld", numFramesPlayed, (long long)numFramesPlayedAtUs);
+    } else if (res == WOULD_BLOCK) { // case 2: transitory state on start of a new track
+        numFramesPlayed = 0;
+        numFramesPlayedAtUs = nowUs;
+        //ALOGD("getTimestamp: WOULD_BLOCK %d %lld",
+        //        numFramesPlayed, (long long)numFramesPlayedAtUs);
+    } else {                         // case 3: transitory at new track or audio fast tracks.
+        res = mTrack->getPosition(&numFramesPlayed);
+        CHECK_EQ(res, (status_t)OK);
+        numFramesPlayedAtUs = nowUs;
+        numFramesPlayedAtUs += 1000LL * mTrack->latency() / 2; /* XXX */
+        //ALOGD("getPosition: %u %lld", numFramesPlayed, (long long)numFramesPlayedAtUs);
+    }
+
+    // CHECK_EQ(numFramesPlayed & (1 << 31), 0);  // can't be negative until 12.4 hrs, test
+    // TODO: remove the (int32_t) casting below as it may overflow at 12.4 hours.
+    int64_t durationUs = (int64_t)((int32_t)numFramesPlayed * 1000000LL / mSampleRateHz)
+            + nowUs - numFramesPlayedAtUs;
+    if (durationUs < 0) {
+        // Occurs when numFramesPlayed position is very small and the following:
+        // (1) In case 1, the time nowUs is computed before getTimestamp() is called and
+        //     numFramesPlayedAtUs is greater than nowUs by time more than numFramesPlayed.
+        // (2) In case 3, using getPosition and adding mAudioSink->latency() to
+        //     numFramesPlayedAtUs, by a time amount greater than numFramesPlayed.
+        //
+        // Both of these are transitory conditions.
+        ALOGV("getPlayedOutDurationUs: negative duration %lld set to zero", (long long)durationUs);
+        durationUs = 0;
+    }
+    ALOGV("getPlayedOutDurationUs(%lld) nowUs(%lld) frames(%u) framesAt(%lld)",
+            (long long)durationUs, (long long)nowUs,
+            numFramesPlayed, (long long)numFramesPlayedAtUs);
+    return durationUs;
+}
+
+status_t MediaPlayer2AudioOutput::getFramesWritten(uint32_t *frameswritten) const {
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return NO_INIT;
+    }
+    ExtendedTimestamp ets;
+    status_t status = mTrack->getTimestamp(&ets);
+    if (status == OK || status == WOULD_BLOCK) {
+        *frameswritten = (uint32_t)ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT];
+    }
+    return status;
+}
+
+status_t MediaPlayer2AudioOutput::setParameters(const String8& keyValuePairs) {
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return NO_INIT;
+    }
+    return mTrack->setParameters(keyValuePairs);
+}
+
+String8  MediaPlayer2AudioOutput::getParameters(const String8& keys) {
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return String8::empty();
+    }
+    return mTrack->getParameters(keys);
+}
+
+void MediaPlayer2AudioOutput::setAudioAttributes(const audio_attributes_t * attributes) {
+    Mutex::Autolock lock(mLock);
+    if (attributes == NULL) {
+        free(mAttributes);
+        mAttributes = NULL;
+    } else {
+        if (mAttributes == NULL) {
+            mAttributes = (audio_attributes_t *) calloc(1, sizeof(audio_attributes_t));
+        }
+        memcpy(mAttributes, attributes, sizeof(audio_attributes_t));
+        mStreamType = audio_attributes_to_stream_type(attributes);
+    }
+}
+
+void MediaPlayer2AudioOutput::setAudioStreamType(audio_stream_type_t streamType) {
+    Mutex::Autolock lock(mLock);
+    // do not allow direct stream type modification if attributes have been set
+    if (mAttributes == NULL) {
+        mStreamType = streamType;
+    }
+}
+
+void MediaPlayer2AudioOutput::close_l() {
+    mTrack.clear();
+}
+
+status_t MediaPlayer2AudioOutput::open(
+        uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
+        audio_format_t format, int bufferCount,
+        AudioCallback cb, void *cookie,
+        audio_output_flags_t flags,
+        const audio_offload_info_t *offloadInfo,
+        bool doNotReconnect,
+        uint32_t suggestedFrameCount) {
+    ALOGV("open(%u, %d, 0x%x, 0x%x, %d, %d 0x%x)", sampleRate, channelCount, channelMask,
+                format, bufferCount, mSessionId, flags);
+
+    // offloading is only supported in callback mode for now.
+    // offloadInfo must be present if offload flag is set
+    if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) &&
+            ((cb == NULL) || (offloadInfo == NULL))) {
+        return BAD_VALUE;
+    }
+
+    // compute frame count for the AudioTrack internal buffer
+    size_t frameCount;
+    if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+        frameCount = 0; // AudioTrack will get frame count from AudioFlinger
+    } else {
+        // try to estimate the buffer processing fetch size from AudioFlinger.
+        // framesPerBuffer is approximate and generally correct, except when it's not :-).
+        uint32_t afSampleRate;
+        size_t afFrameCount;
+        if (AudioSystem::getOutputFrameCount(&afFrameCount, mStreamType) != NO_ERROR) {
+            return NO_INIT;
+        }
+        if (AudioSystem::getOutputSamplingRate(&afSampleRate, mStreamType) != NO_ERROR) {
+            return NO_INIT;
+        }
+        const size_t framesPerBuffer =
+                (unsigned long long)sampleRate * afFrameCount / afSampleRate;
+
+        if (bufferCount == 0) {
+            // use suggestedFrameCount
+            bufferCount = (suggestedFrameCount + framesPerBuffer - 1) / framesPerBuffer;
+        }
+        // Check argument bufferCount against the mininum buffer count
+        if (bufferCount != 0 && bufferCount < mMinBufferCount) {
+            ALOGV("bufferCount (%d) increased to %d", bufferCount, mMinBufferCount);
+            bufferCount = mMinBufferCount;
+        }
+        // if frameCount is 0, then AudioTrack will get frame count from AudioFlinger
+        // which will be the minimum size permitted.
+        frameCount = bufferCount * framesPerBuffer;
+    }
+
+    if (channelMask == CHANNEL_MASK_USE_CHANNEL_ORDER) {
+        channelMask = audio_channel_out_mask_from_count(channelCount);
+        if (0 == channelMask) {
+            ALOGE("open() error, can\'t derive mask for %d audio channels", channelCount);
+            return NO_INIT;
+        }
+    }
+
+    Mutex::Autolock lock(mLock);
+    mCallback = cb;
+    mCallbackCookie = cookie;
+
+    sp<AudioTrack> t;
+    CallbackData *newcbd = NULL;
+
+    ALOGV("creating new AudioTrack");
+
+    if (mCallback != NULL) {
+        newcbd = new CallbackData(this);
+        t = new AudioTrack(
+                mStreamType,
+                sampleRate,
+                format,
+                channelMask,
+                frameCount,
+                flags,
+                CallbackWrapper,
+                newcbd,
+                0,  // notification frames
+                mSessionId,
+                AudioTrack::TRANSFER_CALLBACK,
+                offloadInfo,
+                mUid,
+                mPid,
+                mAttributes,
+                doNotReconnect,
+                1.0f,  // default value for maxRequiredSpeed
+                mSelectedDeviceId);
+    } else {
+        // TODO: Due to buffer memory concerns, we use a max target playback speed
+        // based on mPlaybackRate at the time of open (instead of kMaxRequiredSpeed),
+        // also clamping the target speed to 1.0 <= targetSpeed <= kMaxRequiredSpeed.
+        const float targetSpeed =
+                std::min(std::max(mPlaybackRate.mSpeed, 1.0f), kMaxRequiredSpeed);
+        ALOGW_IF(targetSpeed != mPlaybackRate.mSpeed,
+                "track target speed:%f clamped from playback speed:%f",
+                targetSpeed, mPlaybackRate.mSpeed);
+        t = new AudioTrack(
+                mStreamType,
+                sampleRate,
+                format,
+                channelMask,
+                frameCount,
+                flags,
+                NULL, // callback
+                NULL, // user data
+                0, // notification frames
+                mSessionId,
+                AudioTrack::TRANSFER_DEFAULT,
+                NULL, // offload info
+                mUid,
+                mPid,
+                mAttributes,
+                doNotReconnect,
+                targetSpeed,
+                mSelectedDeviceId);
+    }
+
+    if ((t == 0) || (t->initCheck() != NO_ERROR)) {
+        ALOGE("Unable to create audio track");
+        delete newcbd;
+        // t goes out of scope, so reference count drops to zero
+        return NO_INIT;
+    } else {
+        // successful AudioTrack initialization implies a legacy stream type was generated
+        // from the audio attributes
+        mStreamType = t->streamType();
+    }
+
+    CHECK((t != NULL) && ((mCallback == NULL) || (newcbd != NULL)));
+
+    mCallbackData = newcbd;
+    ALOGV("setVolume");
+    t->setVolume(mLeftVolume, mRightVolume);
+
+    mSampleRateHz = sampleRate;
+    mFlags = flags;
+    mMsecsPerFrame = 1E3f / (mPlaybackRate.mSpeed * sampleRate);
+    mFrameSize = t->frameSize();
+    mTrack = t;
+
+    return updateTrack_l();
+}
+
+status_t MediaPlayer2AudioOutput::updateTrack_l() {
+    if (mTrack == NULL) {
+        return NO_ERROR;
+    }
+
+    status_t res = NO_ERROR;
+    // Note some output devices may give us a direct track even though we don't specify it.
+    // Example: Line application b/17459982.
+    if ((mTrack->getFlags()
+            & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT)) == 0) {
+        res = mTrack->setPlaybackRate(mPlaybackRate);
+        if (res == NO_ERROR) {
+            mTrack->setAuxEffectSendLevel(mSendLevel);
+            res = mTrack->attachAuxEffect(mAuxEffectId);
+        }
+    }
+    mTrack->setOutputDevice(mSelectedDeviceId);
+    if (mDeviceCallbackEnabled) {
+        mTrack->addAudioDeviceCallback(mDeviceCallback.promote());
+    }
+    ALOGV("updateTrack_l() DONE status %d", res);
+    return res;
+}
+
+status_t MediaPlayer2AudioOutput::start() {
+    ALOGV("start");
+    Mutex::Autolock lock(mLock);
+    if (mCallbackData != NULL) {
+        mCallbackData->endTrackSwitch();
+    }
+    if (mTrack != 0) {
+        mTrack->setVolume(mLeftVolume, mRightVolume);
+        mTrack->setAuxEffectSendLevel(mSendLevel);
+        status_t status = mTrack->start();
+        return status;
+    }
+    return NO_INIT;
+}
+
+ssize_t MediaPlayer2AudioOutput::write(const void* buffer, size_t size, bool blocking) {
+    Mutex::Autolock lock(mLock);
+    LOG_ALWAYS_FATAL_IF(mCallback != NULL, "Don't call write if supplying a callback.");
+
+    //ALOGV("write(%p, %u)", buffer, size);
+    if (mTrack != 0) {
+        return mTrack->write(buffer, size, blocking);
+    }
+    return NO_INIT;
+}
+
+void MediaPlayer2AudioOutput::stop() {
+    ALOGV("stop");
+    Mutex::Autolock lock(mLock);
+    if (mTrack != 0) {
+        mTrack->stop();
+    }
+}
+
+void MediaPlayer2AudioOutput::flush() {
+    ALOGV("flush");
+    Mutex::Autolock lock(mLock);
+    if (mTrack != 0) {
+        mTrack->flush();
+    }
+}
+
+void MediaPlayer2AudioOutput::pause() {
+    ALOGV("pause");
+    Mutex::Autolock lock(mLock);
+    if (mTrack != 0) {
+        mTrack->pause();
+    }
+}
+
+void MediaPlayer2AudioOutput::close() {
+    ALOGV("close");
+    sp<AudioTrack> track;
+    {
+        Mutex::Autolock lock(mLock);
+        track = mTrack;
+        close_l(); // clears mTrack
+    }
+    // destruction of the track occurs outside of mutex.
+}
+
+void MediaPlayer2AudioOutput::setVolume(float left, float right) {
+    ALOGV("setVolume(%f, %f)", left, right);
+    Mutex::Autolock lock(mLock);
+    mLeftVolume = left;
+    mRightVolume = right;
+    if (mTrack != 0) {
+        mTrack->setVolume(left, right);
+    }
+}
+
+status_t MediaPlayer2AudioOutput::setPlaybackRate(const AudioPlaybackRate &rate) {
+    ALOGV("setPlaybackRate(%f %f %d %d)",
+                rate.mSpeed, rate.mPitch, rate.mFallbackMode, rate.mStretchMode);
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        // remember rate so that we can set it when the track is opened
+        mPlaybackRate = rate;
+        return OK;
+    }
+    status_t res = mTrack->setPlaybackRate(rate);
+    if (res != NO_ERROR) {
+        return res;
+    }
+    // rate.mSpeed is always greater than 0 if setPlaybackRate succeeded
+    CHECK_GT(rate.mSpeed, 0.f);
+    mPlaybackRate = rate;
+    if (mSampleRateHz != 0) {
+        mMsecsPerFrame = 1E3f / (rate.mSpeed * mSampleRateHz);
+    }
+    return res;
+}
+
+status_t MediaPlayer2AudioOutput::getPlaybackRate(AudioPlaybackRate *rate) {
+    ALOGV("setPlaybackRate");
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return NO_INIT;
+    }
+    *rate = mTrack->getPlaybackRate();
+    return NO_ERROR;
+}
+
+status_t MediaPlayer2AudioOutput::setAuxEffectSendLevel(float level) {
+    ALOGV("setAuxEffectSendLevel(%f)", level);
+    Mutex::Autolock lock(mLock);
+    mSendLevel = level;
+    if (mTrack != 0) {
+        return mTrack->setAuxEffectSendLevel(level);
+    }
+    return NO_ERROR;
+}
+
+status_t MediaPlayer2AudioOutput::attachAuxEffect(int effectId) {
+    ALOGV("attachAuxEffect(%d)", effectId);
+    Mutex::Autolock lock(mLock);
+    mAuxEffectId = effectId;
+    if (mTrack != 0) {
+        return mTrack->attachAuxEffect(effectId);
+    }
+    return NO_ERROR;
+}
+
+status_t MediaPlayer2AudioOutput::setOutputDevice(audio_port_handle_t deviceId) {
+    ALOGV("setOutputDevice(%d)", deviceId);
+    Mutex::Autolock lock(mLock);
+    mSelectedDeviceId = deviceId;
+    if (mTrack != 0) {
+        return mTrack->setOutputDevice(deviceId);
+    }
+    return NO_ERROR;
+}
+
+status_t MediaPlayer2AudioOutput::getRoutedDeviceId(audio_port_handle_t* deviceId) {
+    ALOGV("getRoutedDeviceId");
+    Mutex::Autolock lock(mLock);
+    if (mTrack != 0) {
+        mRoutedDeviceId = mTrack->getRoutedDeviceId();
+    }
+    *deviceId = mRoutedDeviceId;
+    return NO_ERROR;
+}
+
+status_t MediaPlayer2AudioOutput::enableAudioDeviceCallback(bool enabled) {
+    ALOGV("enableAudioDeviceCallback, %d", enabled);
+    Mutex::Autolock lock(mLock);
+    mDeviceCallbackEnabled = enabled;
+    if (mTrack != 0) {
+        status_t status;
+        if (enabled) {
+            status = mTrack->addAudioDeviceCallback(mDeviceCallback.promote());
+        } else {
+            status = mTrack->removeAudioDeviceCallback(mDeviceCallback.promote());
+        }
+        return status;
+    }
+    return NO_ERROR;
+}
+
+// static
+void MediaPlayer2AudioOutput::CallbackWrapper(
+        int event, void *cookie, void *info) {
+    //ALOGV("callbackwrapper");
+    CallbackData *data = (CallbackData*)cookie;
+    // lock to ensure we aren't caught in the middle of a track switch.
+    data->lock();
+    MediaPlayer2AudioOutput *me = data->getOutput();
+    AudioTrack::Buffer *buffer = (AudioTrack::Buffer *)info;
+    if (me == NULL) {
+        // no output set, likely because the track was scheduled to be reused
+        // by another player, but the format turned out to be incompatible.
+        data->unlock();
+        if (buffer != NULL) {
+            buffer->size = 0;
+        }
+        return;
+    }
+
+    switch(event) {
+    case AudioTrack::EVENT_MORE_DATA: {
+        size_t actualSize = (*me->mCallback)(
+                me, buffer->raw, buffer->size, me->mCallbackCookie,
+                CB_EVENT_FILL_BUFFER);
+
+        // Log when no data is returned from the callback.
+        // (1) We may have no data (especially with network streaming sources).
+        // (2) We may have reached the EOS and the audio track is not stopped yet.
+        // Note that AwesomePlayer/AudioPlayer will only return zero size when it reaches the EOS.
+        // NuPlayer2Renderer will return zero when it doesn't have data (it doesn't block to fill).
+        //
+        // This is a benign busy-wait, with the next data request generated 10 ms or more later;
+        // nevertheless for power reasons, we don't want to see too many of these.
+
+        ALOGV_IF(actualSize == 0 && buffer->size > 0, "callbackwrapper: empty buffer returned");
+
+        buffer->size = actualSize;
+        } break;
+
+    case AudioTrack::EVENT_STREAM_END:
+        // currently only occurs for offloaded callbacks
+        ALOGV("callbackwrapper: deliver EVENT_STREAM_END");
+        (*me->mCallback)(me, NULL /* buffer */, 0 /* size */,
+                me->mCallbackCookie, CB_EVENT_STREAM_END);
+        break;
+
+    case AudioTrack::EVENT_NEW_IAUDIOTRACK :
+        ALOGV("callbackwrapper: deliver EVENT_TEAR_DOWN");
+        (*me->mCallback)(me,  NULL /* buffer */, 0 /* size */,
+                me->mCallbackCookie, CB_EVENT_TEAR_DOWN);
+        break;
+
+    case AudioTrack::EVENT_UNDERRUN:
+        // This occurs when there is no data available, typically
+        // when there is a failure to supply data to the AudioTrack.  It can also
+        // occur in non-offloaded mode when the audio device comes out of standby.
+        //
+        // If an AudioTrack underruns it outputs silence. Since this happens suddenly
+        // it may sound like an audible pop or glitch.
+        //
+        // The underrun event is sent once per track underrun; the condition is reset
+        // when more data is sent to the AudioTrack.
+        ALOGD("callbackwrapper: EVENT_UNDERRUN (discarded)");
+        break;
+
+    default:
+        ALOGE("received unknown event type: %d inside CallbackWrapper !", event);
+    }
+
+    data->unlock();
+}
+
+audio_session_t MediaPlayer2AudioOutput::getSessionId() const
+{
+    Mutex::Autolock lock(mLock);
+    return mSessionId;
+}
+
+uint32_t MediaPlayer2AudioOutput::getSampleRate() const
+{
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return 0;
+    }
+    return mTrack->getSampleRate();
+}
+
+int64_t MediaPlayer2AudioOutput::getBufferDurationInUs() const
+{
+    Mutex::Autolock lock(mLock);
+    if (mTrack == 0) {
+        return 0;
+    }
+    int64_t duration;
+    if (mTrack->getBufferDurationInUs(&duration) != OK) {
+        return 0;
+    }
+    return duration;
+}
+
+} // namespace android