diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index fa49592..d4eea3d 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -8,7 +8,8 @@
     AudioResampler.cpp.arm      \
     AudioResamplerSinc.cpp.arm  \
     AudioResamplerCubic.cpp.arm \
-    AudioPolicyService.cpp
+    AudioPolicyService.cpp      \
+    AudioBufferProvider.cpp
 
 LOCAL_C_INCLUDES := \
     system/media/audio_effects/include
@@ -22,7 +23,8 @@
     libhardware_legacy \
     libeffects \
     libdl \
-    libpowermanager
+    libpowermanager \
+    libaah_timesrv_client
 
 LOCAL_STATIC_LIBRARIES := \
     libcpustats \
diff --git a/services/audioflinger/AudioBufferProvider.cpp b/services/audioflinger/AudioBufferProvider.cpp
new file mode 100644
index 0000000..678fd58
--- /dev/null
+++ b/services/audioflinger/AudioBufferProvider.cpp
@@ -0,0 +1,28 @@
+/*
+ * Copyright (C) 2011 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.
+ */
+
+#undef __STRICT_ANSI__
+#define __STDINT_LIMITS
+#define __STDC_LIMIT_MACROS
+#include <stdint.h>
+
+#include "AudioBufferProvider.h"
+
+namespace android {
+
+const int64_t AudioBufferProvider::kInvalidPTS = INT64_MAX;
+
+}; // namespace android
diff --git a/services/audioflinger/AudioBufferProvider.h b/services/audioflinger/AudioBufferProvider.h
index 81c5c39..62ad6bd 100644
--- a/services/audioflinger/AudioBufferProvider.h
+++ b/services/audioflinger/AudioBufferProvider.h
@@ -38,8 +38,15 @@
     };
 
     virtual ~AudioBufferProvider() {}
-    
-    virtual status_t getNextBuffer(Buffer* buffer) = 0;
+
+    // value representing an invalid presentation timestamp
+    static const int64_t kInvalidPTS;
+
+    // pts is the local time when the next sample yielded by getNextBuffer
+    // will be rendered.
+    // Pass kInvalidPTS if the PTS is unknown or not applicable.
+    virtual status_t getNextBuffer(Buffer* buffer, int64_t pts) = 0;
+
     virtual void releaseBuffer(Buffer* buffer) = 0;
 };
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 69560e5..a0f1e67 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -58,6 +58,9 @@
 #include <powermanager/PowerManager.h>
 // #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
 
+#include <aah_timesrv/cc_helper.h>
+#include <aah_timesrv/local_clock.h>
+
 // ----------------------------------------------------------------------------
 
 
@@ -152,8 +155,9 @@
 
 AudioFlinger::AudioFlinger()
     : BnAudioFlinger(),
-        mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
-        mBtNrecIsOff(false)
+      mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterVolumeSW(1.0f),
+      mMasterVolumeSupportLvl(MVS_NONE), mMasterMute(false), mNextUniqueId(1),
+      mBtNrecIsOff(false)
 {
 }
 
@@ -192,6 +196,33 @@
         return;
     }
 
+    // Determine the level of master volume support the primary audio HAL has,
+    // and set the initial master volume at the same time.
+    float initialVolume = 1.0;
+    mMasterVolumeSupportLvl = MVS_NONE;
+    if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) {
+        AutoMutex lock(mHardwareLock);
+        audio_hw_device_t *dev = mPrimaryHardwareDev;
+
+        mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
+        if ((NULL != dev->get_master_volume) &&
+            (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) {
+            mMasterVolumeSupportLvl = MVS_FULL;
+        } else {
+            mMasterVolumeSupportLvl = MVS_SETONLY;
+            initialVolume = 1.0;
+        }
+
+        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+        if ((NULL == dev->set_master_volume) ||
+            (NO_ERROR != dev->set_master_volume(dev, initialVolume))) {
+            mMasterVolumeSupportLvl = MVS_NONE;
+        }
+        mHardwareStatus = AUDIO_HW_INIT;
+    }
+
+    // Set the mode for each audio HAL, and try to set the initial volume (if
+    // supported) for all of the non-primary audio HALs.
     for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
         audio_hw_device_t *dev = mAudioHwDevs[i];
 
@@ -203,11 +234,22 @@
             mMode = AUDIO_MODE_NORMAL;
             mHardwareStatus = AUDIO_HW_SET_MODE;
             dev->set_mode(dev, mMode);
-            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
-            dev->set_master_volume(dev, 1.0f);
-            mHardwareStatus = AUDIO_HW_IDLE;
+
+            if ((dev != mPrimaryHardwareDev) &&
+                (NULL != dev->set_master_volume)) {
+                mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+                dev->set_master_volume(dev, initialVolume);
+            }
+
+            mHardwareStatus = AUDIO_HW_INIT;
         }
     }
+
+    mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl)
+                    ? initialVolume
+                    : 1.0;
+    mMasterVolume   = initialVolume;
+    mHardwareStatus = AUDIO_HW_IDLE;
 }
 
 status_t AudioFlinger::initCheck() const
@@ -378,6 +420,7 @@
         uint32_t flags,
         const sp<IMemory>& sharedBuffer,
         int output,
+        bool isTimed,
         int *sessionId,
         status_t *status)
 {
@@ -441,7 +484,7 @@
         LOGV("createTrack() lSessionId: %d", lSessionId);
 
         track = thread->createTrack_l(client, streamType, sampleRate, format,
-                channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
+                channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus);
 
         // move effect chain to this output thread if an effect on same session was waiting
         // for a track to be created
@@ -534,20 +577,29 @@
         return PERMISSION_DENIED;
     }
 
+    float swmv = value;
+
     // when hw supports master volume, don't scale in sw mixer
-    { // scope for the lock
-        AutoMutex lock(mHardwareLock);
-        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
-        if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
-            value = 1.0f;
+    if (MVS_NONE != mMasterVolumeSupportLvl) {
+        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
+            AutoMutex lock(mHardwareLock);
+            audio_hw_device_t *dev = mAudioHwDevs[i];
+
+            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+            if (NULL != dev->set_master_volume) {
+                dev->set_master_volume(dev, value);
+            }
+            mHardwareStatus = AUDIO_HW_IDLE;
         }
-        mHardwareStatus = AUDIO_HW_IDLE;
+
+        swmv = 1.0;
     }
 
     Mutex::Autolock _l(mLock);
-    mMasterVolume = value;
+    mMasterVolume   = value;
+    mMasterVolumeSW = swmv;
     for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
-       mPlaybackThreads.valueAt(i)->setMasterVolume(value);
+       mPlaybackThreads.valueAt(i)->setMasterVolume(swmv);
 
     return NO_ERROR;
 }
@@ -635,9 +687,27 @@
 
 float AudioFlinger::masterVolume() const
 {
+    if (MVS_FULL == mMasterVolumeSupportLvl) {
+        float ret_val;
+        AutoMutex lock(mHardwareLock);
+
+        mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
+        assert(NULL != mPrimaryHardwareDev);
+        assert(NULL != mPrimaryHardwareDev->get_master_volume);
+
+        mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val);
+        mHardwareStatus = AUDIO_HW_IDLE;
+        return ret_val;
+    }
+
     return mMasterVolume;
 }
 
+float AudioFlinger::masterVolumeSW() const
+{
+    return mMasterVolumeSW;
+}
+
 bool AudioFlinger::masterMute() const
 {
     return mMasterMute;
@@ -1368,7 +1438,8 @@
 
     readOutputParameters();
 
-    mMasterVolume = mAudioFlinger->masterVolume();
+    mMasterVolume = mAudioFlinger->masterVolumeSW();
+
     mMasterMute = mAudioFlinger->masterMute();
 
     for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
@@ -1479,6 +1550,7 @@
         int frameCount,
         const sp<IMemory>& sharedBuffer,
         int sessionId,
+        bool isTimed,
         status_t *status)
 {
     sp<Track> track;
@@ -1528,9 +1600,14 @@
             }
         }
 
-        track = new Track(this, client, streamType, sampleRate, format,
-                channelMask, frameCount, sharedBuffer, sessionId);
-        if (track->getCblk() == NULL || track->name() < 0) {
+        if (!isTimed) {
+            track = new Track(this, client, streamType, sampleRate, format,
+                    channelMask, frameCount, sharedBuffer, sessionId);
+        } else {
+            track = TimedTrack::create(this, client, streamType, sampleRate, format,
+                    channelMask, frameCount, sharedBuffer, sessionId);
+        }
+        if (track == NULL || track->getCblk() == NULL || track->name() < 0) {
             lStatus = NO_MEMORY;
             goto Exit;
         }
@@ -1854,6 +1931,10 @@
 
     acquireWakeLock();
 
+    LocalClock lc;
+    uint64_t localTimeFreq;
+    localTimeFreq = lc.getLocalFreq();
+
     while (!exitPending())
     {
 #ifdef DEBUG_CPU_USAGE
@@ -1950,8 +2031,16 @@
        }
 
         if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+            // obtain the presentation timestamp of the next output buffer
+            int64_t pts;
+            status_t status = mOutput->stream->get_next_write_timestamp(
+                mOutput->stream, &pts);
+            if (status != NO_ERROR) {
+                pts = AudioBufferProvider::kInvalidPTS;
+            }
+
             // mix buffers...
-            mAudioMixer->process();
+            mAudioMixer->process(pts);
             sleepTime = 0;
             standbyTime = systemTime() + kStandbyTimeInNsecs;
             //TODO: delay standby when effects have a tail
@@ -2066,7 +2155,7 @@
         // 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() &&
+        if (track->framesReady() && track->isReady() &&
                 !track->isPaused() && !track->isTerminated())
         {
             //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
@@ -2722,7 +2811,8 @@
             // output audio to hardware
             while (frameCount) {
                 buffer.frameCount = frameCount;
-                activeTrack->getNextBuffer(&buffer);
+                activeTrack->getNextBuffer(&buffer,
+                                           AudioBufferProvider::kInvalidPTS);
                 if (UNLIKELY(buffer.raw == 0)) {
                     memset(curBuf, 0, frameCount * mFrameSize);
                     break;
@@ -2991,7 +3081,7 @@
         if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
             // mix buffers...
             if (outputsReady(outputTracks)) {
-                mAudioMixer->process();
+                mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
             } else {
                 memset(mMixBuffer, 0, mixBufferSize);
             }
@@ -3386,7 +3476,8 @@
             (int)mAuxBuffer);
 }
 
-status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
+    AudioBufferProvider::Buffer* buffer, int64_t pts)
 {
      audio_track_cblk_t* cblk = this->cblk();
      uint32_t framesReady;
@@ -3427,10 +3518,14 @@
      return NOT_ENOUGH_DATA;
 }
 
+uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{
+    return mCblk->framesReady();
+}
+
 bool AudioFlinger::PlaybackThread::Track::isReady() const {
     if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
 
-    if (mCblk->framesReady() >= mCblk->frameCount ||
+    if (framesReady() >= mCblk->frameCount ||
             (mCblk->flags & CBLK_FORCEREADY_MSK)) {
         mFillingUpStatus = FS_FILLED;
         android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
@@ -3599,6 +3694,393 @@
     mAuxBuffer = buffer;
 }
 
+// timed audio tracks
+
+sp<AudioFlinger::PlaybackThread::TimedTrack>
+AudioFlinger::PlaybackThread::TimedTrack::create(
+            const wp<ThreadBase>& thread,
+            const sp<Client>& client,
+            int streamType,
+            uint32_t sampleRate,
+            uint32_t format,
+            uint32_t channelMask,
+            int frameCount,
+            const sp<IMemory>& sharedBuffer,
+            int sessionId) {
+    if (!client->reserveTimedTrack())
+        return NULL;
+
+    sp<TimedTrack> track = new TimedTrack(
+        thread, client, streamType, sampleRate, format, channelMask, frameCount,
+        sharedBuffer, sessionId);
+
+    if (track == NULL) {
+        client->releaseTimedTrack();
+        return NULL;
+    }
+
+    return track;
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
+            const wp<ThreadBase>& thread,
+            const sp<Client>& client,
+            int streamType,
+            uint32_t sampleRate,
+            uint32_t format,
+            uint32_t channelMask,
+            int frameCount,
+            const sp<IMemory>& sharedBuffer,
+            int sessionId)
+    : Track(thread, client, streamType, sampleRate, format, channelMask,
+            frameCount, sharedBuffer, sessionId),
+      mTimedSilenceBuffer(NULL),
+      mTimedSilenceBufferSize(0),
+      mTimedAudioOutputOnTime(false),
+      mMediaTimeTransformValid(false)
+{
+    LocalClock lc;
+    mLocalTimeFreq = lc.getLocalFreq();
+
+    mLocalTimeToSampleTransform.a_zero = 0;
+    mLocalTimeToSampleTransform.b_zero = 0;
+    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
+    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
+    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
+                            &mLocalTimeToSampleTransform.a_to_b_denom);
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
+    mClient->releaseTimedTrack();
+    delete [] mTimedSilenceBuffer;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
+    size_t size, sp<IMemory>* buffer) {
+
+    Mutex::Autolock _l(mTimedBufferQueueLock);
+
+    trimTimedBufferQueue_l();
+
+    // lazily initialize the shared memory heap for timed buffers
+    if (mTimedMemoryDealer == NULL) {
+        const int kTimedBufferHeapSize = 512 << 10;
+
+        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
+                                              "AudioFlingerTimed");
+        if (mTimedMemoryDealer == NULL)
+            return NO_MEMORY;
+    }
+
+    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
+    if (newBuffer == NULL) {
+        newBuffer = mTimedMemoryDealer->allocate(size);
+        if (newBuffer == NULL)
+            return NO_MEMORY;
+    }
+
+    *buffer = newBuffer;
+    return NO_ERROR;
+}
+
+// caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
+    int64_t mediaTimeNow;
+    {
+        Mutex::Autolock mttLock(mMediaTimeTransformLock);
+        if (!mMediaTimeTransformValid)
+            return;
+
+        int64_t targetTimeNow;
+        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
+            ? CCHelper::getCommonTime(&targetTimeNow)
+            : CCHelper::getLocalTime(&targetTimeNow);
+
+        if (OK != res)
+            return;
+
+        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
+                                                    &mediaTimeNow)) {
+            return;
+        }
+    }
+
+    size_t trimIndex;
+    for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) {
+        if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow)
+            break;
+    }
+
+    if (trimIndex) {
+        mTimedBufferQueue.removeItemsAt(0, trimIndex);
+    }
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
+    const sp<IMemory>& buffer, int64_t pts) {
+
+    {
+        Mutex::Autolock mttLock(mMediaTimeTransformLock);
+        if (!mMediaTimeTransformValid)
+            return INVALID_OPERATION;
+    }
+
+    Mutex::Autolock _l(mTimedBufferQueueLock);
+
+    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
+
+    return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
+    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
+
+    LOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__,
+         xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
+         target);
+
+    if (!(target == TimedAudioTrack::LOCAL_TIME ||
+          target == TimedAudioTrack::COMMON_TIME)) {
+        return BAD_VALUE;
+    }
+
+    Mutex::Autolock lock(mMediaTimeTransformLock);
+    mMediaTimeTransform = xform;
+    mMediaTimeTransformTarget = target;
+    mMediaTimeTransformValid = true;
+
+    return NO_ERROR;
+}
+
+#define min(a, b) ((a) < (b) ? (a) : (b))
+
+// implementation of getNextBuffer for tracks whose buffers have timestamps
+status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
+    AudioBufferProvider::Buffer* buffer, int64_t pts)
+{
+    if (pts == AudioBufferProvider::kInvalidPTS) {
+        buffer->raw = 0;
+        buffer->frameCount = 0;
+        return INVALID_OPERATION;
+    }
+
+    // get ahold of the output stream that these samples will be written to
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread == NULL) {
+        buffer->raw = 0;
+        buffer->frameCount = 0;
+        return INVALID_OPERATION;
+    }
+    PlaybackThread* playbackThread = static_cast<PlaybackThread*>(thread.get());
+
+    Mutex::Autolock _l(mTimedBufferQueueLock);
+
+    while (true) {
+
+        // if we have no timed buffers, then fail
+        if (mTimedBufferQueue.isEmpty()) {
+            buffer->raw = 0;
+            buffer->frameCount = 0;
+            return NOT_ENOUGH_DATA;
+        }
+
+        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+
+        // calculate the PTS of the head of the timed buffer queue expressed in
+        // local time
+        int64_t headLocalPTS;
+        {
+            Mutex::Autolock mttLock(mMediaTimeTransformLock);
+
+            assert(mMediaTimeTransformValid);
+
+            if (mMediaTimeTransform.a_to_b_denom == 0) {
+                // the transform represents a pause, so yield silence
+                timedYieldSilence(buffer->frameCount, buffer);
+                return NO_ERROR;
+            }
+
+            int64_t transformedPTS;
+            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
+                                                        &transformedPTS)) {
+                // the transform failed.  this shouldn't happen, but if it does
+                // then just drop this buffer
+                LOGW("timedGetNextBuffer transform failed");
+                buffer->raw = 0;
+                buffer->frameCount = 0;
+                mTimedBufferQueue.removeAt(0);
+                return NO_ERROR;
+            }
+
+            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
+                if (OK != CCHelper::commonTimeToLocalTime(transformedPTS,
+                                                          &headLocalPTS)) {
+                    buffer->raw = 0;
+                    buffer->frameCount = 0;
+                    return INVALID_OPERATION;
+                }
+            } else {
+                headLocalPTS = transformedPTS;
+            }
+        }
+
+        // adjust the head buffer's PTS to reflect the portion of the head buffer
+        // that has already been consumed
+        int64_t effectivePTS = headLocalPTS +
+                ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate());
+
+        // Calculate the delta in samples between the head of the input buffer
+        // queue and the start of the next output buffer that will be written.
+        // If the transformation fails because of over or underflow, it means
+        // that the sample's position in the output stream is so far out of
+        // whack that it should just be dropped.
+        int64_t sampleDelta;
+        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
+            LOGV("*** head buffer is too far from PTS: dropped buffer");
+            mTimedBufferQueue.removeAt(0);
+            continue;
+        }
+        if (!mLocalTimeToSampleTransform.doForwardTransform(
+                (effectivePTS - pts) << 32, &sampleDelta)) {
+            LOGV("*** too late during sample rate transform: dropped buffer");
+            mTimedBufferQueue.removeAt(0);
+            continue;
+        }
+
+        LOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]",
+             __PRETTY_FUNCTION__, head.pts(), head.position(), pts,
+             static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)),
+             static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
+
+        // if the delta between the ideal placement for the next input sample and
+        // the current output position is within this threshold, then we will
+        // concatenate the next input samples to the previous output
+        const int64_t kSampleContinuityThreshold =
+                (static_cast<int64_t>(sampleRate()) << 32) / 10;
+
+        // if this is the first buffer of audio that we're emitting from this track
+        // then it should be almost exactly on time.
+        const int64_t kSampleStartupThreshold = 1LL << 32;
+
+        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
+            (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
+            // the next input is close enough to being on time, so concatenate it
+            // with the last output
+            timedYieldSamples(buffer);
+
+            LOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
+            return NO_ERROR;
+        } else if (sampleDelta > 0) {
+            // the gap between the current output position and the proper start of
+            // the next input sample is too big, so fill it with silence
+            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
+
+            timedYieldSilence(framesUntilNextInput, buffer);
+            LOGV("*** silence: frameCount=%u", buffer->frameCount);
+            return NO_ERROR;
+        } else {
+            // the next input sample is late
+            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
+            size_t onTimeSamplePosition =
+                    head.position() + lateFrames * mCblk->frameSize;
+
+            if (onTimeSamplePosition > head.buffer()->size()) {
+                // all the remaining samples in the head are too late, so
+                // drop it and move on
+                LOGV("*** too late: dropped buffer");
+                mTimedBufferQueue.removeAt(0);
+                continue;
+            } else {
+                // skip over the late samples
+                head.setPosition(onTimeSamplePosition);
+
+                // yield the available samples
+                timedYieldSamples(buffer);
+
+                LOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
+                return NO_ERROR;
+            }
+        }
+    }
+}
+
+// Yield samples from the timed buffer queue head up to the given output
+// buffer's capacity.
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples(
+    AudioBufferProvider::Buffer* buffer) {
+
+    const TimedBuffer& head = mTimedBufferQueue[0];
+
+    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
+                   head.position());
+
+    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
+                                 mCblk->frameSize);
+    size_t framesRequested = buffer->frameCount;
+    buffer->frameCount = min(framesLeftInHead, framesRequested);
+
+    mTimedAudioOutputOnTime = true;
+}
+
+// Yield samples of silence up to the given output buffer's capacity
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence(
+    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
+
+    // lazily allocate a buffer filled with silence
+    if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) {
+        delete [] mTimedSilenceBuffer;
+        mTimedSilenceBufferSize = numFrames * mCblk->frameSize;
+        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
+        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
+    }
+
+    buffer->raw = mTimedSilenceBuffer;
+    size_t framesRequested = buffer->frameCount;
+    buffer->frameCount = min(numFrames, framesRequested);
+
+    mTimedAudioOutputOnTime = false;
+}
+
+void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
+    AudioBufferProvider::Buffer* buffer) {
+
+    Mutex::Autolock _l(mTimedBufferQueueLock);
+
+    if (buffer->raw != mTimedSilenceBuffer) {
+        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+        head.setPosition(head.position() + buffer->frameCount * mCblk->frameSize);
+        if (static_cast<size_t>(head.position()) >= head.buffer()->size()) {
+            mTimedBufferQueue.removeAt(0);
+        }
+    }
+
+    buffer->raw = 0;
+    buffer->frameCount = 0;
+}
+
+uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
+    Mutex::Autolock _l(mTimedBufferQueueLock);
+
+    uint32_t frames = 0;
+    for (size_t i = 0; i < mTimedBufferQueue.size(); i++) {
+        const TimedBuffer& tb = mTimedBufferQueue[i];
+        frames += (tb.buffer()->size() - tb.position())  / mCblk->frameSize;
+    }
+
+    return frames;
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
+        : mPTS(0), mPosition(0) {}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
+    const sp<IMemory>& buffer, int64_t pts)
+        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
+
 // ----------------------------------------------------------------------------
 
 // RecordTrack constructor must be called with AudioFlinger::mLock held
@@ -3635,7 +4117,7 @@
     }
 }
 
-status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
 {
     audio_track_cblk_t* cblk = this->cblk();
     uint32_t framesAvail;
@@ -3957,7 +4439,8 @@
     :   RefBase(),
         mAudioFlinger(audioFlinger),
         mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
-        mPid(pid)
+        mPid(pid),
+        mTimedTrackCount(0)
 {
     // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
 }
@@ -3973,6 +4456,31 @@
     return mMemoryDealer;
 }
 
+// Reserve one of the limited slots for a timed audio track associated
+// with this client
+bool AudioFlinger::Client::reserveTimedTrack()
+{
+    const int kMaxTimedTracksPerClient = 4;
+
+    Mutex::Autolock _l(mTimedTrackLock);
+
+    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
+        LOGW("can not create timed track - pid %d has exceeded the limit",
+             mPid);
+        return false;
+    }
+
+    mTimedTrackCount++;
+    return true;
+}
+
+// Release a slot for a timed audio track
+void AudioFlinger::Client::releaseTimedTrack()
+{
+    Mutex::Autolock _l(mTimedTrackLock);
+    mTimedTrackCount--;
+}
+
 // ----------------------------------------------------------------------------
 
 AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
@@ -4044,6 +4552,38 @@
     return mTrack->attachAuxEffect(EffectId);
 }
 
+status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
+                                                         sp<IMemory>* buffer) {
+    if (!mTrack->isTimedTrack())
+        return INVALID_OPERATION;
+
+    PlaybackThread::TimedTrack* tt =
+            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+    return tt->allocateTimedBuffer(size, buffer);
+}
+
+status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
+                                                     int64_t pts) {
+    if (!mTrack->isTimedTrack())
+        return INVALID_OPERATION;
+
+    PlaybackThread::TimedTrack* tt =
+            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+    return tt->queueTimedBuffer(buffer, pts);
+}
+
+status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
+    const LinearTransform& xform, int target) {
+
+    if (!mTrack->isTimedTrack())
+        return INVALID_OPERATION;
+
+    PlaybackThread::TimedTrack* tt =
+            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+    return tt->setMediaTimeTransform(
+        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
+}
+
 status_t AudioFlinger::TrackHandle::onTransact(
     uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
 {
@@ -4281,7 +4821,8 @@
             }
 
             buffer.frameCount = mFrameCount;
-            if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+            if (LIKELY(mActiveTrack->getNextBuffer(
+                    &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) {
                 size_t framesOut = buffer.frameCount;
                 if (mResampler == 0) {
                     // no resampling
@@ -4560,7 +5101,7 @@
     return NO_ERROR;
 }
 
-status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
 {
     size_t framesReq = buffer->frameCount;
     size_t framesReady = mFrameCount - mRsmpInIndex;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 4b794ef..6bac595 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -84,6 +84,7 @@
                                 uint32_t flags,
                                 const sp<IMemory>& sharedBuffer,
                                 int output,
+                                bool isTimed,
                                 int *sessionId,
                                 status_t *status);
 
@@ -97,6 +98,7 @@
     virtual     status_t    setMasterMute(bool muted);
 
     virtual     float       masterVolume() const;
+    virtual     float       masterVolumeSW() const;
     virtual     bool        masterMute() const;
 
     virtual     status_t    setStreamVolume(int stream, float value, int output);
@@ -188,6 +190,7 @@
         AUDIO_HW_SET_MIC_MUTE,
         AUDIO_SET_VOICE_VOLUME,
         AUDIO_SET_PARAMETER,
+        AUDIO_HW_GET_MASTER_VOLUME,
     };
 
     // record interface
@@ -235,12 +238,18 @@
         pid_t               pid() const { return mPid; }
         sp<AudioFlinger>    audioFlinger() { return mAudioFlinger; }
 
+        bool reserveTimedTrack();
+        void releaseTimedTrack();
+
     private:
                             Client(const Client&);
                             Client& operator = (const Client&);
         sp<AudioFlinger>    mAudioFlinger;
         sp<MemoryDealer>    mMemoryDealer;
         pid_t               mPid;
+
+        Mutex               mTimedTrackLock;
+        int                 mTimedTrackCount;
     };
 
     // --- Notification Client ---
@@ -346,7 +355,9 @@
                                 TrackBase(const TrackBase&);
                                 TrackBase& operator = (const TrackBase&);
 
-            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
+            virtual status_t getNextBuffer(
+                AudioBufferProvider::Buffer* buffer,
+                int64_t pts) = 0;
             virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
 
             uint32_t format() const {
@@ -611,7 +622,6 @@
                     int16_t     *mainBuffer() { return mMainBuffer; }
                     int         auxEffectId() { return mAuxEffectId; }
 
-
         protected:
             friend class ThreadBase;
             friend class TrackHandle;
@@ -622,7 +632,11 @@
                                 Track(const Track&);
                                 Track& operator = (const Track&);
 
-            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+            virtual status_t getNextBuffer(
+                AudioBufferProvider::Buffer* buffer,
+                int64_t pts);
+            virtual uint32_t framesReady() const;
+
             bool isMuted() { return mMute; }
             bool isPausing() const {
                 return mState == PAUSING;
@@ -638,6 +652,8 @@
                 return (mStreamType == AUDIO_STREAM_CNT);
             }
 
+            virtual bool isTimedTrack() const { return false; }
+
             // we don't really need a lock for these
             float               mVolume[2];
             volatile bool       mMute;
@@ -655,6 +671,78 @@
             bool                mHasVolumeController;
         };  // end of Track
 
+        class TimedTrack : public Track {
+          public:
+            static sp<TimedTrack> create(const wp<ThreadBase>& thread,
+                                         const sp<Client>& client,
+                                         int streamType,
+                                         uint32_t sampleRate,
+                                         uint32_t format,
+                                         uint32_t channelMask,
+                                         int frameCount,
+                                         const sp<IMemory>& sharedBuffer,
+                                         int sessionId);
+            ~TimedTrack();
+
+            class TimedBuffer {
+              public:
+                TimedBuffer();
+                TimedBuffer(const sp<IMemory>& buffer, int64_t pts);
+                const sp<IMemory>& buffer() const { return mBuffer; }
+                int64_t pts() const { return mPTS; }
+                int position() const { return mPosition; }
+                void setPosition(int pos) { mPosition = pos; }
+              private:
+                sp<IMemory> mBuffer;
+                int64_t mPTS;
+                int mPosition;
+            };
+
+            virtual bool isTimedTrack() const { return true; }
+
+            virtual uint32_t framesReady() const;
+
+            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
+                                           int64_t pts);
+            virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+            void timedYieldSamples(AudioBufferProvider::Buffer* buffer);
+            void timedYieldSilence(uint32_t numFrames,
+                                   AudioBufferProvider::Buffer* buffer);
+
+            status_t    allocateTimedBuffer(size_t size,
+                                            sp<IMemory>* buffer);
+            status_t    queueTimedBuffer(const sp<IMemory>& buffer,
+                                         int64_t pts);
+            status_t    setMediaTimeTransform(const LinearTransform& xform,
+                                              TimedAudioTrack::TargetTimeline target);
+            void        trimTimedBufferQueue_l();
+
+          private:
+            TimedTrack(const wp<ThreadBase>& thread,
+                       const sp<Client>& client,
+                       int streamType,
+                       uint32_t sampleRate,
+                       uint32_t format,
+                       uint32_t channelMask,
+                       int frameCount,
+                       const sp<IMemory>& sharedBuffer,
+                       int sessionId);
+
+            uint64_t            mLocalTimeFreq;
+            LinearTransform     mLocalTimeToSampleTransform;
+            sp<MemoryDealer>    mTimedMemoryDealer;
+            Vector<TimedBuffer> mTimedBufferQueue;
+            uint8_t*            mTimedSilenceBuffer;
+            uint32_t            mTimedSilenceBufferSize;
+            mutable Mutex       mTimedBufferQueueLock;
+            bool                mTimedAudioOutputOnTime;
+
+            Mutex               mMediaTimeTransformLock;
+            LinearTransform     mMediaTimeTransform;
+            bool                mMediaTimeTransformValid;
+            TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget;
+        };
+
 
         // playback track
         class OutputTrack : public Track {
@@ -728,6 +816,7 @@
                                     int frameCount,
                                     const sp<IMemory>& sharedBuffer,
                                     int sessionId,
+                                    bool isTimed,
                                     status_t *status);
 
                     AudioStreamOut* getOutput();
@@ -916,6 +1005,12 @@
         virtual void        setVolume(float left, float right);
         virtual sp<IMemory> getCblk() const;
         virtual status_t    attachAuxEffect(int effectId);
+        virtual status_t    allocateTimedBuffer(size_t size,
+                                                sp<IMemory>* buffer);
+        virtual status_t    queueTimedBuffer(const sp<IMemory>& buffer,
+                                             int64_t pts);
+        virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
+                                                  int target);
         virtual status_t onTransact(
             uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
     private:
@@ -963,7 +1058,9 @@
                                 RecordTrack(const RecordTrack&);
                                 RecordTrack& operator = (const RecordTrack&);
 
-            virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+            virtual status_t getNextBuffer(
+                AudioBufferProvider::Buffer* buffer,
+                int64_t pts);
 
             bool                mOverflow;
         };
@@ -999,7 +1096,8 @@
                 AudioStreamIn* clearInput();
                 virtual audio_stream_t* stream();
 
-        virtual status_t    getNextBuffer(AudioBufferProvider::Buffer* buffer);
+        virtual status_t    getNextBuffer(AudioBufferProvider::Buffer* buffer,
+                                          int64_t pts);
         virtual void        releaseBuffer(AudioBufferProvider::Buffer* buffer);
         virtual bool        checkForNewParameters_l();
         virtual String8     getParameters(const String8& keys);
@@ -1376,6 +1474,12 @@
     friend class RecordThread;
     friend class PlaybackThread;
 
+    enum master_volume_support {
+        MVS_NONE,
+        MVS_SETONLY,
+        MVS_FULL,
+    };
+
     mutable     Mutex                               mLock;
 
                 DefaultKeyedVector< pid_t, wp<Client> >     mClients;
@@ -1389,6 +1493,8 @@
                 DefaultKeyedVector< int, sp<PlaybackThread> >  mPlaybackThreads;
                 PlaybackThread::stream_type_t       mStreamTypes[AUDIO_STREAM_CNT];
                 float                               mMasterVolume;
+                float                               mMasterVolumeSW;
+                master_volume_support               mMasterVolumeSupportLvl;
                 bool                                mMasterMute;
 
                 DefaultKeyedVector< int, sp<RecordThread> >    mRecordThreads;
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 6e9319d..760981b 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -18,6 +18,7 @@
 #define LOG_TAG "AudioMixer"
 //#define LOG_NDEBUG 0
 
+#include <assert.h>
 #include <stdint.h>
 #include <string.h>
 #include <stdlib.h>
@@ -30,6 +31,9 @@
 
 #include <system/audio.h>
 
+#include <aah_timesrv/local_clock.h>
+#include <aah_timesrv/cc_helper.h>
+
 #include "AudioMixer.h"
 
 namespace android {
@@ -47,6 +51,9 @@
 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
     :   mActiveTrack(0), mTrackNames(0), mSampleRate(sampleRate)
 {
+    LocalClock lc;
+    mLocalTimeFreq = lc.getLocalFreq();
+
     mState.enabledTracks= 0;
     mState.needsChanged = 0;
     mState.frameCount   = frameCount;
@@ -74,6 +81,7 @@
         t->in = 0;
         t->mainBuffer = NULL;
         t->auxBuffer = NULL;
+        t->localTimeFreq = mLocalTimeFreq;
         t++;
     }
 }
@@ -293,6 +301,7 @@
             if (resampler == 0) {
                 resampler = AudioResampler::create(
                         format, channelCount, devSampleRate);
+                resampler->setLocalTimeFreq(localTimeFreq);
             }
             return true;
         }
@@ -340,13 +349,13 @@
 
 
 
-void AudioMixer::process()
+void AudioMixer::process(int64_t pts)
 {
-    mState.hook(&mState);
+    mState.hook(&mState, pts);
 }
 
 
-void AudioMixer::process__validate(state_t* state)
+void AudioMixer::process__validate(state_t* state, int64_t pts)
 {
     LOGW_IF(!state->needsChanged,
         "in process__validate() but nothing's invalid");
@@ -450,7 +459,7 @@
         countActiveTracks, state->enabledTracks,
         all16BitsStereoNoResample, resampling, volumeRamp);
 
-   state->hook(state);
+   state->hook(state, pts);
 
    // Now that the volume ramp has been done, set optimal state and
    // track hooks for subsequent mixer process
@@ -859,7 +868,7 @@
 }
 
 // no-op case
-void AudioMixer::process__nop(state_t* state)
+void AudioMixer::process__nop(state_t* state, int64_t pts)
 {
     uint32_t e0 = state->enabledTracks;
     size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
@@ -889,7 +898,9 @@
             size_t outFrames = state->frameCount;
             while (outFrames) {
                 t1.buffer.frameCount = outFrames;
-                t1.bufferProvider->getNextBuffer(&t1.buffer);
+                int64_t outputPTS = calculateOutputPTS(
+                    t1, pts, state->frameCount - outFrames);
+                t1.bufferProvider->getNextBuffer(&t1.buffer, outputPTS);
                 if (!t1.buffer.raw) break;
                 outFrames -= t1.buffer.frameCount;
                 t1.bufferProvider->releaseBuffer(&t1.buffer);
@@ -899,7 +910,7 @@
 }
 
 // generic code without resampling
-void AudioMixer::process__genericNoResampling(state_t* state)
+void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
 {
     int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
 
@@ -911,7 +922,7 @@
         e0 &= ~(1<<i);
         track_t& t = state->tracks[i];
         t.buffer.frameCount = state->frameCount;
-        t.bufferProvider->getNextBuffer(&t.buffer);
+        t.bufferProvider->getNextBuffer(&t.buffer, pts);
         t.frameCount = t.buffer.frameCount;
         t.in = t.buffer.raw;
         // t.in == NULL can happen if the track was flushed just after having
@@ -965,7 +976,9 @@
                     if (t.frameCount == 0 && outFrames) {
                         t.bufferProvider->releaseBuffer(&t.buffer);
                         t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames);
-                        t.bufferProvider->getNextBuffer(&t.buffer);
+                        int64_t outputPTS = calculateOutputPTS(
+                            t, pts, numFrames + (BLOCKSIZE - outFrames));
+                        t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
                         t.in = t.buffer.raw;
                         if (t.in == NULL) {
                             enabledTracks &= ~(1<<i);
@@ -994,7 +1007,7 @@
 
 
   // generic code with resampling
-void AudioMixer::process__genericResampling(state_t* state)
+void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
 {
     int32_t* const outTemp = state->outputTemp;
     const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
@@ -1033,6 +1046,7 @@
             // acquire/release the buffers because it's done by
             // the resampler.
             if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
+                t.resampler->setPTS(pts);
                 (t.hook)(&t, outTemp, numFrames, state->resampleTemp, aux);
             } else {
 
@@ -1040,7 +1054,8 @@
 
                 while (outFrames < numFrames) {
                     t.buffer.frameCount = numFrames - outFrames;
-                    t.bufferProvider->getNextBuffer(&t.buffer);
+                    int64_t outputPTS = calculateOutputPTS(t, pts, outFrames);
+                    t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
                     t.in = t.buffer.raw;
                     // t.in == NULL can happen if the track was flushed just after having
                     // been enabled for mixing.
@@ -1060,7 +1075,8 @@
 }
 
 // one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
+void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
+                                                           int64_t pts)
 {
     const int i = 31 - __builtin_clz(state->enabledTracks);
     const track_t& t = state->tracks[i];
@@ -1075,7 +1091,8 @@
     const uint32_t vrl = t.volumeRL;
     while (numFrames) {
         b.frameCount = numFrames;
-        t.bufferProvider->getNextBuffer(&b);
+        int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer);
+        t.bufferProvider->getNextBuffer(&b, outputPTS);
         int16_t const *in = b.i16;
 
         // in == NULL can happen if the track was flushed just after having
@@ -1118,7 +1135,8 @@
 // 2 tracks is also a common case
 // NEVER used in current implementation of process__validate()
 // only use if the 2 tracks have the same output buffer
-void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
+void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state,
+                                                            int64_t pts)
 {
     int i;
     uint32_t en = state->enabledTracks;
@@ -1152,7 +1170,9 @@
 
         if (frameCount0 == 0) {
             b0.frameCount = numFrames;
-            t0.bufferProvider->getNextBuffer(&b0);
+            int64_t outputPTS = calculateOutputPTS(t0, pts,
+                                                   out - t0.mainBuffer);
+            t0.bufferProvider->getNextBuffer(&b0, outputPTS);
             if (b0.i16 == NULL) {
                 if (buff == NULL) {
                     buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
@@ -1166,7 +1186,9 @@
         }
         if (frameCount1 == 0) {
             b1.frameCount = numFrames;
-            t1.bufferProvider->getNextBuffer(&b1);
+            int64_t outputPTS = calculateOutputPTS(t1, pts,
+                                                   out - t0.mainBuffer);
+            t1.bufferProvider->getNextBuffer(&b1, outputPTS);
             if (b1.i16 == NULL) {
                 if (buff == NULL) {
                     buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
@@ -1213,6 +1235,11 @@
     }
 }
 
+int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
+                                       int outputFrameIndex)
+{
+    return basePTS + ((outputFrameIndex * t.localTimeFreq) / t.sampleRate);
+}
+
 // ----------------------------------------------------------------------------
 }; // namespace android
-
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 75c9170..d32a366 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -85,7 +85,7 @@
     status_t    setParameter(int target, int name, void *value);
 
     status_t    setBufferProvider(AudioBufferProvider* bufferProvider);
-    void        process();
+    void        process(int64_t pts);
 
     uint32_t    trackNames() const { return mTrackNames; }
 
@@ -125,7 +125,7 @@
     struct state_t;
     struct track_t;
 
-    typedef void (*mix_t)(state_t* state);
+    typedef void (*mix_t)(state_t* state, int64_t pts);
     typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
     static const int BLOCKSIZE = 16; // 4 cache lines
 
@@ -163,6 +163,8 @@
         int32_t*           mainBuffer;
         int32_t*           auxBuffer;
 
+        uint64_t    localTimeFreq;
+
         bool        setResampler(uint32_t sampleRate, uint32_t devSampleRate);
         bool        doesResample() const;
         void        resetResampler();
@@ -185,6 +187,8 @@
     uint32_t        mTrackNames;
     const uint32_t  mSampleRate;
 
+    int64_t         mLocalTimeFreq;
+
     state_t         mState __attribute__((aligned(32)));
 
     void invalidateState(uint32_t mask);
@@ -196,12 +200,17 @@
     static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
     static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
 
-    static void process__validate(state_t* state);
-    static void process__nop(state_t* state);
-    static void process__genericNoResampling(state_t* state);
-    static void process__genericResampling(state_t* state);
-    static void process__OneTrack16BitsStereoNoResampling(state_t* state);
-    static void process__TwoTracks16BitsStereoNoResampling(state_t* state);
+    static void process__validate(state_t* state, int64_t pts);
+    static void process__nop(state_t* state, int64_t pts);
+    static void process__genericNoResampling(state_t* state, int64_t pts);
+    static void process__genericResampling(state_t* state, int64_t pts);
+    static void process__OneTrack16BitsStereoNoResampling(state_t* state,
+                                                          int64_t pts);
+    static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
+                                                           int64_t pts);
+
+    static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
+                                      int outputFrameIndex);
 };
 
 // ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 9ee5a30..47f6242 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -118,7 +118,8 @@
         int32_t sampleRate) :
     mBitDepth(bitDepth), mChannelCount(inChannelCount),
             mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
-            mPhaseFraction(0) {
+            mPhaseFraction(0), mLocalTimeFreq(0),
+            mPTS(AudioBufferProvider::kInvalidPTS) {
     // sanity check on format
     if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
         LOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
@@ -152,6 +153,23 @@
     mVolume[1] = right;
 }
 
+void AudioResampler::setLocalTimeFreq(uint64_t freq) {
+    mLocalTimeFreq = freq;
+}
+
+void AudioResampler::setPTS(int64_t pts) {
+    mPTS = pts;
+}
+
+int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) {
+
+    if (mPTS == AudioBufferProvider::kInvalidPTS) {
+        return AudioBufferProvider::kInvalidPTS;
+    } else {
+        return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate);
+    }
+}
+
 void AudioResampler::reset() {
     mInputIndex = 0;
     mPhaseFraction = 0;
@@ -198,7 +216,8 @@
         // buffer is empty, fetch a new one
         while (mBuffer.frameCount == 0) {
             mBuffer.frameCount = inFrameCount;
-            provider->getNextBuffer(&mBuffer);
+            provider->getNextBuffer(&mBuffer,
+                                    calculateOutputPTS(outputIndex / 2));
             if (mBuffer.raw == NULL) {
                 goto resampleStereo16_exit;
             }
@@ -292,7 +311,8 @@
         // buffer is empty, fetch a new one
         while (mBuffer.frameCount == 0) {
             mBuffer.frameCount = inFrameCount;
-            provider->getNextBuffer(&mBuffer);
+            provider->getNextBuffer(&mBuffer,
+                                    calculateOutputPTS(outputIndex / 2));
             if (mBuffer.raw == NULL) {
                 mInputIndex = inputIndex;
                 mPhaseFraction = phaseFraction;
@@ -602,4 +622,3 @@
 // ----------------------------------------------------------------------------
 }
 ; // namespace android
-
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 9f06c1c..da736ab 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -49,6 +49,10 @@
     virtual void init() = 0;
     virtual void setSampleRate(int32_t inSampleRate);
     virtual void setVolume(int16_t left, int16_t right);
+    virtual void setLocalTimeFreq(uint64_t freq);
+
+    // set the PTS of the next buffer output by the resampler
+    virtual void setPTS(int64_t pts);
 
     virtual void resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider) = 0;
@@ -72,6 +76,8 @@
     AudioResampler(const AudioResampler&);
     AudioResampler& operator=(const AudioResampler&);
 
+    int64_t calculateOutputPTS(int outputFrameIndex);
+
     int32_t mBitDepth;
     int32_t mChannelCount;
     int32_t mSampleRate;
@@ -86,6 +92,8 @@
     size_t mInputIndex;
     int32_t mPhaseIncrement;
     uint32_t mPhaseFraction;
+    uint64_t mLocalTimeFreq;
+    int64_t mPTS;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 4d721f6..e864b30 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -65,7 +65,7 @@
     // fetch first buffer
     if (mBuffer.frameCount == 0) {
         mBuffer.frameCount = inFrameCount;
-        provider->getNextBuffer(&mBuffer);
+        provider->getNextBuffer(&mBuffer, mPTS);
         if (mBuffer.raw == NULL)
             return;
         // LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
@@ -95,7 +95,8 @@
                 inputIndex = 0;
                 provider->releaseBuffer(&mBuffer);
                 mBuffer.frameCount = inFrameCount;
-                provider->getNextBuffer(&mBuffer);
+                provider->getNextBuffer(&mBuffer,
+                                        calculateOutputPTS(outputIndex / 2));
                 if (mBuffer.raw == NULL)
                     goto save_state;  // ugly, but efficient
                 in = mBuffer.i16;
@@ -130,7 +131,7 @@
     // fetch first buffer
     if (mBuffer.frameCount == 0) {
         mBuffer.frameCount = inFrameCount;
-        provider->getNextBuffer(&mBuffer);
+        provider->getNextBuffer(&mBuffer, mPTS);
         if (mBuffer.raw == NULL)
             return;
         // LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
@@ -160,7 +161,8 @@
                 inputIndex = 0;
                 provider->releaseBuffer(&mBuffer);
                 mBuffer.frameCount = inFrameCount;
-                provider->getNextBuffer(&mBuffer);
+                provider->getNextBuffer(&mBuffer,
+                                        calculateOutputPTS(outputIndex / 2));
                 if (mBuffer.raw == NULL)
                     goto save_state;  // ugly, but efficient
                 // LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
@@ -181,4 +183,3 @@
 // ----------------------------------------------------------------------------
 }
 ; // namespace android
-
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 9e5e254..6184795 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -204,7 +204,8 @@
         // buffer is empty, fetch a new one
         while (buffer.frameCount == 0) {
             buffer.frameCount = inFrameCount;
-            provider->getNextBuffer(&buffer);
+            provider->getNextBuffer(&buffer,
+                                    calculateOutputPTS(outputIndex / 2));
             if (buffer.raw == NULL) {
                 goto resample_exit;
             }
@@ -355,4 +356,3 @@
 
 // ----------------------------------------------------------------------------
 }; // namespace android
-