diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index f379ee5..f6646ab 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -661,7 +661,7 @@
     sp<AudioTrackThread>    mAudioTrackThread;
     float                   mVolume[2];
     float                   mSendLevel;
-    uint32_t                mSampleRate;
+    mutable uint32_t        mSampleRate;            // mutable because getSampleRate() can update it.
     size_t                  mFrameCount;            // corresponds to current IAudioTrack
     size_t                  mReqFrameCount;         // frame count to request the next time a new
                                                     // IAudioTrack is needed
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index cc244f0..26d8729 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -100,6 +100,7 @@
         virtual status_t    getFramesWritten(uint32_t *frameswritten) const = 0;
         virtual int         getSessionId() const = 0;
         virtual audio_stream_type_t getAudioStreamType() const = 0;
+        virtual uint32_t    getSampleRate() const = 0;
 
         // If no callback is specified, use the "write" API below to submit
         // audio data.
diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 912a43c..14afb85 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -129,7 +129,7 @@
     void reset();
 
     uint32_t getNumFramesPendingPlayout() const;
-    int64_t getOutputPlayPositionUs_l() const;
+    int64_t getOutputPlayPositionUs_l();
 
     bool allowDeepBuffering() const { return (mCreateFlags & ALLOW_DEEP_BUFFERING) != 0; }
     bool useOffload() const { return (mCreateFlags & USE_OFFLOAD) != 0; }
diff --git a/include/media/stagefright/MetaData.h b/include/media/stagefright/MetaData.h
index de3fc36..3a87474 100644
--- a/include/media/stagefright/MetaData.h
+++ b/include/media/stagefright/MetaData.h
@@ -134,6 +134,7 @@
     kKeyRequiresSecureBuffers = 'secu',  // bool (int32_t)
 
     kKeyIsADTS            = 'adts',  // bool (int32_t)
+    kKeyAACAOT            = 'aaot',  // int32_t
 
     // If a MediaBuffer's data represents (at least partially) encrypted
     // data, the following fields aid in decryption.
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index 5aeba4f..8d656c4 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -585,4 +585,11 @@
     return mSessionId;
 }
 
+uint32_t VideoEditorPlayer::VeAudioOutput::getSampleRate() const {
+    if (mMsecsPerFrame == 0) {
+        return 0;
+    }
+    return (uint32_t)(1.e3 / mMsecsPerFrame);
+}
+
 }  // namespace android
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.h b/libvideoeditor/lvpp/VideoEditorPlayer.h
index 5862c08..b8c1254 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.h
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.h
@@ -48,6 +48,7 @@
         virtual status_t        getPosition(uint32_t *position) const;
         virtual status_t        getFramesWritten(uint32_t*) const;
         virtual int             getSessionId() const;
+        virtual uint32_t        getSampleRate() const;
 
         virtual status_t        open(
                 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 666fafa..ccbc5a3 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -545,13 +545,13 @@
     }
 
     const struct timespec *requested;
+    struct timespec timeout;
     if (waitCount == -1) {
         requested = &ClientProxy::kForever;
     } else if (waitCount == 0) {
         requested = &ClientProxy::kNonBlocking;
     } else if (waitCount > 0) {
         long long ms = WAIT_PERIOD_MS * (long long) waitCount;
-        struct timespec timeout;
         timeout.tv_sec = ms / 1000;
         timeout.tv_nsec = (int) (ms % 1000) * 1000000;
         requested = &timeout;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 7c4a990..11b0b89 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -603,6 +603,19 @@
     }
 
     AutoMutex lock(mLock);
+
+    // sample rate can be updated during playback by the offloaded decoder so we need to
+    // query the HAL and update if needed.
+// FIXME use Proxy return channel to update the rate from server and avoid polling here
+    if (isOffloaded()) {
+        if (mOutput != 0) {
+            uint32_t sampleRate = 0;
+            status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate);
+            if (status == NO_ERROR) {
+                mSampleRate = sampleRate;
+            }
+        }
+    }
     return mSampleRate;
 }
 
@@ -866,7 +879,8 @@
     ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
 
     // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
-    //  n = 1   fast track; nBuffering is ignored
+    //  n = 1   fast track with single buffering; nBuffering is ignored
+    //  n = 2   fast track with double buffering
     //  n = 2   normal track, no sample rate conversion
     //  n = 3   normal track, with sample rate conversion
     //          (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)
@@ -1006,9 +1020,11 @@
             ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
             mAwaitBoost = true;
             if (sharedBuffer == 0) {
-                // double-buffering is not required for fast tracks, due to tighter scheduling
-                if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) {
-                    mNotificationFramesAct = frameCount;
+                // Theoretically double-buffering is not required for fast tracks,
+                // due to tighter scheduling.  But in practice, to accommodate kernels with
+                // scheduling jitter, and apps with computation jitter, we use double-buffering.
+                if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
+                    mNotificationFramesAct = frameCount/nBuffering;
                 }
             }
         } else {
@@ -1091,13 +1107,13 @@
     }
 
     const struct timespec *requested;
+    struct timespec timeout;
     if (waitCount == -1) {
         requested = &ClientProxy::kForever;
     } else if (waitCount == 0) {
         requested = &ClientProxy::kNonBlocking;
     } else if (waitCount > 0) {
         long long ms = WAIT_PERIOD_MS * (long long) waitCount;
-        struct timespec timeout;
         timeout.tv_sec = ms / 1000;
         timeout.tv_nsec = (int) (ms % 1000) * 1000000;
         requested = &timeout;
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index cd052e6..9ac9105 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1813,6 +1813,12 @@
     return mSessionId;
 }
 
+uint32_t MediaPlayerService::AudioOutput::getSampleRate() const
+{
+    if (mTrack == 0) return 0;
+    return mTrack->getSampleRate();
+}
+
 #undef LOG_TAG
 #define LOG_TAG "AudioCache"
 MediaPlayerService::AudioCache::AudioCache(const sp<IMemoryHeap>& heap) :
@@ -2015,6 +2021,14 @@
     return 0;
 }
 
+uint32_t MediaPlayerService::AudioCache::getSampleRate() const
+{
+    if (mMsecsPerFrame == 0) {
+        return 0;
+    }
+    return (uint32_t)(1.e3 / mMsecsPerFrame);
+}
+
 void MediaPlayerService::addBatteryData(uint32_t params)
 {
     Mutex::Autolock lock(mLock);
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index a486cb5..9c084e1 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -86,6 +86,7 @@
         virtual status_t        getPosition(uint32_t *position) const;
         virtual status_t        getFramesWritten(uint32_t *frameswritten) const;
         virtual int             getSessionId() const;
+        virtual uint32_t        getSampleRate() const;
 
         virtual status_t        open(
                 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
@@ -195,6 +196,7 @@
         virtual status_t        getPosition(uint32_t *position) const;
         virtual status_t        getFramesWritten(uint32_t *frameswritten) const;
         virtual int             getSessionId() const;
+        virtual uint32_t        getSampleRate() const;
 
         virtual status_t        open(
                 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index a8a8786..05ee34e 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -721,16 +721,27 @@
     return result + diffUs;
 }
 
-int64_t AudioPlayer::getOutputPlayPositionUs_l() const
+int64_t AudioPlayer::getOutputPlayPositionUs_l()
 {
     uint32_t playedSamples = 0;
+    uint32_t sampleRate;
     if (mAudioSink != NULL) {
         mAudioSink->getPosition(&playedSamples);
+        sampleRate = mAudioSink->getSampleRate();
     } else {
         mAudioTrack->getPosition(&playedSamples);
+        sampleRate = mAudioTrack->getSampleRate();
+    }
+    if (sampleRate != 0) {
+        mSampleRate = sampleRate;
     }
 
-    const int64_t playedUs = (static_cast<int64_t>(playedSamples) * 1000000 ) / mSampleRate;
+    int64_t playedUs;
+    if (mSampleRate != 0) {
+        playedUs = (static_cast<int64_t>(playedSamples) * 1000000 ) / mSampleRate;
+    } else {
+        playedUs = 0;
+    }
 
     // HAL position is relative to the first buffer we sent at mStartPosUs
     const int64_t renderedDuration = mStartPosUs + playedUs;
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 1ba1c6e..491b4d1 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -2285,6 +2285,11 @@
         return ERROR_MALFORMED;
     }
 
+    static uint32_t kSamplingRate[] = {
+        96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+        16000, 12000, 11025, 8000, 7350
+    };
+
     ABitReader br(csd, csd_size);
     uint32_t objectType = br.getBits(5);
 
@@ -2292,6 +2297,9 @@
         objectType = 32 + br.getBits(6);
     }
 
+    //keep AOT type
+    mLastTrack->meta->setInt32(kKeyAACAOT, objectType);
+
     uint32_t freqIndex = br.getBits(4);
 
     int32_t sampleRate = 0;
@@ -2304,29 +2312,30 @@
         numChannels = br.getBits(4);
     } else {
         numChannels = br.getBits(4);
-        if (objectType == 5) {
-            // SBR specific config per 14496-3 table 1.13
-            freqIndex = br.getBits(4);
-            if (freqIndex == 15) {
-                if (csd_size < 8) {
-                    return ERROR_MALFORMED;
-                }
-                sampleRate = br.getBits(24);
-            }
+
+        if (freqIndex == 13 || freqIndex == 14) {
+            return ERROR_MALFORMED;
         }
 
-        if (sampleRate == 0) {
-            static uint32_t kSamplingRate[] = {
-                96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
-                16000, 12000, 11025, 8000, 7350
-            };
+        sampleRate = kSamplingRate[freqIndex];
+    }
 
-            if (freqIndex == 13 || freqIndex == 14) {
+    if (objectType == 5 || objectType == 29) { // SBR specific config per 14496-3 table 1.13
+        uint32_t extFreqIndex = br.getBits(4);
+        int32_t extSampleRate;
+        if (extFreqIndex == 15) {
+            if (csd_size < 8) {
                 return ERROR_MALFORMED;
             }
-
-            sampleRate = kSamplingRate[freqIndex];
+            extSampleRate = br.getBits(24);
+        } else {
+            if (extFreqIndex == 13 || extFreqIndex == 14) {
+                return ERROR_MALFORMED;
+            }
+            extSampleRate = kSamplingRate[extFreqIndex];
         }
+        //TODO: save the extension sampling rate value in meta data =>
+        //      mLastTrack->meta->setInt32(kKeyExtSampleRate, extSampleRate);
     }
 
     if (numChannels == 0) {
diff --git a/media/libstagefright/TimedEventQueue.cpp b/media/libstagefright/TimedEventQueue.cpp
index 1a9a26b..dedd186 100644
--- a/media/libstagefright/TimedEventQueue.cpp
+++ b/media/libstagefright/TimedEventQueue.cpp
@@ -217,6 +217,7 @@
     for (;;) {
         int64_t now_us = 0;
         sp<Event> event;
+        bool wakeLocked = false;
 
         {
             Mutex::Autolock autoLock(mLock);
@@ -283,26 +284,28 @@
             // removeEventFromQueue_l will return NULL.
             // Otherwise, the QueueItem will be removed
             // from the queue and the referenced event returned.
-            event = removeEventFromQueue_l(eventID);
+            event = removeEventFromQueue_l(eventID, &wakeLocked);
         }
 
         if (event != NULL) {
             // Fire event with the lock NOT held.
             event->fire(this, now_us);
+            if (wakeLocked) {
+                Mutex::Autolock autoLock(mLock);
+                releaseWakeLock_l();
+            }
         }
     }
 }
 
 sp<TimedEventQueue::Event> TimedEventQueue::removeEventFromQueue_l(
-        event_id id) {
+        event_id id, bool *wakeLocked) {
     for (List<QueueItem>::iterator it = mQueue.begin();
          it != mQueue.end(); ++it) {
         if ((*it).event->eventID() == id) {
             sp<Event> event = (*it).event;
             event->setEventID(0);
-            if ((*it).has_wakelock) {
-                releaseWakeLock_l();
-            }
+            *wakeLocked = (*it).has_wakelock;
             mQueue.erase(it);
             return event;
         }
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 9041c21..216a329 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -562,6 +562,17 @@
         return false;
     }
 
+    // check whether it is ELD/LD content -> no offloading
+    // FIXME: this should depend on audio DSP capabilities. mapMimeToAudioFormat() should use the
+    // metadata to refine the AAC format and the audio HAL should only list supported profiles.
+    int32_t aacaot = -1;
+    if (meta->findInt32(kKeyAACAOT, &aacaot)) {
+        if (aacaot == 23 || aacaot == 39 ) {
+            ALOGV("track of type '%s' is ELD/LD content", mime);
+            return false;
+        }
+    }
+
     int32_t srate = -1;
     if (!meta->findInt32(kKeySampleRate, &srate)) {
         ALOGV("track of type '%s' does not publish sample rate", mime);
diff --git a/media/libstagefright/include/TimedEventQueue.h b/media/libstagefright/include/TimedEventQueue.h
index 38a08b1..3e84256 100644
--- a/media/libstagefright/include/TimedEventQueue.h
+++ b/media/libstagefright/include/TimedEventQueue.h
@@ -145,7 +145,7 @@
     static void *ThreadWrapper(void *me);
     void threadEntry();
 
-    sp<Event> removeEventFromQueue_l(event_id id);
+    sp<Event> removeEventFromQueue_l(event_id id, bool *wakeLocked);
 
     void acquireWakeLock_l();
     void releaseWakeLock_l(bool force = false);
diff --git a/media/libstagefright/wifi-display/source/TSPacketizer.cpp b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
index c674700..eeb3700 100644
--- a/media/libstagefright/wifi-display/source/TSPacketizer.cpp
+++ b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
@@ -216,7 +216,7 @@
     uint8_t *ptr = dup->data();
 
     *ptr++ = 0xff;
-    *ptr++ = 0xf1;  // b11110001, ID=0, layer=0, protection_absent=1
+    *ptr++ = 0xf9;  // b11111001, ID=1(MPEG-2), layer=0, protection_absent=1
 
     *ptr++ =
         profile << 6
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index df4e029..07dc6dd 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -1122,10 +1122,6 @@
         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
-        // been enabled for mixing.
-        if (t.in == NULL)
-            enabledTracks &= ~(1<<i);
     }
 
     e0 = enabledTracks;
@@ -1161,6 +1157,13 @@
                     aux = t.auxBuffer + numFrames;
                 }
                 while (outFrames) {
+                    // t.in == NULL can happen if the track was flushed just after having
+                    // been enabled for mixing.
+                   if (t.in == NULL) {
+                        enabledTracks &= ~(1<<i);
+                        e1 &= ~(1<<i);
+                        break;
+                    }
                     size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
                     if (inFrames) {
                         t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames,
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 110e45c..3d657b3 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -135,12 +135,12 @@
 
 // IAudioFlinger::createTrack() reports back to client the total size of shared memory area
 // for the track.  The client then sub-divides this into smaller buffers for its use.
-// Currently the client uses double-buffering by default, but doesn't tell us about that.
-// So for now we just assume that client is double-buffered.
-// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or
-// N-buffering, so AudioFlinger could allocate the right amount of memory.
+// Currently the client uses N-buffering by default, but doesn't tell us about the value of N.
+// So for now we just assume that client is double-buffered for fast tracks.
+// FIXME It would be better for client to tell AudioFlinger the value of N,
+// so AudioFlinger could allocate the right amount of memory.
 // See the client's minBufCount and mNotificationFramesAct calculations for details.
-static const int kFastTrackMultiplier = 1;
+static const int kFastTrackMultiplier = 2;
 
 // ----------------------------------------------------------------------------
 
@@ -1210,7 +1210,7 @@
               (
                 (tid != -1) &&
                 ((frameCount == 0) ||
-                (frameCount >= (mFrameCount * kFastTrackMultiplier)))
+                (frameCount >= mFrameCount))
               )
             ) &&
             // PCM data
@@ -1915,7 +1915,7 @@
     // otherwise use the HAL / AudioStreamOut directly
     } else {
         // Direct output and offload threads
-        size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t);
+        size_t offset = (mCurrentWriteLength - mBytesRemaining);
         if (mUseAsyncWrite) {
             ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
             mWriteAckSequence += 2;
@@ -1926,7 +1926,7 @@
         // FIXME We should have an implementation of timestamps for direct output threads.
         // They are used e.g for multichannel PCM playback over HDMI.
         bytesWritten = mOutput->stream->write(mOutput->stream,
-                                                   mMixBuffer + offset, mBytesRemaining);
+                                                   (char *)mMixBuffer + offset, mBytesRemaining);
         if (mUseAsyncWrite &&
                 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
             // do not wait for async callback in case of error of full write
@@ -3038,15 +3038,8 @@
                 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
             minFrames = desiredFrames;
         }
-        // It's not safe to call framesReady() for a static buffer track, so assume it's ready
-        size_t framesReady;
-        if (track->sharedBuffer() == 0) {
-            framesReady = track->framesReady();
-        } else if (track->isStopped()) {
-            framesReady = 0;
-        } else {
-            framesReady = 1;
-        }
+
+        size_t framesReady = track->framesReady();
         if ((framesReady >= minFrames) && track->isReady() &&
                 !track->isPaused() && !track->isTerminated())
         {
@@ -4694,7 +4687,7 @@
             (
                 (tid != -1) &&
                 ((frameCount == 0) ||
-                (frameCount >= (mFrameCount * kFastTrackMultiplier)))
+                (frameCount >= mFrameCount))
             ) &&
             // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format)
             // mono or stereo