diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index 9c9721a..56896b9 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -52,7 +52,10 @@
 // the combinations that are actually needed within AudioFlinger.  If the list of combinations grows
 // too large, then this decision should be re-visited.
 // Sample rate and channel count are explicit, PCM interleaved 16-bit is assumed.
-typedef unsigned NBAIO_Format;
+struct NBAIO_Format {
+//private:
+    unsigned    mPacked;
+};
 
 extern const NBAIO_Format Format_Invalid;
 
@@ -60,9 +63,11 @@
 size_t Format_frameSize(const NBAIO_Format& format);
 
 // Return the frame size of an NBAIO_Format as a bit shift
-size_t Format_frameBitShift(const NBAIO_Format& format);
+// or -1 if frame size is not a power of 2
+int Format_frameBitShift(const NBAIO_Format& format);
 
 // Convert a sample rate in Hz and channel count to an NBAIO_Format
+// FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
 NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount);
 
 // Return the sample rate in Hz of an NBAIO_Format
@@ -76,6 +81,12 @@
 typedef ssize_t (*readVia_t)(void *user, const void *buffer,
                              size_t count, int64_t readPTS);
 
+// Check whether an NBAIO_Format is valid
+bool Format_isValid(const NBAIO_Format& format);
+
+// Compare two NBAIO_Format values
+bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2);
+
 // Abstract class (interface) representing a data port.
 class NBAIO_Port : public RefBase {
 
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 51dba7b..72a60ea 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -84,13 +84,15 @@
     return DEAD_OBJECT;
 }
 
-status_t AudioSystem::muteMicrophone(bool state) {
+status_t AudioSystem::muteMicrophone(bool state)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
     return af->setMicMute(state);
 }
 
-status_t AudioSystem::isMicrophoneMuted(bool* state) {
+status_t AudioSystem::isMicrophoneMuted(bool* state)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
     *state = af->getMicMute();
@@ -175,13 +177,15 @@
     return af->setMode(mode);
 }
 
-status_t AudioSystem::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) {
+status_t AudioSystem::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
     return af->setParameters(ioHandle, keyValuePairs);
 }
 
-String8 AudioSystem::getParameters(audio_io_handle_t ioHandle, const String8& keys) {
+String8 AudioSystem::getParameters(audio_io_handle_t ioHandle, const String8& keys)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     String8 result = String8("");
     if (af == 0) return result;
@@ -413,7 +417,8 @@
     return af->getRenderPosition(halFrames, dspFrames, output);
 }
 
-uint32_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
+uint32_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     uint32_t result = 0;
     if (af == 0) return result;
@@ -423,20 +428,23 @@
     return result;
 }
 
-int AudioSystem::newAudioSessionId() {
+int AudioSystem::newAudioSessionId()
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return 0;
     return af->newAudioSessionId();
 }
 
-void AudioSystem::acquireAudioSessionId(int audioSession) {
+void AudioSystem::acquireAudioSessionId(int audioSession)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af != 0) {
         af->acquireAudioSessionId(audioSession);
     }
 }
 
-void AudioSystem::releaseAudioSessionId(int audioSession) {
+void AudioSystem::releaseAudioSessionId(int audioSession)
+{
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af != 0) {
         af->releaseAudioSessionId(audioSession);
@@ -445,7 +453,8 @@
 
 // ---------------------------------------------------------------------------
 
-void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who __unused) {
+void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who __unused)
+{
     Mutex::Autolock _l(AudioSystem::gLock);
 
     AudioSystem::gAudioFlinger.clear();
@@ -522,12 +531,14 @@
     }
 }
 
-void AudioSystem::setErrorCallback(audio_error_callback cb) {
+void AudioSystem::setErrorCallback(audio_error_callback cb)
+{
     Mutex::Autolock _l(gLock);
     gAudioErrorCallback = cb;
 }
 
-bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType) {
+bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType)
+{
     switch (streamType) {
     case AUDIO_STREAM_MUSIC:
     case AUDIO_STREAM_VOICE_CALL:
@@ -831,7 +842,8 @@
 
 // ---------------------------------------------------------------------------
 
-void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who __unused) {
+void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who __unused)
+{
     Mutex::Autolock _l(AudioSystem::gLock);
     AudioSystem::gAudioPolicyService.clear();
 
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index 8319cd7..1074da9 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -81,8 +81,14 @@
     {"timelapseqvga", CAMCORDER_QUALITY_TIME_LAPSE_QVGA},
 };
 
+#if LOG_NDEBUG
+#define UNUSED __unused
+#else
+#define UNUSED
+#endif
+
 /*static*/ void
-MediaProfiles::logVideoCodec(const MediaProfiles::VideoCodec& codec)
+MediaProfiles::logVideoCodec(const MediaProfiles::VideoCodec& codec UNUSED)
 {
     ALOGV("video codec:");
     ALOGV("codec = %d", codec.mCodec);
@@ -93,7 +99,7 @@
 }
 
 /*static*/ void
-MediaProfiles::logAudioCodec(const MediaProfiles::AudioCodec& codec)
+MediaProfiles::logAudioCodec(const MediaProfiles::AudioCodec& codec UNUSED)
 {
     ALOGV("audio codec:");
     ALOGV("codec = %d", codec.mCodec);
@@ -103,7 +109,7 @@
 }
 
 /*static*/ void
-MediaProfiles::logVideoEncoderCap(const MediaProfiles::VideoEncoderCap& cap)
+MediaProfiles::logVideoEncoderCap(const MediaProfiles::VideoEncoderCap& cap UNUSED)
 {
     ALOGV("video encoder cap:");
     ALOGV("codec = %d", cap.mCodec);
@@ -114,7 +120,7 @@
 }
 
 /*static*/ void
-MediaProfiles::logAudioEncoderCap(const MediaProfiles::AudioEncoderCap& cap)
+MediaProfiles::logAudioEncoderCap(const MediaProfiles::AudioEncoderCap& cap UNUSED)
 {
     ALOGV("audio encoder cap:");
     ALOGV("codec = %d", cap.mCodec);
@@ -124,21 +130,21 @@
 }
 
 /*static*/ void
-MediaProfiles::logVideoDecoderCap(const MediaProfiles::VideoDecoderCap& cap)
+MediaProfiles::logVideoDecoderCap(const MediaProfiles::VideoDecoderCap& cap UNUSED)
 {
     ALOGV("video decoder cap:");
     ALOGV("codec = %d", cap.mCodec);
 }
 
 /*static*/ void
-MediaProfiles::logAudioDecoderCap(const MediaProfiles::AudioDecoderCap& cap)
+MediaProfiles::logAudioDecoderCap(const MediaProfiles::AudioDecoderCap& cap UNUSED)
 {
     ALOGV("audio codec cap:");
     ALOGV("codec = %d", cap.mCodec);
 }
 
 /*static*/ void
-MediaProfiles::logVideoEditorCap(const MediaProfiles::VideoEditorCap& cap)
+MediaProfiles::logVideoEditorCap(const MediaProfiles::VideoEditorCap& cap UNUSED)
 {
     ALOGV("videoeditor cap:");
     ALOGV("mMaxInputFrameWidth = %d", cap.mMaxInputFrameWidth);
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index 9ed010d..4885b4f 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -737,7 +737,8 @@
                     count = b->size;
                 }
                 memcpy(q, p, count);
-//              ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size, count);
+//              ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size,
+//                      count);
             } else if (mPos < mAudioBufferSize) {
                 count = mAudioBufferSize - mPos;
                 if (count > b->size) {
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 65b28dc..fc355b0 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -211,12 +211,12 @@
         virtual void            flush() {}
         virtual void            pause() {}
         virtual void            close() {}
-                void            setAudioStreamType(audio_stream_type_t streamType) {}
+                void            setAudioStreamType(audio_stream_type_t streamType __unused) {}
                 // stream type is not used for AudioCache
         virtual audio_stream_type_t getAudioStreamType() const { return AUDIO_STREAM_DEFAULT; }
 
-                void            setVolume(float left, float right) {}
-        virtual status_t        setPlaybackRatePermille(int32_t ratePermille) { return INVALID_OPERATION; }
+                void            setVolume(float left __unused, float right __unused) {}
+        virtual status_t        setPlaybackRatePermille(int32_t ratePermille __unused) { return INVALID_OPERATION; }
                 uint32_t        sampleRate() const { return mSampleRate; }
                 audio_format_t  format() const { return mFormat; }
                 size_t          size() const { return mSize; }
diff --git a/media/libmediaplayerservice/nuplayer/mp4/MP4Source.cpp b/media/libmediaplayerservice/nuplayer/mp4/MP4Source.cpp
index d31d947..2aae4dd 100644
--- a/media/libmediaplayerservice/nuplayer/mp4/MP4Source.cpp
+++ b/media/libmediaplayerservice/nuplayer/mp4/MP4Source.cpp
@@ -86,7 +86,7 @@
             total += n;
         }
 
-        ALOGV("read %ld bytes at offset %lld", n, mPosition);
+        ALOGV("read %ld bytes at offset %lld", total, mPosition);
 
         mPosition += total;
 
diff --git a/media/libnbaio/AudioBufferProviderSource.cpp b/media/libnbaio/AudioBufferProviderSource.cpp
index e952a00..4a69104 100644
--- a/media/libnbaio/AudioBufferProviderSource.cpp
+++ b/media/libnbaio/AudioBufferProviderSource.cpp
@@ -28,7 +28,7 @@
     NBAIO_Source(format), mProvider(provider), mConsumed(0)
 {
     ALOG_ASSERT(provider != NULL);
-    ALOG_ASSERT(format != Format_Invalid);
+    ALOG_ASSERT(Format_isValid(format));
 }
 
 AudioBufferProviderSource::~AudioBufferProviderSource()
diff --git a/media/libnbaio/AudioStreamInSource.cpp b/media/libnbaio/AudioStreamInSource.cpp
index 05273f6..ae8fac8 100644
--- a/media/libnbaio/AudioStreamInSource.cpp
+++ b/media/libnbaio/AudioStreamInSource.cpp
@@ -40,7 +40,7 @@
 ssize_t AudioStreamInSource::negotiate(const NBAIO_Format offers[], size_t numOffers,
                                       NBAIO_Format counterOffers[], size_t& numCounterOffers)
 {
-    if (mFormat == Format_Invalid) {
+    if (!Format_isValid(mFormat)) {
         mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common);
         audio_format_t streamFormat = mStream->common.get_format(&mStream->common);
         if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) {
@@ -67,7 +67,7 @@
 
 ssize_t AudioStreamInSource::read(void *buffer, size_t count)
 {
-    if (CC_UNLIKELY(mFormat == Format_Invalid)) {
+    if (CC_UNLIKELY(!Format_isValid(mFormat))) {
         return NEGOTIATE;
     }
     ssize_t bytesRead = mStream->read(mStream, buffer, count << mBitShift);
diff --git a/media/libnbaio/AudioStreamOutSink.cpp b/media/libnbaio/AudioStreamOutSink.cpp
index e4341d7..aa9810e 100644
--- a/media/libnbaio/AudioStreamOutSink.cpp
+++ b/media/libnbaio/AudioStreamOutSink.cpp
@@ -37,7 +37,7 @@
 ssize_t AudioStreamOutSink::negotiate(const NBAIO_Format offers[], size_t numOffers,
                                       NBAIO_Format counterOffers[], size_t& numCounterOffers)
 {
-    if (mFormat == Format_Invalid) {
+    if (!Format_isValid(mFormat)) {
         mStreamBufferSizeBytes = mStream->common.get_buffer_size(&mStream->common);
         audio_format_t streamFormat = mStream->common.get_format(&mStream->common);
         if (streamFormat == AUDIO_FORMAT_PCM_16_BIT) {
@@ -56,7 +56,7 @@
     if (!mNegotiated) {
         return NEGOTIATE;
     }
-    ALOG_ASSERT(mFormat != Format_Invalid);
+    ALOG_ASSERT(Format_isValid(mFormat));
     ssize_t ret = mStream->write(mStream, buffer, count << mBitShift);
     if (ret > 0) {
         ret >>= mBitShift;
diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp
index 8a8155d..51514de 100644
--- a/media/libnbaio/NBAIO.cpp
+++ b/media/libnbaio/NBAIO.cpp
@@ -24,13 +24,16 @@
 
 size_t Format_frameSize(const NBAIO_Format& format)
 {
+    // FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
     return Format_channelCount(format) * sizeof(short);
 }
 
-size_t Format_frameBitShift(const NBAIO_Format& format)
+int Format_frameBitShift(const NBAIO_Format& format)
 {
+    // FIXME The sample format is hard-coded to AUDIO_FORMAT_PCM_16_BIT
     // sizeof(short) == 2, so frame size == 1 << channels
     return Format_channelCount(format);
+    // FIXME must return -1 for non-power of 2
 }
 
 const NBAIO_Format Format_Invalid = { 0 };
@@ -55,10 +58,10 @@
 
 unsigned Format_sampleRate(const NBAIO_Format& format)
 {
-    if (format == Format_Invalid) {
+    if (!Format_isValid(format)) {
         return 0;
     }
-    switch (format & Format_SR_Mask) {
+    switch (format.mPacked & Format_SR_Mask) {
     case Format_SR_8000:
         return 8000;
     case Format_SR_11025:
@@ -82,10 +85,10 @@
 
 unsigned Format_channelCount(const NBAIO_Format& format)
 {
-    if (format == Format_Invalid) {
+    if (!Format_isValid(format)) {
         return 0;
     }
-    switch (format & Format_C_Mask) {
+    switch (format.mPacked & Format_C_Mask) {
     case Format_C_1:
         return 1;
     case Format_C_2:
@@ -97,7 +100,7 @@
 
 NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount)
 {
-    NBAIO_Format format;
+    unsigned format;
     switch (sampleRate) {
     case 8000:
         format = Format_SR_8000;
@@ -136,7 +139,9 @@
     default:
         return Format_Invalid;
     }
-    return format;
+    NBAIO_Format ret;
+    ret.mPacked = format;
+    return ret;
 }
 
 // This is a default implementation; it is expected that subclasses will optimize this.
@@ -218,9 +223,9 @@
 {
     ALOGV("negotiate offers=%p numOffers=%u countersOffers=%p numCounterOffers=%u",
             offers, numOffers, counterOffers, numCounterOffers);
-    if (mFormat != Format_Invalid) {
+    if (Format_isValid(mFormat)) {
         for (size_t i = 0; i < numOffers; ++i) {
-            if (offers[i] == mFormat) {
+            if (Format_isEqual(offers[i], mFormat)) {
                 mNegotiated = true;
                 return i;
             }
@@ -235,4 +240,14 @@
     return (ssize_t) NEGOTIATE;
 }
 
+bool Format_isValid(const NBAIO_Format& format)
+{
+    return format.mPacked != Format_Invalid.mPacked;
+}
+
+bool Format_isEqual(const NBAIO_Format& format1, const NBAIO_Format& format2)
+{
+    return format1.mPacked == format2.mPacked;
+}
+
 }   // namespace android
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index ea79885..5c7e741 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -794,7 +794,9 @@
                 dstOffset += pos.nalSize + 4;
             }
 
+#if !LOG_NDEBUG
             ALOGV("accessUnit contains nal types %s", out.c_str());
+#endif
 
             const NALPosition &pos = nals.itemAt(nals.size() - 1);
             size_t nextScan = pos.nalOffset + pos.nalSize;
diff --git a/media/libstagefright/wifi-display/source/TSPacketizer.cpp b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
index c674700..50d317a 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
@@ -565,7 +565,7 @@
             }
         }
 
-        // size_t numPaddingBytes = sizeAvailableForPayload - numBytesOfPayload;
+        size_t numPaddingBytes = sizeAvailableForPayload - numBytesOfPayload;
         ALOGV("packet 1 contains %zd padding bytes and %zd bytes of payload",
               numPaddingBytes, numBytesOfPayload);
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index e15e632..7c59938 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1783,7 +1783,7 @@
                                         popcount(inStream->common.get_channels(&inStream->common)));
         if (!mTeeSinkInputEnabled) {
             kind = TEE_SINK_NO;
-        } else if (format == Format_Invalid) {
+        } else if (!Format_isValid(format)) {
             kind = TEE_SINK_NO;
         } else if (mRecordTeeSink == 0) {
             kind = TEE_SINK_NEW;
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 6a115ff..03a5390 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -60,7 +60,7 @@
 // ----------------------------------------------------------------------------
 
 AudioPolicyService::AudioPolicyService()
-    : BnAudioPolicyService() , mpAudioPolicyDev(NULL) , mpAudioPolicy(NULL)
+    : BnAudioPolicyService(), mpAudioPolicyDev(NULL), mpAudioPolicy(NULL)
 {
     char value[PROPERTY_VALUE_MAX];
     const struct hw_module_t *module;
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 7126e92..28b9d77 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -239,7 +239,7 @@
                 dumpState->mSampleRate = sampleRate;
             }
 
-            if ((format != previousFormat) || (frameCount != previous->mFrameCount)) {
+            if ((!Format_isEqual(format, previousFormat)) || (frameCount != previous->mFrameCount)) {
                 // FIXME to avoid priority inversion, don't delete here
                 delete mixer;
                 mixer = NULL;
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 4b6c74d..6600161 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -94,6 +94,8 @@
     bool isReady() const;
     void setPaused() { mState = PAUSED; }
     void reset();
+    bool isFlushPending() const { return mFlushHwPending; }
+    void flushAck();
 
     bool isOutputTrack() const {
         return (mStreamType == AUDIO_STREAM_CNT);
@@ -155,6 +157,7 @@
     bool                mIsInvalid; // non-resettable latch, set by invalidate()
     AudioTrackServerProxy*  mAudioTrackServerProxy;
     bool                mResumeToStopping; // track was paused in stopping state.
+    bool                mFlushHwPending; // track requests for thread flush
 };  // end of Track
 
 class TimedTrack : public Track {
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 179c458..410656a 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1490,9 +1490,7 @@
         status = NO_ERROR;
     }
 
-    ALOGV("signal playback thread");
-    broadcast_l();
-
+    onAddNewTrack_l();
     return status;
 }
 
@@ -2362,20 +2360,20 @@
                         (mMixerStatus == MIXER_DRAIN_ALL)) {
                     threadLoop_drain();
                 }
-if (mType == MIXER) {
-                // write blocked detection
-                nsecs_t now = systemTime();
-                nsecs_t delta = now - mLastWriteTime;
-                if (!mStandby && delta > maxPeriod) {
-                    mNumDelayedWrites++;
-                    if ((now - lastWarning) > kWarningThrottleNs) {
-                        ATRACE_NAME("underrun");
-                        ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
-                                ns2ms(delta), mNumDelayedWrites, this);
-                        lastWarning = now;
+                if (mType == MIXER) {
+                    // write blocked detection
+                    nsecs_t now = systemTime();
+                    nsecs_t delta = now - mLastWriteTime;
+                    if (!mStandby && delta > maxPeriod) {
+                        mNumDelayedWrites++;
+                        if ((now - lastWarning) > kWarningThrottleNs) {
+                            ATRACE_NAME("underrun");
+                            ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
+                                    ns2ms(delta), mNumDelayedWrites, this);
+                            lastWarning = now;
+                        }
                     }
                 }
-}
 
             } else {
                 usleep(sleepTime);
@@ -2727,12 +2725,6 @@
     PlaybackThread::threadLoop_standby();
 }
 
-// Empty implementation for standard mixer
-// Overridden for offloaded playback
-void AudioFlinger::PlaybackThread::flushOutput_l()
-{
-}
-
 bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
 {
     return false;
@@ -2764,6 +2756,12 @@
     }
 }
 
+void AudioFlinger::PlaybackThread::onAddNewTrack_l()
+{
+    ALOGV("signal playback thread");
+    broadcast_l();
+}
+
 void AudioFlinger::MixerThread::threadLoop_mix()
 {
     // obtain the presentation timestamp of the next output buffer
@@ -4003,6 +4001,17 @@
         sp<Track> l = mLatestActiveTrack.promote();
         bool last = l.get() == track;
 
+        if (track->isInvalid()) {
+            ALOGW("An invalidated track shouldn't be in active list");
+            tracksToRemove->add(track);
+            continue;
+        }
+
+        if (track->mState == TrackBase::IDLE) {
+            ALOGW("An idle track shouldn't be in active list");
+            continue;
+        }
+
         if (track->isPausing()) {
             track->setPaused();
             if (last) {
@@ -4021,6 +4030,11 @@
                 mBytesRemaining = 0;    // stop writing
             }
             tracksToRemove->add(track);
+        } else if (track->isFlushPending()) {
+            track->flushAck();
+            if (last) {
+                mFlushPending = true;
+            }
         } else if (track->framesReady() && track->isReady() &&
                 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
             ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer);
@@ -4070,7 +4084,6 @@
                         // seek when resuming.
                         if (previousTrack->sessionId() != track->sessionId()) {
                             previousTrack->invalidate();
-                            mFlushPending = true;
                         }
                     }
                 }
@@ -4146,9 +4159,6 @@
     // if resume is received before pause is executed.
     if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
         mOutput->stream->pause(mOutput->stream);
-        if (!doHwPause) {
-            doHwResume = true;
-        }
     }
     if (mFlushPending) {
         flushHw_l();
@@ -4164,11 +4174,6 @@
     return mixerStatus;
 }
 
-void AudioFlinger::OffloadThread::flushOutput_l()
-{
-    mFlushPending = true;
-}
-
 // must be called with thread mutex locked
 bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
 {
@@ -4209,6 +4214,8 @@
     mBytesRemaining = 0;
     mPausedWriteLength = 0;
     mPausedBytesRemaining = 0;
+    mHwPaused = false;
+
     if (mUseAsyncWrite) {
         // discard any pending drain or write ack by incrementing sequence
         mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
@@ -4219,6 +4226,18 @@
     }
 }
 
+void AudioFlinger::OffloadThread::onAddNewTrack_l()
+{
+    sp<Track> previousTrack = mPreviousTrack.promote();
+    sp<Track> latestTrack = mLatestActiveTrack.promote();
+
+    if (previousTrack != 0 && latestTrack != 0 &&
+        (previousTrack->sessionId() != latestTrack->sessionId())) {
+        mFlushPending = true;
+    }
+    PlaybackThread::onAddNewTrack_l();
+}
+
 // ----------------------------------------------------------------------------
 
 AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
@@ -4612,8 +4631,7 @@
                             readInto = mRsmpInBuffer;
                             mRsmpInIndex = 0;
                         }
-                        mBytesRead = mInput->stream->read(mInput->stream, readInto,
-                                mBufferSize);
+                        mBytesRead = mInput->stream->read(mInput->stream, readInto, mBufferSize);
                         if (mBytesRead <= 0) {
                             // TODO: verify that it's benign to use a stale track state
                             if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE))
@@ -4800,6 +4818,7 @@
         ALOGE("createRecordTrack_l() audio driver not initialized");
         goto Exit;
     }
+
     // client expresses a preference for FAST, but we get the final say
     if (*flags & IAudioFlinger::TRACK_FAST) {
       if (
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 4700c0a..f5b5efd 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -307,12 +307,12 @@
                 Vector<ConfigEvent *>     mConfigEvents;
 
                 // These fields are written and read by thread itself without lock or barrier,
-                // and read by other threads without lock or barrier via standby() , outDevice()
+                // and read by other threads without lock or barrier via standby(), outDevice()
                 // and inDevice().
                 // Because of the absence of a lock or barrier, any other thread that reads
                 // these fields must use the information in isolation, or be prepared to deal
                 // with possibility that it might be inconsistent with other information.
-                bool                    mStandby;   // Whether thread is currently in standby.
+                bool                    mStandby;     // Whether thread is currently in standby.
                 audio_devices_t         mOutDevice;   // output device
                 audio_devices_t         mInDevice;    // input device
                 audio_source_t          mAudioSource; // (see audio.h, audio_source_t)
@@ -394,7 +394,7 @@
     virtual     bool        waitingAsyncCallback();
     virtual     bool        waitingAsyncCallback_l();
     virtual     bool        shouldStandby_l();
-
+    virtual     void        onAddNewTrack_l();
 
     // ThreadBase virtuals
     virtual     void        preExit();
@@ -631,7 +631,6 @@
 protected:
                 // accessed by both binder threads and within threadLoop(), lock on mutex needed
                 unsigned    mFastTrackAvailMask;    // bit i set if fast track [i] is available
-    virtual     void        flushOutput_l();
 
 private:
     // timestamp latch:
@@ -750,11 +749,11 @@
     // threadLoop snippets
     virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
     virtual     void        threadLoop_exit();
-    virtual     void        flushOutput_l();
 
     virtual     bool        waitingAsyncCallback();
     virtual     bool        waitingAsyncCallback_l();
     virtual     bool        shouldStandby_l();
+    virtual     void        onAddNewTrack_l();
 
 private:
                 void        flushHw_l();
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index ecedca9..d1c6a06 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -146,7 +146,7 @@
 #ifdef TEE_SINK
         if (mTeeSinkTrackEnabled) {
             NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
-            if (pipeFormat != Format_Invalid) {
+            if (Format_isValid(pipeFormat)) {
                 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
                 size_t numCounterOffers = 0;
                 const NBAIO_Format offers[1] = {pipeFormat};
@@ -347,7 +347,8 @@
     mCachedVolume(1.0),
     mIsInvalid(false),
     mAudioTrackServerProxy(NULL),
-    mResumeToStopping(false)
+    mResumeToStopping(false),
+    mFlushHwPending(false)
 {
     if (mCblk != NULL) {
         if (sharedBuffer == 0) {
@@ -731,6 +732,7 @@
                 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload;
             }
 
+            mFlushHwPending = true;
             mResumeToStopping = false;
         } else {
             if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
@@ -751,11 +753,19 @@
         // Prevent flush being lost if the track is flushed and then resumed
         // before mixer thread can run. This is important when offloading
         // because the hardware buffer could hold a large amount of audio
-        playbackThread->flushOutput_l();
         playbackThread->broadcast_l();
     }
 }
 
+// must be called with thread lock held
+void AudioFlinger::PlaybackThread::Track::flushAck()
+{
+    if (!isOffloaded())
+        return;
+
+    mFlushHwPending = false;
+}
+
 void AudioFlinger::PlaybackThread::Track::reset()
 {
     // Do not reset twice to avoid discarding data written just after a flush and before
@@ -991,7 +1001,8 @@
             size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId,
-            int uid) {
+            int uid)
+{
     if (!client->reserveTimedTrack())
         return 0;