diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 7d5d772..6de6486 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -139,13 +139,15 @@
      *                     latency of the track. The actual size selected by the AudioTrack could be
      *                     larger if the requested size is not compatible with current audio HAL
      *                     latency.
-     * flags:              Reserved for future use.
+     * flags:              See comments on audio_policy_output_flags_t in <system/audio_policy.h>.
      * cbf:                Callback function. If not null, this function is called periodically
      *                     to request new PCM data.
      * user:               Context for use by the callback receiver.
      * notificationFrames: The callback function is called each time notificationFrames PCM
      *                     frames have been consumed from track input buffer.
      * sessionId:          Specific session ID, or zero to use default.
+     * threadCanCallJava:  Whether callbacks are made from an attached thread and thus can call JNI.
+     *                     If not present in parameter list, then fixed at false.
      */
 
                         AudioTrack( audio_stream_type_t streamType,
@@ -157,7 +159,7 @@
                                     callback_t cbf       = NULL,
                                     void* user           = NULL,
                                     int notificationFrames = 0,
-                                    int sessionId = 0);
+                                    int sessionId        = 0);
 
                         // DEPRECATED
                         explicit AudioTrack( int streamType,
@@ -189,7 +191,7 @@
                                     callback_t cbf      = NULL,
                                     void* user          = NULL,
                                     int notificationFrames = 0,
-                                    int sessionId = 0);
+                                    int sessionId       = 0);
 
     /* Terminates the AudioTrack and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioTrack.
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index d4aa233..9a8f4b0 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -87,8 +87,9 @@
         virtual ssize_t     frameSize() const = 0;
         virtual uint32_t    latency() const = 0;
         virtual float       msecsPerFrame() const = 0;
-        virtual status_t    getPosition(uint32_t *position) = 0;
-        virtual int         getSessionId() = 0;
+        virtual status_t    getPosition(uint32_t *position) const = 0;
+        virtual status_t    getFramesWritten(uint32_t *frameswritten) const = 0;
+        virtual int         getSessionId() const = 0;
 
         // If no callback is specified, use the "write" API below to submit
         // audio data.
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index 57cab08..1ba1f44 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -376,12 +376,19 @@
     return mMsecsPerFrame;
 }
 
-status_t VideoEditorPlayer::VeAudioOutput::getPosition(uint32_t *position) {
+status_t VideoEditorPlayer::VeAudioOutput::getPosition(uint32_t *position) const {
 
     if (mTrack == 0) return NO_INIT;
     return mTrack->getPosition(position);
 }
 
+status_t VideoEditorPlayer::VeAudioOutput::getFramesWritten(uint32_t *written) const {
+
+    if (mTrack == 0) return NO_INIT;
+    *written = mNumFramesWritten;
+    return OK;
+}
+
 status_t VideoEditorPlayer::VeAudioOutput::open(
         uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
         audio_format_t format, int bufferCount,
@@ -569,7 +576,7 @@
     return NO_ERROR;
 }
 
-int VideoEditorPlayer::VeAudioOutput::getSessionId() {
+int VideoEditorPlayer::VeAudioOutput::getSessionId() const {
 
     return mSessionId;
 }
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.h b/libvideoeditor/lvpp/VideoEditorPlayer.h
index 6962501..350b384 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.h
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.h
@@ -45,8 +45,9 @@
         virtual ssize_t         frameSize() const;
         virtual uint32_t        latency() const;
         virtual float           msecsPerFrame() const;
-        virtual status_t        getPosition(uint32_t *position);
-        virtual int             getSessionId();
+        virtual status_t        getPosition(uint32_t *position) const;
+        virtual status_t        getFramesWritten(uint32_t*) const;
+        virtual int             getSessionId() const;
 
         virtual status_t        open(
                 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index d73eabd..c619ad7 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -226,7 +226,8 @@
 
     // force direct flag if format is not linear PCM
     if (!audio_is_linear_pcm(format)) {
-        flags = (audio_policy_output_flags_t) (flags | AUDIO_POLICY_OUTPUT_FLAG_DIRECT);
+        flags = (audio_policy_output_flags_t)
+                ((flags | AUDIO_POLICY_OUTPUT_FLAG_DIRECT) & ~AUDIO_POLICY_OUTPUT_FLAG_FAST);
     }
 
     if (!audio_is_output_channel(channelMask)) {
@@ -252,6 +253,7 @@
     mNotificationFramesReq = notificationFrames;
     mSessionId = sessionId;
     mAuxEffectId = 0;
+    mCbf = cbf;
 
     // create the IAudioTrack
     status_t status = createTrack_l(streamType,
@@ -280,7 +282,6 @@
     mSharedBuffer = sharedBuffer;
     mMuted = false;
     mActive = false;
-    mCbf = cbf;
     mUserData = user;
     mLoopCount = 0;
     mMarkerPosition = 0;
@@ -762,6 +763,18 @@
         return NO_INIT;
     }
 
+    // Client decides whether the track is TIMED (see below), but can only express a preference
+    // for FAST.  Server will perform additional tests.
+    if ((flags & AUDIO_POLICY_OUTPUT_FLAG_FAST) && !(
+            // either of these use cases:
+            // use case 1: shared buffer
+            (sharedBuffer != 0) ||
+            // use case 2: callback handler
+            (mCbf != NULL))) {
+        ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied");
+        flags = (audio_policy_output_flags_t) (flags & ~AUDIO_POLICY_OUTPUT_FLAG_FAST);
+    }
+
     mNotificationFramesAct = mNotificationFramesReq;
     if (!audio_is_linear_pcm(format)) {
         if (sharedBuffer != 0) {
@@ -786,7 +799,7 @@
             if (mNotificationFramesAct > (uint32_t)frameCount/2) {
                 mNotificationFramesAct = frameCount/2;
             }
-            if (frameCount < minFrameCount) {
+            if (frameCount < minFrameCount && !(flags & AUDIO_POLICY_OUTPUT_FLAG_FAST)) {
                 // not ALOGW because it happens all the time when playing key clicks over A2DP
                 ALOGV("Minimum buffer size corrected from %d to %d",
                          frameCount, minFrameCount);
@@ -807,6 +820,10 @@
     if (mIsTimed) {
         trackFlags |= IAudioFlinger::TRACK_TIMED;
     }
+    if (flags & AUDIO_POLICY_OUTPUT_FLAG_FAST) {
+        trackFlags |= IAudioFlinger::TRACK_FAST;
+    }
+
     sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
                                                       streamType,
                                                       sampleRate,
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index cc3138d..a977337 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1417,6 +1417,7 @@
     : mCallback(NULL),
       mCallbackCookie(NULL),
       mCallbackData(NULL),
+      mBytesWritten(0),
       mSessionId(sessionId) {
     ALOGV("AudioOutput(%d)", sessionId);
     mTrack = 0;
@@ -1495,12 +1496,19 @@
     return mMsecsPerFrame;
 }
 
-status_t MediaPlayerService::AudioOutput::getPosition(uint32_t *position)
+status_t MediaPlayerService::AudioOutput::getPosition(uint32_t *position) const
 {
     if (mTrack == 0) return NO_INIT;
     return mTrack->getPosition(position);
 }
 
+status_t MediaPlayerService::AudioOutput::getFramesWritten(uint32_t *frameswritten) const
+{
+    if (mTrack == 0) return NO_INIT;
+    *frameswritten = mBytesWritten / frameSize();
+    return OK;
+}
+
 status_t MediaPlayerService::AudioOutput::open(
         uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
         audio_format_t format, int bufferCount,
@@ -1656,6 +1664,7 @@
         mTrack = NULL;
         mNextOutput->mSampleRateHz = mSampleRateHz;
         mNextOutput->mMsecsPerFrame = mMsecsPerFrame;
+        mNextOutput->mBytesWritten = mBytesWritten;
     }
 }
 
@@ -1666,6 +1675,7 @@
     //ALOGV("write(%p, %u)", buffer, size);
     if (mTrack) {
         ssize_t ret = mTrack->write(buffer, size);
+        mBytesWritten += ret;
         return ret;
     }
     return NO_INIT;
@@ -1777,7 +1787,7 @@
     data->unlock();
 }
 
-int MediaPlayerService::AudioOutput::getSessionId()
+int MediaPlayerService::AudioOutput::getSessionId() const
 {
     return mSessionId;
 }
@@ -1802,13 +1812,20 @@
     return mMsecsPerFrame;
 }
 
-status_t MediaPlayerService::AudioCache::getPosition(uint32_t *position)
+status_t MediaPlayerService::AudioCache::getPosition(uint32_t *position) const
 {
     if (position == 0) return BAD_VALUE;
     *position = mSize;
     return NO_ERROR;
 }
 
+status_t MediaPlayerService::AudioCache::getFramesWritten(uint32_t *written) const
+{
+    if (written == 0) return BAD_VALUE;
+    *written = mSize;
+    return NO_ERROR;
+}
+
 ////////////////////////////////////////////////////////////////////////////////
 
 struct CallbackThread : public Thread {
@@ -1971,7 +1988,7 @@
     p->mSignal.signal();
 }
 
-int MediaPlayerService::AudioCache::getSessionId()
+int MediaPlayerService::AudioCache::getSessionId() const
 {
     return 0;
 }
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index b08dd6c..2a8cfd2 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -84,8 +84,9 @@
         virtual ssize_t         frameSize() const;
         virtual uint32_t        latency() const;
         virtual float           msecsPerFrame() const;
-        virtual status_t        getPosition(uint32_t *position);
-        virtual int             getSessionId();
+        virtual status_t        getPosition(uint32_t *position) const;
+        virtual status_t        getFramesWritten(uint32_t *frameswritten) const;
+        virtual int             getSessionId() const;
 
         virtual status_t        open(
                 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
@@ -122,6 +123,7 @@
         AudioCallback           mCallback;
         void *                  mCallbackCookie;
         CallbackData *          mCallbackData;
+        uint64_t                mBytesWritten;
         audio_stream_type_t     mStreamType;
         float                   mLeftVolume;
         float                   mRightVolume;
@@ -181,8 +183,9 @@
         virtual ssize_t         frameSize() const { return ssize_t(mChannelCount * ((mFormat == AUDIO_FORMAT_PCM_16_BIT)?sizeof(int16_t):sizeof(u_int8_t))); }
         virtual uint32_t        latency() const;
         virtual float           msecsPerFrame() const;
-        virtual status_t        getPosition(uint32_t *position);
-        virtual int             getSessionId();
+        virtual status_t        getPosition(uint32_t *position) const;
+        virtual status_t        getFramesWritten(uint32_t *frameswritten) const;
+        virtual int             getSessionId() const;
 
         virtual status_t        open(
                 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index 544d501..11cea3b 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -39,6 +39,7 @@
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MediaErrors.h>
 #include <media/stagefright/MetaData.h>
+#include <media/stagefright/SkipCutBuffer.h>
 #include <gui/ISurfaceTexture.h>
 
 #include "avc_utils.h"
@@ -63,10 +64,13 @@
       mSkipRenderingVideoUntilMediaTimeUs(-1ll),
       mVideoLateByUs(0ll),
       mNumFramesTotal(0ll),
-      mNumFramesDropped(0ll) {
+      mNumFramesDropped(0ll),
+      mSkipCutBuffer(NULL) {
 }
 
 NuPlayer::~NuPlayer() {
+    delete mSkipCutBuffer;
+    mSkipCutBuffer = NULL;
 }
 
 void NuPlayer::setUID(uid_t uid) {
@@ -234,6 +238,32 @@
 
             mSource->start();
 
+            sp<MetaData> meta = mSource->getFormat(true /* audio */);
+            if (meta != NULL) {
+                int32_t delay = 0;
+                if (!meta->findInt32(kKeyEncoderDelay, &delay)) {
+                    delay = 0;
+                }
+                int32_t padding = 0;
+                if (!meta->findInt32(kKeyEncoderPadding, &padding)) {
+                    padding = 0;
+                }
+                int32_t numchannels = 0;
+                if (delay + padding) {
+                    if (meta->findInt32(kKeyChannelCount, &numchannels)) {
+                        size_t frameSize = numchannels * sizeof(int16_t);
+                        if (mSkipCutBuffer) {
+                            size_t prevbuffersize = mSkipCutBuffer->size();
+                            if (prevbuffersize != 0) {
+                                ALOGW("Replacing SkipCutBuffer holding %d bytes", prevbuffersize);
+                            }
+                            delete mSkipCutBuffer;
+                        }
+                        mSkipCutBuffer = new SkipCutBuffer(delay * frameSize, padding * frameSize);
+                    }
+                }
+            }
+
             mRenderer = new Renderer(
                     mAudioSink,
                     new AMessage(kWhatRendererNotify, id()));
@@ -844,6 +874,10 @@
         skipUntilMediaTimeUs = -1;
     }
 
+    if (audio && mSkipCutBuffer) {
+        mSkipCutBuffer->submit(buffer);
+    }
+
     mRenderer->queueBuffer(audio, buffer, reply);
 }
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 25766e0..f917f64 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -27,6 +27,7 @@
 struct ACodec;
 struct MetaData;
 struct NuPlayerDriver;
+class SkipCutBuffer;
 
 struct NuPlayer : public AHandler {
     NuPlayer();
@@ -128,6 +129,8 @@
     int64_t mVideoLateByUs;
     int64_t mNumFramesTotal, mNumFramesDropped;
 
+    SkipCutBuffer *mSkipCutBuffer;
+
     status_t instantiateDecoder(bool audio, sp<Decoder> *decoder);
 
     status_t feedDecoderInputData(bool audio, const sp<AMessage> &msg);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index ecbc428..1f13955 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -591,6 +591,10 @@
 void NuPlayer::Renderer::onAudioSinkChanged() {
     CHECK(!mDrainAudioQueuePending);
     mNumFramesWritten = 0;
+    uint32_t written;
+    if (mAudioSink->getFramesWritten(&written) == OK) {
+        mNumFramesWritten = written;
+    }
 }
 
 void NuPlayer::Renderer::notifyPosition() {
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 83af5f3..fb5a7e1 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1924,7 +1924,7 @@
 }
 
 // this method must always be called either with ThreadBase mLock held or inside the thread loop
-audio_stream_t* AudioFlinger::PlaybackThread::stream()
+audio_stream_t* AudioFlinger::PlaybackThread::stream() const
 {
     if (mOutput == NULL) {
         return NULL;
@@ -1932,7 +1932,7 @@
     return &mOutput->stream->common;
 }
 
-uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
+uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
 {
     // A2DP output latency is not due only to buffering capacity. It also reflects encoding,
     // decoding and transfer time. So sleeping for half of the latency would likely cause
@@ -2817,12 +2817,12 @@
     return NO_ERROR;
 }
 
-uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
+uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
 {
     return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
 }
 
-uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
+uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
 {
     return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
 }
@@ -3166,7 +3166,7 @@
     return reconfig;
 }
 
-uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
+uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
 {
     uint32_t time;
     if (audio_is_linear_pcm(mFormat)) {
@@ -3177,7 +3177,7 @@
     return time;
 }
 
-uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
+uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
 {
     uint32_t time;
     if (audio_is_linear_pcm(mFormat)) {
@@ -3188,7 +3188,7 @@
     return time;
 }
 
-uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
+uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
 {
     uint32_t time;
     if (audio_is_linear_pcm(mFormat)) {
@@ -3351,7 +3351,7 @@
     return true;
 }
 
-uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
+uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
 {
     return (mWaitTimeMs * 1000) / 2;
 }
@@ -5623,7 +5623,7 @@
 }
 
 // this method must always be called either with ThreadBase mLock held or inside the thread loop
-audio_stream_t* AudioFlinger::RecordThread::stream()
+audio_stream_t* AudioFlinger::RecordThread::stream() const
 {
     if (mInput == NULL) {
         return NULL;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 2376aff..3051514 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -472,7 +472,7 @@
                     audio_io_handle_t id() const { return mId;}
                     bool        standby() const { return mStandby; }
                     uint32_t    device() const { return mDevice; }
-        virtual     audio_stream_t* stream() = 0;
+        virtual     audio_stream_t* stream() const = 0;
 
                     sp<EffectHandle> createEffect_l(
                                         const sp<AudioFlinger::Client>& client,
@@ -918,7 +918,7 @@
 
                     AudioStreamOut* getOutput() const;
                     AudioStreamOut* clearOutput();
-                    virtual audio_stream_t* stream();
+                    virtual audio_stream_t* stream() const;
 
                     void        suspend() { mSuspended++; }
                     void        restore() { if (mSuspended > 0) mSuspended--; }
@@ -960,9 +960,13 @@
         // Allocate a track name.  Returns name >= 0 if successful, -1 on failure.
         virtual int             getTrackName_l() = 0;
         virtual void            deleteTrackName_l(int name) = 0;
-        virtual uint32_t        activeSleepTimeUs();
-        virtual uint32_t        idleSleepTimeUs() = 0;
-        virtual uint32_t        suspendSleepTimeUs() = 0;
+
+        // Time to sleep between cycles when:
+        virtual uint32_t        activeSleepTimeUs() const;      // mixer state MIXER_TRACKS_ENABLED
+        virtual uint32_t        idleSleepTimeUs() const = 0;    // mixer state MIXER_IDLE
+        virtual uint32_t        suspendSleepTimeUs() const = 0; // audio policy manager suspended us
+        // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write()
+        // No sleep in standby mode; waits on a condition
 
         // Code snippets that are temporarily lifted up out of threadLoop() until the merge
                     void        checkSilentMode_l();
@@ -1048,8 +1052,8 @@
         virtual     mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
         virtual     int         getTrackName_l();
         virtual     void        deleteTrackName_l(int name);
-        virtual     uint32_t    idleSleepTimeUs();
-        virtual     uint32_t    suspendSleepTimeUs();
+        virtual     uint32_t    idleSleepTimeUs() const;
+        virtual     uint32_t    suspendSleepTimeUs() const;
         virtual     void        cacheParameters_l();
 
         // threadLoop snippets
@@ -1073,9 +1077,9 @@
     protected:
         virtual     int         getTrackName_l();
         virtual     void        deleteTrackName_l(int name);
-        virtual     uint32_t    activeSleepTimeUs();
-        virtual     uint32_t    idleSleepTimeUs();
-        virtual     uint32_t    suspendSleepTimeUs();
+        virtual     uint32_t    activeSleepTimeUs() const;
+        virtual     uint32_t    idleSleepTimeUs() const;
+        virtual     uint32_t    suspendSleepTimeUs() const;
         virtual     void        cacheParameters_l();
 
         // threadLoop snippets
@@ -1110,9 +1114,9 @@
         // Thread virtuals
                     void        addOutputTrack(MixerThread* thread);
                     void        removeOutputTrack(MixerThread* thread);
-                    uint32_t    waitTimeMs() { return mWaitTimeMs; }
+                    uint32_t    waitTimeMs() const { return mWaitTimeMs; }
     protected:
-        virtual     uint32_t    activeSleepTimeUs();
+        virtual     uint32_t    activeSleepTimeUs() const;
 
     private:
                     bool        outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
@@ -1260,7 +1264,7 @@
                 status_t    dump(int fd, const Vector<String16>& args);
                 AudioStreamIn* getInput() const;
                 AudioStreamIn* clearInput();
-                virtual audio_stream_t* stream();
+                virtual audio_stream_t* stream() const;
 
         // AudioBufferProvider interface
         virtual status_t    getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 3f4c19a..0e6ea12 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -66,32 +66,7 @@
     // and mTrackNames is initially 0.  However, leave it here until that's verified.
     track_t* t = mState.tracks;
     for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
-        t->needs = 0;
-        t->volume[0] = UNITY_GAIN;
-        t->volume[1] = UNITY_GAIN;
-        // no initialization needed
-        // t->prevVolume[0]
-        // t->prevVolume[1]
-        t->volumeInc[0] = 0;
-        t->volumeInc[1] = 0;
-        t->auxLevel = 0;
-        t->auxInc = 0;
-        // no initialization needed
-        // t->prevAuxLevel
-        // t->frameCount
-        t->channelCount = 2;
-        t->enabled = false;
-        t->format = 16;
-        t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
-        t->bufferProvider = NULL;
-        t->buffer.raw = NULL;
-        // t->buffer.frameCount
-        t->hook = NULL;
-        t->in = NULL;
-        t->resampler = NULL;
-        t->sampleRate = mSampleRate;
-        t->mainBuffer = NULL;
-        t->auxBuffer = NULL;
+        // FIXME redundant per track
         t->localTimeFreq = lc.getLocalFreq();
         t++;
     }
@@ -115,6 +90,38 @@
         int n = __builtin_ctz(names);
         ALOGV("add track (%d)", n);
         mTrackNames |= 1 << n;
+        // assume default parameters for the track, except where noted below
+        track_t* t = &mState.tracks[n];
+        t->needs = 0;
+        t->volume[0] = UNITY_GAIN;
+        t->volume[1] = UNITY_GAIN;
+        // no initialization needed
+        // t->prevVolume[0]
+        // t->prevVolume[1]
+        t->volumeInc[0] = 0;
+        t->volumeInc[1] = 0;
+        t->auxLevel = 0;
+        t->auxInc = 0;
+        // no initialization needed
+        // t->prevAuxLevel
+        // t->frameCount
+        t->channelCount = 2;
+        t->enabled = false;
+        t->format = 16;
+        t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
+        // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
+        t->bufferProvider = NULL;
+        t->buffer.raw = NULL;
+        // no initialization needed
+        // t->buffer.frameCount
+        t->hook = NULL;
+        t->in = NULL;
+        t->resampler = NULL;
+        t->sampleRate = mSampleRate;
+        // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
+        t->mainBuffer = NULL;
+        t->auxBuffer = NULL;
+        // see t->localTimeFreq in constructor above
         return TRACK0 + n;
     }
     return -1;
@@ -215,6 +222,9 @@
                 invalidateState(1 << name);
             }
             break;
+        case FORMAT:
+            ALOG_ASSERT(valueInt == AUDIO_FORMAT_PCM_16_BIT);
+            break;
         default:
             LOG_FATAL("bad param");
         }