diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index b92a8a0..1e0e7f8 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -589,7 +589,7 @@
 }
 
 static void usage(const char *me) {
-    fprintf(stderr, "usage: %s\n", me);
+    fprintf(stderr, "usage: %s [options] [input_filename]\n", me);
     fprintf(stderr, "       -h(elp)\n");
     fprintf(stderr, "       -a(udio)\n");
     fprintf(stderr, "       -n repetitions\n");
@@ -607,8 +607,8 @@
                     "(video only)\n");
     fprintf(stderr, "       -S allocate buffers from a surface\n");
     fprintf(stderr, "       -T allocate buffers from a surface texture\n");
-    fprintf(stderr, "       -d(ump) filename (raw stream data to a file)\n");
-    fprintf(stderr, "       -D(ump) filename (decoded PCM data to a file)\n");
+    fprintf(stderr, "       -d(ump) output_filename (raw stream data to a file)\n");
+    fprintf(stderr, "       -D(ump) output_filename (decoded PCM data to a file)\n");
 }
 
 static void dumpCodecProfiles(const sp<IOMX>& omx, bool queryDecoders) {
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 2672db1..cd7ff92 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -95,7 +95,7 @@
      *  - BAD_VALUE: unsupported configuration
      */
 
-     static status_t getMinFrameCount(int* frameCount,
+     static status_t getMinFrameCount(size_t* frameCount,
                                       uint32_t sampleRate,
                                       audio_format_t format,
                                       audio_channel_mask_t channelMask);
@@ -184,7 +184,7 @@
 
             audio_format_t format() const;
             int         channelCount() const;
-            uint32_t    frameCount() const;
+            size_t      frameCount() const;
             size_t      frameSize() const { return mFrameSize; }
             audio_source_t inputSource() const;
 
@@ -352,7 +352,7 @@
             status_t openRecord_l(uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
+                                size_t frameCount,
                                 audio_io_handle_t input);
             audio_io_handle_t getInput_l();
             status_t restoreRecord_l(audio_track_cblk_t*& cblk);
@@ -375,7 +375,7 @@
     uint32_t                mUpdatePeriod;      // in ms
 
     // constant after constructor or set()
-    uint32_t                mFrameCount;
+    size_t                  mFrameCount;
     audio_format_t          mFormat;
     uint8_t                 mChannelCount;
     size_t                  mFrameSize;         // app-level frame size == AudioFlinger frame size
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 2218fad..126ef12 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -87,32 +87,26 @@
     static float linearToLog(int volume);
     static int logToLinear(float volume);
 
-    static status_t getOutputSamplingRate(int* samplingRate,
+    static status_t getOutputSamplingRate(uint32_t* samplingRate,
             audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
-    static status_t getOutputFrameCount(int* frameCount,
+    static status_t getOutputFrameCount(size_t* frameCount,
             audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
     static status_t getOutputLatency(uint32_t* latency,
             audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
     static status_t getSamplingRate(audio_io_handle_t output,
                                           audio_stream_type_t streamType,
-                                          int* samplingRate);
+                                          uint32_t* samplingRate);
     // returns the number of frames per audio HAL write buffer. Corresponds to
     // audio_stream->get_buffer_size()/audio_stream_frame_size()
     static status_t getFrameCount(audio_io_handle_t output,
                                   audio_stream_type_t stream,
-                                  int* frameCount);
+                                  size_t* frameCount);
     // returns the audio output stream latency in ms. Corresponds to
     // audio_stream_out->get_latency()
     static status_t getLatency(audio_io_handle_t output,
                                audio_stream_type_t stream,
                                uint32_t* latency);
 
-    // DEPRECATED
-    static status_t getOutputSamplingRate(int* samplingRate, int stream = AUDIO_STREAM_DEFAULT);
-
-    // DEPRECATED
-    static status_t getOutputFrameCount(int* frameCount, int stream = AUDIO_STREAM_DEFAULT);
-
     static bool routedToA2dpOutput(audio_stream_type_t streamType);
 
     static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
@@ -129,11 +123,11 @@
     // - BAD_VALUE: invalid parameter
     // NOTE: this feature is not supported on all hardware platforms and it is
     // necessary to check returned status before using the returned values.
-    static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+    static status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
             audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
 
     // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
-    static unsigned int  getInputFramesLost(audio_io_handle_t ioHandle);
+    static size_t getInputFramesLost(audio_io_handle_t ioHandle);
 
     static int newAudioSessionId();
     static void acquireAudioSessionId(int audioSession);
@@ -243,8 +237,8 @@
     static const sp<IAudioPolicyService>& get_audio_policy_service();
 
     // helpers for android.media.AudioManager.getProperty(), see description there for meaning
-    static int32_t getPrimaryOutputSamplingRate();
-    static int32_t getPrimaryOutputFrameCount();
+    static uint32_t getPrimaryOutputSamplingRate();
+    static size_t getPrimaryOutputFrameCount();
 
     // ----------------------------------------------------------------------------
 
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 3504f1f..f1b26b5 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -110,7 +110,7 @@
      *  - NO_INIT: audio server or audio hardware not initialized
      */
 
-     static status_t getMinFrameCount(int* frameCount,
+     static status_t getMinFrameCount(size_t* frameCount,
                                       audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
                                       uint32_t sampleRate = 0);
 
@@ -282,7 +282,9 @@
 
     /* Set sample rate for this track in Hz, mostly used for games' sound effects
      */
-            status_t    setSampleRate(int sampleRate);
+            status_t    setSampleRate(uint32_t sampleRate);
+
+    /* Return current sample rate in Hz, or 0 if unknown */
             uint32_t    getSampleRate() const;
 
     /* Enables looping and sets the start and end points of looping.
@@ -422,6 +424,7 @@
      * After error return:
      *  frameCount  0
      *  size        0
+     *  raw         undefined
      * After successful return:
      *  frameCount  actual number of frames available, <= number requested
      *  size        actual number of bytes available
@@ -491,7 +494,7 @@
                                  uint32_t sampleRate,
                                  audio_format_t format,
                                  audio_channel_mask_t channelMask,
-                                 int frameCount,
+                                 size_t frameCount,
                                  audio_output_flags_t flags,
                                  const sp<IMemory>& sharedBuffer,
                                  audio_io_handle_t output);
@@ -510,8 +513,14 @@
     uint32_t                mFrameCount;
 
     audio_track_cblk_t*     mCblk;                  // re-load after mLock.unlock()
-    void*                   mBuffers;               // starting address of buffers in shared memory
-    audio_format_t          mFormat;
+
+            // Starting address of buffers in shared memory.  If there is a shared buffer, mBuffers
+            // is the value of pointer() for the shared buffer, otherwise mBuffers points
+            // immediately after the control block.  This address is for the mapping within client
+            // address space.  AudioFlinger::TrackBase::mBuffer is for the server address space.
+    void*                   mBuffers;
+
+    audio_format_t          mFormat;                // as requested by client, not forced to 16-bit
     audio_stream_type_t     mStreamType;
     uint8_t                 mChannelCount;
     uint8_t                 mMuted;
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 0aa48c6..9727143 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -61,7 +61,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
+                                size_t frameCount,
                                 track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
@@ -75,7 +75,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
+                                size_t frameCount,
                                 track_flags_t flags,
                                 pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                 int *sessionId,
@@ -157,10 +157,10 @@
 
     virtual status_t setVoiceVolume(float volume) = 0;
 
-    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+    virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
                                     audio_io_handle_t output) const = 0;
 
-    virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
+    virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
 
     virtual int newAudioSessionId() = 0;
 
@@ -192,8 +192,8 @@
     // helpers for android.media.AudioManager.getProperty(), see description there for meaning
     // FIXME move these APIs to AudioPolicy to permit a more accurate implementation
     // that looks on primary device for a stream with fast flag, primary flag, or first one.
-    virtual int32_t getPrimaryOutputSamplingRate() = 0;
-    virtual int32_t getPrimaryOutputFrameCount() = 0;
+    virtual uint32_t getPrimaryOutputSamplingRate() = 0;
+    virtual size_t getPrimaryOutputFrameCount() = 0;
 
 };
 
diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
index 29c8fd9..0529bcd 100644
--- a/include/media/ToneGenerator.h
+++ b/include/media/ToneGenerator.h
@@ -263,7 +263,7 @@
 
     unsigned short mLoopCounter; // Current tone loopback count
 
-    int mSamplingRate;  // AudioFlinger Sampling rate
+    uint32_t mSamplingRate;  // AudioFlinger Sampling rate
     AudioTrack *mpAudioTrack;  // Pointer to audio track used for playback
     Mutex mLock;  // Mutex to control concurent access to ToneGenerator object from audio callback and application API
     Mutex mCbkCondLock; // Mutex associated to mWaitCbkCond
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 3063448..bbc5e26 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -54,9 +54,10 @@
                 uint32_t    serverBase;
 
                 int         mPad1;          // unused, but preserves cache line alignment
+
                 uint32_t    frameCount;
 
-                // Cache line boundary
+                // Cache line boundary (32 bytes)
 
                 uint32_t    loopStart;
                 uint32_t    loopEnd;        // read-only for server, read/write for client
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index fc9fb49..a47fc15 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -406,8 +406,8 @@
     }
     ALOGV("open(%u, %d, %d, %d)", sampleRate, channelCount, format, bufferCount);
     if (mTrack) close();
-    int afSampleRate;
-    int afFrameCount;
+    uint32_t afSampleRate;
+    size_t afFrameCount;
     int frameCount;
 
     if (AudioSystem::getOutputFrameCount(&afFrameCount, mStreamType) !=
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 062f546..0587651 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -36,7 +36,7 @@
 
 // static
 status_t AudioRecord::getMinFrameCount(
-        int* frameCount,
+        size_t* frameCount,
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask)
@@ -54,7 +54,7 @@
     }
 
     if (size == 0) {
-        ALOGE("Unsupported configuration: sampleRate %d, format %d, channelMask %#x",
+        ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
             sampleRate, format, channelMask);
         return BAD_VALUE;
     }
@@ -119,15 +119,21 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        int frameCountInt,
         callback_t cbf,
         void* user,
         int notificationFrames,
         bool threadCanCallJava,
         int sessionId)
 {
+    // FIXME "int" here is legacy and will be replaced by size_t later
+    if (frameCountInt < 0) {
+        ALOGE("Invalid frame count %d", frameCountInt);
+        return BAD_VALUE;
+    }
+    size_t frameCount = frameCountInt;
 
-    ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask,
+    ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
             frameCount);
 
     AutoMutex lock(mLock);
@@ -177,7 +183,7 @@
     }
 
     // validate framecount
-    int minFrameCount = 0;
+    size_t minFrameCount = 0;
     status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask);
     if (status != NO_ERROR) {
         return status;
@@ -260,7 +266,7 @@
     return mChannelCount;
 }
 
-uint32_t AudioRecord::frameCount() const
+size_t AudioRecord::frameCount() const
 {
     return mFrameCount;
 }
@@ -427,7 +433,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         audio_io_handle_t input)
 {
     status_t status;
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 767c452..028e4a3 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -205,12 +205,7 @@
     return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
 }
 
-// DEPRECATED
-status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType) {
-    return getOutputSamplingRate(samplingRate, (audio_stream_type_t)streamType);
-}
-
-status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type_t streamType)
+status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream_type_t streamType)
 {
     audio_io_handle_t output;
 
@@ -228,7 +223,7 @@
 
 status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
                                       audio_stream_type_t streamType,
-                                      int* samplingRate)
+                                      uint32_t* samplingRate)
 {
     OutputDescriptor *outputDesc;
 
@@ -246,18 +241,13 @@
         gLock.unlock();
     }
 
-    ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output,
+    ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %u", streamType, output,
             *samplingRate);
 
     return NO_ERROR;
 }
 
-// DEPRECATED
-status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType) {
-    return getOutputFrameCount(frameCount, (audio_stream_type_t)streamType);
-}
-
-status_t AudioSystem::getOutputFrameCount(int* frameCount, audio_stream_type_t streamType)
+status_t AudioSystem::getOutputFrameCount(size_t* frameCount, audio_stream_type_t streamType)
 {
     audio_io_handle_t output;
 
@@ -275,7 +265,7 @@
 
 status_t AudioSystem::getFrameCount(audio_io_handle_t output,
                                     audio_stream_type_t streamType,
-                                    int* frameCount)
+                                    size_t* frameCount)
 {
     OutputDescriptor *outputDesc;
 
@@ -371,7 +361,7 @@
     return af->setVoiceVolume(value);
 }
 
-status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+status_t AudioSystem::getRenderPosition(size_t *halFrames, size_t *dspFrames,
         audio_stream_type_t stream)
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -384,7 +374,7 @@
     return af->getRenderPosition(halFrames, dspFrames, getOutput(stream));
 }
 
-unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
+size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     unsigned int result = 0;
     if (af == 0) return result;
@@ -452,7 +442,7 @@
 
         OutputDescriptor *outputDesc =  new OutputDescriptor(*desc);
         gOutputs.add(ioHandle, outputDesc);
-        ALOGV("ioConfigChanged() new output samplingRate %d, format %d channels %#x frameCount %d "
+        ALOGV("ioConfigChanged() new output samplingRate %u, format %d channels %#x frameCount %u "
                 "latency %d",
                 outputDesc->samplingRate, outputDesc->format, outputDesc->channels,
                 outputDesc->frameCount, outputDesc->latency);
@@ -476,7 +466,7 @@
         if (param2 == NULL) break;
         desc = (const OutputDescriptor *)param2;
 
-        ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %#x "
+        ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %d channels %#x "
                 "frameCount %d latency %d",
                 ioHandle, desc->samplingRate, desc->format,
                 desc->channels, desc->frameCount, desc->latency);
@@ -750,14 +740,14 @@
     return NO_ERROR;
 }
 
-int32_t AudioSystem::getPrimaryOutputSamplingRate()
+uint32_t AudioSystem::getPrimaryOutputSamplingRate()
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return 0;
     return af->getPrimaryOutputSamplingRate();
 }
 
-int32_t AudioSystem::getPrimaryOutputFrameCount()
+size_t AudioSystem::getPrimaryOutputFrameCount()
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return 0;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 4a4759e..979ee37 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -50,7 +50,7 @@
 
 // static
 status_t AudioTrack::getMinFrameCount(
-        int* frameCount,
+        size_t* frameCount,
         audio_stream_type_t streamType,
         uint32_t sampleRate)
 {
@@ -65,11 +65,11 @@
     //          audio_format_t format
     //          audio_channel_mask_t channelMask
     //          audio_output_flags_t flags
-    int afSampleRate;
+    uint32_t afSampleRate;
     if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
         return NO_INIT;
     }
-    int afFrameCount;
+    size_t afFrameCount;
     if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
         return NO_INIT;
     }
@@ -166,7 +166,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        int frameCountInt,
         audio_output_flags_t flags,
         callback_t cbf,
         void* user,
@@ -175,11 +175,17 @@
         bool threadCanCallJava,
         int sessionId)
 {
+    // FIXME "int" here is legacy and will be replaced by size_t later
+    if (frameCountInt < 0) {
+        ALOGE("Invalid frame count %d", frameCountInt);
+        return BAD_VALUE;
+    }
+    size_t frameCount = frameCountInt;
 
     ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
             sharedBuffer->size());
 
-    ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags);
+    ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags);
 
     AutoMutex lock(mLock);
     if (mAudioTrack != 0) {
@@ -193,7 +199,7 @@
     }
 
     if (sampleRate == 0) {
-        int afSampleRate;
+        uint32_t afSampleRate;
         if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
             return NO_INIT;
         }
@@ -336,7 +342,7 @@
     return mChannelCount;
 }
 
-uint32_t AudioTrack::frameCount() const
+size_t AudioTrack::frameCount() const
 {
     return mCblk->frameCount;
 }
@@ -390,7 +396,7 @@
         }
         if (cblk->flags & CBLK_INVALID) {
             audio_track_cblk_t* temp = cblk;
-            status = restoreTrack_l(temp, true);
+            status = restoreTrack_l(temp, true /*fromStart*/);
             cblk = temp;
         }
         cblk->lock.unlock();
@@ -535,9 +541,9 @@
     }
 }
 
-status_t AudioTrack::setSampleRate(int rate)
+status_t AudioTrack::setSampleRate(uint32_t rate)
 {
-    int afSamplingRate;
+    uint32_t afSamplingRate;
 
     if (mIsTimed) {
         return INVALID_OPERATION;
@@ -547,7 +553,7 @@
         return NO_INIT;
     }
     // Resampler implementation limits input sampling rate to 2 x output sampling rate.
-    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
+    if (rate == 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
 
     AutoMutex lock(mLock);
     mCblk->sampleRate = rate;
@@ -557,7 +563,7 @@
 uint32_t AudioTrack::getSampleRate() const
 {
     if (mIsTimed) {
-        return INVALID_OPERATION;
+        return 0;
     }
 
     AutoMutex lock(mLock);
@@ -730,7 +736,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         audio_output_flags_t flags,
         const sp<IMemory>& sharedBuffer,
         audio_io_handle_t output)
@@ -770,7 +776,7 @@
             // Same comment as below about ignoring frameCount parameter for set()
             frameCount = sharedBuffer->size();
         } else if (frameCount == 0) {
-            int afFrameCount;
+            size_t afFrameCount;
             if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
                 return NO_INIT;
             }
@@ -802,11 +808,11 @@
     } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
 
         // FIXME move these calculations and associated checks to server
-        int afSampleRate;
+        uint32_t afSampleRate;
         if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) {
             return NO_INIT;
         }
-        int afFrameCount;
+        size_t afFrameCount;
         if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) {
             return NO_INIT;
         }
@@ -815,8 +821,8 @@
         uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
         if (minBufCount < 2) minBufCount = 2;
 
-        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
-        ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d"
+        size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+        ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
                 ", afLatency=%d",
                 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
 
@@ -828,7 +834,7 @@
         }
         // Make sure that application is notified with sufficient margin
         // before underrun
-        if (mNotificationFramesAct > (uint32_t)frameCount/2) {
+        if (mNotificationFramesAct > frameCount/2) {
             mNotificationFramesAct = frameCount/2;
         }
         if (frameCount < minFrameCount) {
@@ -858,7 +864,9 @@
     sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
                                                       streamType,
                                                       sampleRate,
-                                                      format,
+                                                      // AudioFlinger only sees 16-bit PCM
+                                                      format == AUDIO_FORMAT_PCM_8_BIT ?
+                                                              AUDIO_FORMAT_PCM_16_BIT : format,
                                                       channelMask,
                                                       frameCount,
                                                       &trackFlags,
@@ -988,7 +996,7 @@
                             android_atomic_or(CBLK_INVALID, &cblk->flags);
 create_new_track:
                             audio_track_cblk_t* temp = cblk;
-                            result = restoreTrack_l(temp, false);
+                            result = restoreTrack_l(temp, false /*fromStart*/);
                             cblk = temp;
                         }
                         if (result != NO_ERROR) {
@@ -1147,7 +1155,7 @@
     if (cblk->flags & CBLK_INVALID) {
         cblk->lock.lock();
         audio_track_cblk_t* temp = cblk;
-        result = restoreTrack_l(temp, false);
+        result = restoreTrack_l(temp, false /*fromStart*/);
         cblk = temp;
         cblk->lock.unlock();
 
@@ -1423,7 +1431,7 @@
     snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat,
             mChannelCount, cblk->frameCount);
     result.append(buffer);
-    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n",
+    snprintf(buffer, 255, "  sample rate(%u), status(%d), muted(%d)\n",
             (cblk == 0) ? 0 : cblk->sampleRate, mStatus, mMuted);
     result.append(buffer);
     snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index bb936ec..79c3361 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -89,7 +89,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
+                                size_t frameCount,
                                 track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
@@ -106,7 +106,7 @@
         data.writeInt32(format);
         data.writeInt32(channelMask);
         data.writeInt32(frameCount);
-        track_flags_t lFlags = flags != NULL ? *flags : TRACK_DEFAULT;
+        track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
         data.writeInt32(lFlags);
         data.writeStrongBinder(sharedBuffer->asBinder());
         data.writeInt32((int32_t) output);
@@ -143,7 +143,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
+                                size_t frameCount,
                                 track_flags_t flags,
                                 pid_t tid,
                                 int *sessionId,
@@ -527,7 +527,7 @@
         return status;
     }
 
-    virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const
+    virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -695,7 +695,7 @@
         return (audio_module_handle_t) reply.readInt32();
     }
 
-    virtual int32_t getPrimaryOutputSamplingRate()
+    virtual uint32_t getPrimaryOutputSamplingRate()
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -703,7 +703,7 @@
         return reply.readInt32();
     }
 
-    virtual int32_t getPrimaryOutputFrameCount()
+    virtual size_t getPrimaryOutputFrameCount()
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -728,7 +728,7 @@
             uint32_t sampleRate = data.readInt32();
             audio_format_t format = (audio_format_t) data.readInt32();
             audio_channel_mask_t channelMask = data.readInt32();
-            size_t bufferCount = data.readInt32();
+            size_t frameCount = data.readInt32();
             track_flags_t flags = (track_flags_t) data.readInt32();
             sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
             audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
@@ -737,7 +737,7 @@
             status_t status;
             sp<IAudioTrack> track = createTrack(pid,
                     (audio_stream_type_t) streamType, sampleRate, format,
-                    channelMask, bufferCount, &flags, buffer, output, tid, &sessionId, &status);
+                    channelMask, frameCount, &flags, buffer, output, tid, &sessionId, &status);
             reply->writeInt32(flags);
             reply->writeInt32(sessionId);
             reply->writeInt32(status);
@@ -751,13 +751,13 @@
             uint32_t sampleRate = data.readInt32();
             audio_format_t format = (audio_format_t) data.readInt32();
             audio_channel_mask_t channelMask = data.readInt32();
-            size_t bufferCount = data.readInt32();
+            size_t frameCount = data.readInt32();
             track_flags_t flags = (track_flags_t) data.readInt32();
             pid_t tid = (pid_t) data.readInt32();
             int sessionId = data.readInt32();
             status_t status;
             sp<IAudioRecord> record = openRecord(pid, input,
-                    sampleRate, format, channelMask, bufferCount, flags, tid, &sessionId, &status);
+                    sampleRate, format, channelMask, frameCount, flags, tid, &sessionId, &status);
             reply->writeInt32(sessionId);
             reply->writeInt32(status);
             reply->writeStrongBinder(record->asBinder());
@@ -972,8 +972,8 @@
         case GET_RENDER_POSITION: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
-            uint32_t halFrames;
-            uint32_t dspFrames;
+            size_t halFrames;
+            size_t dspFrames;
             status_t status = getRenderPosition(&halFrames, &dspFrames, output);
             reply->writeInt32(status);
             if (status == NO_ERROR) {
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index abc8899..204e0ce 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -568,8 +568,8 @@
         }
 
         // initialize track
-        int afFrameCount;
-        int afSampleRate;
+        size_t afFrameCount;
+        uint32_t afSampleRate;
         audio_stream_type_t streamType = mSoundPool->streamType();
         if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
             afFrameCount = kDefaultFrameCount;
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 9bedff1..c3e5c40 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1387,8 +1387,8 @@
     }
     ALOGV("open(%u, %d, 0x%x, %d, %d, %d)", sampleRate, channelCount, channelMask,
             format, bufferCount, mSessionId);
-    int afSampleRate;
-    int afFrameCount;
+    uint32_t afSampleRate;
+    size_t afFrameCount;
     uint32_t frameCount;
 
     if (AudioSystem::getOutputFrameCount(&afFrameCount, mStreamType) != NO_ERROR) {
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 861aebe..3cf4d5c 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -58,7 +58,7 @@
     ALOGV("sampleRate: %d, channelCount: %d", sampleRate, channelCount);
     CHECK(channelCount == 1 || channelCount == 2);
 
-    int minFrameCount;
+    size_t minFrameCount;
     status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
                                            sampleRate,
                                            AUDIO_FORMAT_PCM_16_BIT,
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
index bf9ab3a..a400b4c 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
@@ -66,7 +66,7 @@
     def.eDir = OMX_DirInput;
     def.nBufferCountMin = kNumBuffers;
     def.nBufferCountActual = def.nBufferCountMin;
-    def.nBufferSize = 256 * 1024;
+    def.nBufferSize = 768 * 1024;
     def.bEnabled = OMX_TRUE;
     def.bPopulated = OMX_FALSE;
     def.eDomain = OMX_PortDomainVideo;
diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index 8f7d12b..7fc7037 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -758,31 +758,69 @@
     esds = NULL;
 }
 
-void addVorbisCodecInfo(
+status_t addVorbisCodecInfo(
         const sp<MetaData> &meta,
         const void *_codecPrivate, size_t codecPrivateSize) {
-    // printf("vorbis private data follows:\n");
     // hexdump(_codecPrivate, codecPrivateSize);
 
-    CHECK(codecPrivateSize >= 3);
+    if (codecPrivateSize < 1) {
+        return ERROR_MALFORMED;
+    }
 
     const uint8_t *codecPrivate = (const uint8_t *)_codecPrivate;
-    CHECK(codecPrivate[0] == 0x02);
 
-    size_t len1 = codecPrivate[1];
-    size_t len2 = codecPrivate[2];
+    if (codecPrivate[0] != 0x02) {
+        return ERROR_MALFORMED;
+    }
 
-    CHECK(codecPrivateSize > 3 + len1 + len2);
+    // codecInfo starts with two lengths, len1 and len2, that are
+    // "Xiph-style-lacing encoded"...
 
-    CHECK(codecPrivate[3] == 0x01);
-    meta->setData(kKeyVorbisInfo, 0, &codecPrivate[3], len1);
+    size_t offset = 1;
+    size_t len1 = 0;
+    while (offset < codecPrivateSize && codecPrivate[offset] == 0xff) {
+        len1 += 0xff;
+        ++offset;
+    }
+    if (offset >= codecPrivateSize) {
+        return ERROR_MALFORMED;
+    }
+    len1 += codecPrivate[offset++];
 
-    CHECK(codecPrivate[len1 + 3] == 0x03);
+    size_t len2 = 0;
+    while (offset < codecPrivateSize && codecPrivate[offset] == 0xff) {
+        len2 += 0xff;
+        ++offset;
+    }
+    if (offset >= codecPrivateSize) {
+        return ERROR_MALFORMED;
+    }
+    len2 += codecPrivate[offset++];
 
-    CHECK(codecPrivate[len1 + len2 + 3] == 0x05);
+    if (codecPrivateSize < offset + len1 + len2) {
+        return ERROR_MALFORMED;
+    }
+
+    if (codecPrivate[offset] != 0x01) {
+        return ERROR_MALFORMED;
+    }
+    meta->setData(kKeyVorbisInfo, 0, &codecPrivate[offset], len1);
+
+    offset += len1;
+    if (codecPrivate[offset] != 0x03) {
+        return ERROR_MALFORMED;
+    }
+
+    offset += len2;
+    if (codecPrivate[offset] != 0x05) {
+        return ERROR_MALFORMED;
+    }
+
     meta->setData(
-            kKeyVorbisBooks, 0, &codecPrivate[len1 + len2 + 3],
-            codecPrivateSize - len1 - len2 - 3);
+            kKeyVorbisBooks, 0, &codecPrivate[offset],
+            codecPrivateSize - offset);
+
+    return OK;
 }
 
 void MatroskaExtractor::addTracks() {
@@ -809,6 +847,8 @@
 
         sp<MetaData> meta = new MetaData;
 
+        status_t err = OK;
+
         switch (track->GetType()) {
             case VIDEO_TRACK:
             {
@@ -855,7 +895,8 @@
                 } else if (!strcmp("A_VORBIS", codecID)) {
                     meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_VORBIS);
 
-                    addVorbisCodecInfo(meta, codecPrivate, codecPrivateSize);
+                    err = addVorbisCodecInfo(
+                            meta, codecPrivate, codecPrivateSize);
                 } else if (!strcmp("A_MPEG/L3", codecID)) {
                     meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
                 } else {
@@ -872,6 +913,11 @@
                 continue;
         }
 
+        if (err != OK) {
+            ALOGE("skipping track, codec specific data was malformed.");
+            continue;
+        }
+
         long long durationNs = mSegment->GetDuration();
         meta->setInt64(kKeyDuration, (durationNs + 500) / 1000);
 
diff --git a/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp b/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp
index bc35aef..b913124 100644
--- a/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp
+++ b/media/libstagefright/wifi-display/sink/TunnelRenderer.cpp
@@ -271,6 +271,7 @@
     if (mFirstFailedAttemptUs + 50000ll > ALooper::GetNowUs()) {
         // We're willing to wait a little while to get the right packet.
 
+#if 0
         if (!mRequestedRetransmission) {
             ALOGI("requesting retransmission of seqNo %d",
                   (mLastDequeuedExtSeqNo + 1) & 0xffff);
@@ -280,7 +281,9 @@
             notify->post();
 
             mRequestedRetransmission = true;
-        } else {
+        } else
+#endif
+        {
             ALOGI("still waiting for the correct packet to arrive.");
         }
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 9353e70..384f268 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -450,7 +450,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         IAudioFlinger::track_flags_t *flags,
         const sp<IMemory>& sharedBuffer,
         audio_io_handle_t output,
@@ -472,6 +472,14 @@
         goto Exit;
     }
 
+    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
+    // and we don't yet support 8.24 or 32-bit PCM
+    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
+        ALOGE("createTrack() invalid format %d", format);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
     {
         Mutex::Autolock _l(mLock);
         PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -1291,7 +1299,7 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "standby: %d\n", mStandby);
     result.append(buffer);
-    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
+    snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate);
     result.append(buffer);
     snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
     result.append(buffer);
@@ -1722,7 +1730,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         const sp<IMemory>& sharedBuffer,
         int sessionId,
         IAudioFlinger::track_flags_t *flags,
@@ -1776,7 +1784,7 @@
                 frameCount, mFrameCount);
       } else {
         ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
-                "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d "
+                "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
                 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
                 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format,
                 audio_is_linear_pcm(format),
@@ -1791,7 +1799,7 @@
         if (minBufCount < 2) {
             minBufCount = 2;
         }
-        int minFrameCount = mNormalFrameCount * minBufCount;
+        size_t minFrameCount = mNormalFrameCount * minBufCount;
         if (frameCount < minFrameCount) {
             frameCount = minFrameCount;
         }
@@ -1801,7 +1809,7 @@
     if (mType == DIRECT) {
         if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
             if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
-                ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x "
+                ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x "
                         "for output %p with format %d",
                         sampleRate, format, channelMask, mOutput, mFormat);
                 lStatus = BAD_VALUE;
@@ -1811,7 +1819,7 @@
     } else {
         // Resampler implementation limits input sampling rate to 2 x output sampling rate.
         if (sampleRate > mSampleRate*2) {
-            ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
+            ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
             lStatus = BAD_VALUE;
             goto Exit;
         }
@@ -2280,7 +2288,7 @@
         // mNormalSink below
 {
     ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
-    ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "
+    ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, "
             "mFrameCount=%d, mNormalFrameCount=%d",
             mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
             mNormalFrameCount);
@@ -3126,7 +3134,7 @@
         uint32_t minFrames = 1;
         if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
                 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
-            if (t->sampleRate() == (int)mSampleRate) {
+            if (t->sampleRate() == mSampleRate) {
                 minFrames = mNormalFrameCount;
             } else {
                 // +1 for rounding and +1 for additional sample needed for interpolation
@@ -3624,7 +3632,7 @@
             NBAIO_Format format = teeSource->format();
             unsigned channelCount = Format_channelCount(format);
             ALOG_ASSERT(channelCount <= FCC_2);
-            unsigned sampleRate = Format_sampleRate(format);
+            uint32_t sampleRate = Format_sampleRate(format);
             wavHeader[22] = channelCount;       // number of channels
             wavHeader[24] = sampleRate;         // sample rate
             wavHeader[25] = sampleRate >> 8;
@@ -4089,7 +4097,7 @@
 {
     Mutex::Autolock _l(mLock);
     // FIXME explain this formula
-    int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
+    size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
     OutputTrack *outputTrack = new OutputTrack(thread,
                                             this,
                                             mSampleRate,
@@ -4177,7 +4185,7 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount,
+            size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId)
     :   RefBase(),
@@ -4190,11 +4198,12 @@
         mState(IDLE),
         mSampleRate(sampleRate),
         mFormat(format),
-        mFrameSize(0),  // will be set to correct value in constructor
+        mChannelMask(channelMask),
+        mChannelCount(popcount(channelMask)),
+        mFrameSize(audio_is_linear_pcm(format) ?
+                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
         mStepServerFailed(false),
         mSessionId(sessionId)
-        // mChannelCount
-        // mChannelMask
 {
     // client == 0 implies sharedBuffer == 0
     ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
@@ -4204,8 +4213,7 @@
 
     // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
     size_t size = sizeof(audio_track_cblk_t);
-    uint8_t channelCount = popcount(channelMask);
-    size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
+    size_t bufferSize = frameCount * mFrameSize;
     if (sharedBuffer == 0) {
         size += bufferSize;
     }
@@ -4236,11 +4244,9 @@
 //      mCblk->server = 0xffff0000;
 //      mCblk->userBase = 0xffff0000;
 //      mCblk->serverBase = 0xffff0000;
-        mChannelCount = channelCount;
-        mChannelMask = channelMask;
         if (sharedBuffer == 0) {
             mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
-            memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
+            memset(mBuffer, 0, bufferSize);
             // Force underrun condition to avoid false underrun callback until first data is
             // written to buffer (other flags are cleared)
             mCblk->flags = CBLK_UNDERRUN;
@@ -4306,23 +4312,22 @@
     ALOGV("TrackBase::reset");
 }
 
-int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
-    return (int)mCblk->sampleRate;
+uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
+    return mCblk->sampleRate;
 }
 
 void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
     audio_track_cblk_t* cblk = this->cblk();
-    size_t frameSize = mFrameSize;
-    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
-    int8_t *bufferEnd = bufferStart + frames * frameSize;
+    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize;
+    int8_t *bufferEnd = bufferStart + frames * mFrameSize;
 
     // Check validity of returned pointer in case the track control block would have been corrupted.
     ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd),
             "TrackBase::getBuffer buffer out of range:\n"
                 "    start: %p, end %p , mBuffer %p mBufferEnd %p\n"
-                "    server %u, serverBase %u, user %u, userBase %u, frameSize %d",
+                "    server %u, serverBase %u, user %u, userBase %u, frameSize %u",
                 bufferStart, bufferEnd, mBuffer, mBufferEnd,
-                cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize);
+                cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize);
 
     return bufferStart;
 }
@@ -4343,7 +4348,7 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount,
+            size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId,
             IAudioFlinger::track_flags_t flags)
@@ -4364,10 +4369,6 @@
     mUnderrunCount(0),
     mCachedVolume(1.0)
 {
-    // NOTE: frame size for 8 bit PCM data is based on a sample size of
-    // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
-    mFrameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) :
-            sizeof(uint8_t);
     if (mCblk != NULL) {
         // to avoid leaking a track name, do not allocate one unless there is an mCblk
         mName = thread->getTrackName_l(channelMask, sessionId);
@@ -4709,7 +4710,7 @@
     if (thread != 0) {
         Mutex::Autolock _l(thread->mLock);
         if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED &&
-                mState != PAUSING) {
+                mState != PAUSING && mState != IDLE && mState != FLUSHED) {
             return;
         }
         // No point remaining in PAUSED state after a flush => go to
@@ -4891,7 +4892,7 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount,
+            size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId) {
     if (!client->reserveTimedTrack())
@@ -4909,7 +4910,7 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount,
+            size_t frameCount,
             const sp<IMemory>& sharedBuffer,
             int sessionId)
     : Track(thread, client, streamType, sampleRate, format, channelMask,
@@ -5404,20 +5405,13 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount,
+            size_t frameCount,
             int sessionId)
     :   TrackBase(thread, client, sampleRate, format,
                   channelMask, frameCount, 0 /*sharedBuffer*/, sessionId),
         mOverflow(false)
 {
     ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
-    if (format == AUDIO_FORMAT_PCM_16_BIT) {
-        mFrameSize = mChannelCount * sizeof(int16_t);
-    } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
-        mFrameSize = mChannelCount * sizeof(int8_t);
-    } else {
-        mFrameSize = sizeof(int8_t);
-    }
 }
 
 AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
@@ -5530,7 +5524,7 @@
             uint32_t sampleRate,
             audio_format_t format,
             audio_channel_mask_t channelMask,
-            int frameCount)
+            size_t frameCount)
     :   Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
                 NULL, 0, IAudioFlinger::TRACK_DEFAULT),
     mActive(false), mSourceThread(sourceThread), mBuffers(NULL)
@@ -5541,7 +5535,7 @@
         mOutBuffer.frameCount = 0;
         playbackThread->mTracks.add(this);
         ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
-                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
+                "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p",
                 mCblk, mBuffer, mCblk->buffers,
                 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
     } else {
@@ -5924,7 +5918,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         IAudioFlinger::track_flags_t flags,
         pid_t tid,
         int *sessionId,
@@ -6301,7 +6295,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        int frameCount,
+        size_t frameCount,
         int sessionId,
         IAudioFlinger::track_flags_t flags,
         pid_t tid,
@@ -6558,7 +6552,7 @@
         result.append(buffer);
         snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
         result.append(buffer);
-        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
+        snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate);
         result.append(buffer);
     } else {
         result.append("No active record client\n");
@@ -6653,7 +6647,7 @@
         AudioParameter param = AudioParameter(keyValuePair);
         int value;
         audio_format_t reqFormat = mFormat;
-        int reqSamplingRate = mReqSampleRate;
+        uint32_t reqSamplingRate = mReqSampleRate;
         int reqChannelCount = mReqChannelCount;
 
         if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
@@ -6987,14 +6981,14 @@
 
 // ----------------------------------------------------------------------------
 
-int32_t AudioFlinger::getPrimaryOutputSamplingRate()
+uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
 {
     Mutex::Autolock _l(mLock);
     PlaybackThread *thread = primaryPlaybackThread_l();
     return thread != NULL ? thread->sampleRate() : 0;
 }
 
-int32_t AudioFlinger::getPrimaryOutputFrameCount()
+size_t AudioFlinger::getPrimaryOutputFrameCount()
 {
     Mutex::Autolock _l(mLock);
     PlaybackThread *thread = primaryPlaybackThread_l();
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 8cf58b1..830dfe9 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -92,7 +92,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
+                                size_t frameCount,
                                 IAudioFlinger::track_flags_t *flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
@@ -106,7 +106,7 @@
                                 uint32_t sampleRate,
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
-                                int frameCount,
+                                size_t frameCount,
                                 IAudioFlinger::track_flags_t flags,
                                 pid_t tid,
                                 int *sessionId,
@@ -207,8 +207,8 @@
 
     virtual audio_module_handle_t loadHwModule(const char *name);
 
-    virtual int32_t getPrimaryOutputSamplingRate();
-    virtual int32_t getPrimaryOutputFrameCount();
+    virtual uint32_t getPrimaryOutputSamplingRate();
+    virtual size_t getPrimaryOutputFrameCount();
 
     virtual     status_t    onTransact(
                                 uint32_t code,
@@ -390,7 +390,7 @@
                                         uint32_t sampleRate,
                                         audio_format_t format,
                                         audio_channel_mask_t channelMask,
-                                        int frameCount,
+                                        size_t frameCount,
                                         const sp<IMemory>& sharedBuffer,
                                         int sessionId);
             virtual             ~TrackBase();
@@ -423,7 +423,7 @@
 
             audio_channel_mask_t channelMask() const { return mChannelMask; }
 
-            int sampleRate() const; // FIXME inline after cblk sr moved
+            uint32_t sampleRate() const;    // FIXME inline after cblk sr moved
 
             // Return a pointer to the start of a contiguous slice of the track buffer.
             // Parameter 'offset' is the requested start position, expressed in
@@ -473,14 +473,13 @@
             const uint32_t      mSampleRate;    // initial sample rate only; for tracks which
                                 // support dynamic rates, the current value is in control block
             const audio_format_t mFormat;
-            size_t              mFrameSize; // AudioFlinger's view of frame size in shared memory,
+            const audio_channel_mask_t mChannelMask;
+            const uint8_t       mChannelCount;
+            const size_t        mFrameSize; // AudioFlinger's view of frame size in shared memory,
                                             // where for AudioTrack (but not AudioRecord),
                                             // 8-bit PCM samples are stored as 16-bit
-                                            // FIXME should be const
             bool                mStepServerFailed;
             const int           mSessionId;
-            uint8_t             mChannelCount;
-            audio_channel_mask_t mChannelMask;
             Vector < sp<SyncEvent> >mSyncEvents;
         };
 
@@ -791,7 +790,7 @@
                                         uint32_t sampleRate,
                                         audio_format_t format,
                                         audio_channel_mask_t channelMask,
-                                        int frameCount,
+                                        size_t frameCount,
                                         const sp<IMemory>& sharedBuffer,
                                         int sessionId,
                                         IAudioFlinger::track_flags_t flags);
@@ -861,6 +860,9 @@
 
             sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
 
+            // framesWritten is cumulative, never reset, and is shared all tracks
+            // audioHalFrames is derived from output latency
+            // FIXME parameters not needed, could get them from the thread
             bool presentationComplete(size_t framesWritten, size_t audioHalFrames);
 
         public:
@@ -893,6 +895,7 @@
             bool                mHasVolumeController;
             size_t              mPresentationCompleteFrames; // number of frames written to the
                                             // audio HAL when this track will be fully rendered
+                                            // zero means not monitoring
         private:
             IAudioFlinger::track_flags_t mFlags;
 
@@ -919,7 +922,7 @@
                                          uint32_t sampleRate,
                                          audio_format_t format,
                                          audio_channel_mask_t channelMask,
-                                         int frameCount,
+                                         size_t frameCount,
                                          const sp<IMemory>& sharedBuffer,
                                          int sessionId);
             virtual ~TimedTrack();
@@ -962,7 +965,7 @@
                        uint32_t sampleRate,
                        audio_format_t format,
                        audio_channel_mask_t channelMask,
-                       int frameCount,
+                       size_t frameCount,
                        const sp<IMemory>& sharedBuffer,
                        int sessionId);
 
@@ -997,7 +1000,7 @@
         };
 
 
-        // playback track
+        // playback track, used by DuplicatingThread
         class OutputTrack : public Track {
         public:
 
@@ -1011,7 +1014,7 @@
                                         uint32_t sampleRate,
                                         audio_format_t format,
                                         audio_channel_mask_t channelMask,
-                                        int frameCount);
+                                        size_t frameCount);
             virtual             ~OutputTrack();
 
             virtual status_t    start(AudioSystem::sync_event_t event =
@@ -1096,7 +1099,7 @@
                                     uint32_t sampleRate,
                                     audio_format_t format,
                                     audio_channel_mask_t channelMask,
-                                    int frameCount,
+                                    size_t frameCount,
                                     const sp<IMemory>& sharedBuffer,
                                     int sessionId,
                                     IAudioFlinger::track_flags_t *flags,
@@ -1460,7 +1463,7 @@
                                         uint32_t sampleRate,
                                         audio_format_t format,
                                         audio_channel_mask_t channelMask,
-                                        int frameCount,
+                                        size_t frameCount,
                                         int sessionId);
             virtual             ~RecordTrack();
 
@@ -1523,7 +1526,7 @@
                         uint32_t sampleRate,
                         audio_format_t format,
                         audio_channel_mask_t channelMask,
-                        int frameCount,
+                        size_t frameCount,
                         int sessionId,
                         IAudioFlinger::track_flags_t flags,
                         pid_t tid,
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 7d3681c..d68b839 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -470,6 +470,9 @@
      *
      */
 
+    mVolumeSIMD[0] = 0;
+    mVolumeSIMD[1] = 0;
+
     // Load the constants for coefficients
     int ok = pthread_once(&once_control, init_routine);
     if (ok != 0) {
@@ -494,6 +497,12 @@
     mRingFull = mImpulse + (numCoefs+1)*mChannelCount;
 }
 
+void AudioResamplerSinc::setVolume(int16_t left, int16_t right) {
+    AudioResampler::setVolume(left, right);
+    mVolumeSIMD[0] = int32_t(left)<<16;
+    mVolumeSIMD[1] = int32_t(right)<<16;
+}
+
 void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider)
 {
@@ -568,11 +577,9 @@
         }
 
         // handle boundary case
-        int32_t l, r;
         while (CC_LIKELY(outputIndex < outputSampleCount)) {
-            filterCoefficient<CHANNELS>(l, r, phaseFraction, impulse, vRL);
-            out[outputIndex++] += l;
-            out[outputIndex++] += r;
+            filterCoefficient<CHANNELS>(&out[outputIndex], phaseFraction, impulse, vRL);
+            outputIndex += 2;
 
             phaseFraction += phaseIncrement;
             const size_t phaseIndex = phaseFraction >> kNumPhaseBits;
@@ -628,19 +635,22 @@
 
 template<int CHANNELS>
 void AudioResamplerSinc::filterCoefficient(
-        int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples, uint32_t vRL)
+        int32_t* out, uint32_t phase, const int16_t *samples, uint32_t vRL)
 {
+    // NOTE: be very careful when modifying the code here. register
+    // pressure is very high and a small change might cause the compiler
+    // to generate far less efficient code.
+    // Always sanity check the result with objdump or test-resample.
+
     // compute the index of the coefficient on the positive side and
     // negative side
     const Constants& c(*mConstants);
+    const int32_t ONE = c.cMask | c.pMask;
     uint32_t indexP = ( phase & c.cMask) >> c.cShift;
-    uint32_t indexN = (-phase & c.cMask) >> c.cShift;
     uint32_t lerpP  = ( phase & c.pMask) >> c.pShift;
-    uint32_t lerpN  = (-phase & c.pMask) >> c.pShift;
-    if ((indexP == 0) && (lerpP == 0)) {
-        indexN = c.cMask >> c.cShift;
-        lerpN  = c.pMask >> c.pShift;
-    }
+    uint32_t indexN = ((ONE-phase) & c.cMask) >> c.cShift;
+    uint32_t lerpN  = ((ONE-phase) & c.pMask) >> c.pShift;
+
     const size_t offset = c.halfNumCoefs;
     indexP *= offset;
     indexN *= offset;
@@ -650,19 +660,19 @@
     int16_t const* sP = samples;
     int16_t const* sN = samples + CHANNELS;
 
-    l = 0;
-    r = 0;
     size_t count = offset;
 
     if (!USE_NEON) {
+        int32_t l = 0;
+        int32_t r = 0;
         for (size_t i=0 ; i<count ; i++) {
             interpolate<CHANNELS>(l, r, coefsP++, offset, lerpP, sP);
             sP -= CHANNELS;
             interpolate<CHANNELS>(l, r, coefsN++, offset, lerpN, sN);
             sN += CHANNELS;
         }
-        l = 2 * mulRL(1, l, vRL);
-        r = 2 * mulRL(0, r, vRL);
+        out[0] += 2 * mulRL(1, l, vRL);
+        out[1] += 2 * mulRL(0, r, vRL);
     } else if (CHANNELS == 1) {
         int32_t const* coefsP1 = coefsP + offset;
         int32_t const* coefsN1 = coefsN + offset;
@@ -670,15 +680,16 @@
         asm (
             "vmov.32        d2[0], %[lerpP]          \n"    // load the positive phase
             "vmov.32        d2[1], %[lerpN]          \n"    // load the negative phase
-            "veor           q0, q0                   \n"    // result, initialize to 0
+            "veor           q0, q0, q0               \n"    // result, initialize to 0
+            "vshl.s32       d2, d2, #16              \n"    // convert to 32 bits
 
             "1:                                      \n"
             "vld1.16        { d4}, [%[sP]]           \n"    // load 4 16-bits stereo samples
-            "vld1.32        { q8}, [%[coefsP0]]!     \n"    // load 4 32-bits coefs
-            "vld1.32        { q9}, [%[coefsP1]]!     \n"    // load 4 32-bits coefs for interpolation
+            "vld1.32        { q8}, [%[coefsP0]:128]! \n"    // load 4 32-bits coefs
+            "vld1.32        { q9}, [%[coefsP1]:128]! \n"    // load 4 32-bits coefs for interpolation
             "vld1.16        { d6}, [%[sN]]!          \n"    // load 4 16-bits stereo samples
-            "vld1.32        {q10}, [%[coefsN0]]!     \n"    // load 4 32-bits coefs
-            "vld1.32        {q11}, [%[coefsN1]]!     \n"    // load 4 32-bits coefs for interpolation
+            "vld1.32        {q10}, [%[coefsN0]:128]! \n"    // load 4 32-bits coefs
+            "vld1.32        {q11}, [%[coefsN1]:128]! \n"    // load 4 32-bits coefs for interpolation
 
             "vrev64.16      d4, d4                   \n"    // reverse 2 frames of the positive side
 
@@ -703,12 +714,16 @@
 
             "bne            1b                       \n"    // loop
 
+            "vld1.s32       {d2}, [%[vLR]]           \n"    // load volumes
+            "vld1.s32       {d3}, %[out]             \n"    // load the output
             "vpadd.s32      d0, d0, d1               \n"    // add all 4 partial sums
             "vpadd.s32      d0, d0, d0               \n"    // together
+            "vdup.i32       d0, d0[0]                \n"    // interleave L,R channels
+            "vqrdmulh.s32   d0, d0, d2               \n"    // apply volume
+            "vadd.s32       d3, d3, d0               \n"    // accumulate result
+            "vst1.s32       {d0}, %[out]             \n"    // store result
 
-            "vmov.s32       %[l], d0[0]              \n"    // save result in ARM register
-
-            : [l]  "=r" (l),
+            : [out]     "=Uv" (out[0]),
               [count]   "+r" (count),
               [coefsP0] "+r" (coefsP),
               [coefsP1] "+r" (coefsP1),
@@ -716,16 +731,14 @@
               [coefsN1] "+r" (coefsN1),
               [sP]      "+r" (sP),
               [sN]      "+r" (sN)
-            : [lerpP]   "r" (lerpP<<16),
-              [lerpN]   "r" (lerpN<<16),
-              [vRL]     "r" (vRL)
+            : [lerpP]   "r" (lerpP),
+              [lerpN]   "r" (lerpN),
+              [vLR]     "r" (mVolumeSIMD)
             : "cc", "memory",
               "q0", "q1", "q2", "q3",
               "q8", "q9", "q10", "q11",
               "q12", "q14"
         );
-        l = 2 * mulRL(1, l, vRL);
-        r = l;
     } else if (CHANNELS == 2) {
         int32_t const* coefsP1 = coefsP + offset;
         int32_t const* coefsN1 = coefsN + offset;
@@ -733,16 +746,17 @@
         asm (
             "vmov.32        d2[0], %[lerpP]          \n"    // load the positive phase
             "vmov.32        d2[1], %[lerpN]          \n"    // load the negative phase
-            "veor           q0, q0                   \n"    // result, initialize to 0
-            "veor           q4, q4                   \n"    // result, initialize to 0
+            "veor           q0, q0, q0               \n"    // result, initialize to 0
+            "veor           q4, q4, q4               \n"    // result, initialize to 0
+            "vshl.s32       d2, d2, #16              \n"    // convert to 32 bits
 
             "1:                                      \n"
             "vld2.16        {d4,d5}, [%[sP]]         \n"    // load 4 16-bits stereo samples
-            "vld1.32        { q8}, [%[coefsP0]]!     \n"    // load 4 32-bits coefs
-            "vld1.32        { q9}, [%[coefsP1]]!     \n"    // load 4 32-bits coefs for interpolation
+            "vld1.32        { q8}, [%[coefsP0]:128]! \n"    // load 4 32-bits coefs
+            "vld1.32        { q9}, [%[coefsP1]:128]! \n"    // load 4 32-bits coefs for interpolation
             "vld2.16        {d6,d7}, [%[sN]]!        \n"    // load 4 16-bits stereo samples
-            "vld1.32        {q10}, [%[coefsN0]]!     \n"    // load 4 32-bits coefs
-            "vld1.32        {q11}, [%[coefsN1]]!     \n"    // load 4 32-bits coefs for interpolation
+            "vld1.32        {q10}, [%[coefsN0]:128]! \n"    // load 4 32-bits coefs
+            "vld1.32        {q11}, [%[coefsN1]:128]! \n"    // load 4 32-bits coefs for interpolation
 
             "vrev64.16      d4, d4                   \n"    // reverse 2 frames of the positive side
             "vrev64.16      d5, d5                   \n"    // reverse 2 frames of the positive side
@@ -774,16 +788,18 @@
 
             "bne            1b                       \n"    // loop
 
-            "vpadd.s32      d0, d0, d1               \n"    // add all 4 partial sums
-            "vpadd.s32      d8, d8, d9               \n"    // add all 4 partial sums
+            "vld1.s32       {d2}, [%[vLR]]           \n"    // load volumes
+            "vld1.s32       {d3}, %[out]             \n"    // load the output
+            "vpadd.s32      d0, d0, d1               \n"    // add all 4 partial sums from q0
+            "vpadd.s32      d8, d8, d9               \n"    // add all 4 partial sums from q4
             "vpadd.s32      d0, d0, d0               \n"    // together
             "vpadd.s32      d8, d8, d8               \n"    // together
+            "vtrn.s32       d0, d8                   \n"    // interlace L,R channels
+            "vqrdmulh.s32   d0, d0, d2               \n"    // apply volume
+            "vadd.s32       d3, d3, d0               \n"    // accumulate result
+            "vst1.s32       {d0}, %[out]             \n"    // store result
 
-            "vmov.s32       %[l], d0[0]              \n"    // save result in ARM register
-            "vmov.s32       %[r], d8[0]              \n"    // save result in ARM register
-
-            : [l]  "=r" (l),
-              [r]  "=r" (r),
+            : [out]     "=Uv" (out[0]),
               [count]   "+r" (count),
               [coefsP0] "+r" (coefsP),
               [coefsP1] "+r" (coefsP1),
@@ -791,16 +807,14 @@
               [coefsN1] "+r" (coefsN1),
               [sP]      "+r" (sP),
               [sN]      "+r" (sN)
-            : [lerpP]   "r" (lerpP<<16),
-              [lerpN]   "r" (lerpN<<16),
-              [vRL]     "r" (vRL)
+            : [lerpP]   "r" (lerpP),
+              [lerpN]   "r" (lerpN),
+              [vLR]     "r" (mVolumeSIMD)
             : "cc", "memory",
               "q0", "q1", "q2", "q3", "q4",
               "q8", "q9", "q10", "q11",
               "q12", "q13", "q14", "q15"
         );
-        l = 2 * mulRL(1, l, vRL);
-        r = 2 * mulRL(0, r, vRL);
     }
 }
 
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index 1b14019..09c6866 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -44,13 +44,15 @@
 private:
     void init();
 
+    virtual void setVolume(int16_t left, int16_t right);
+
     template<int CHANNELS>
     void resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 
     template<int CHANNELS>
     inline void filterCoefficient(
-            int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples, uint32_t vRL);
+            int32_t* out, uint32_t phase, const int16_t *samples, uint32_t vRL);
 
     template<int CHANNELS>
     inline void interpolate(
@@ -65,6 +67,7 @@
     int16_t *mState;
     int16_t *mImpulse;
     int16_t *mRingFull;
+    int32_t mVolumeSIMD[2];
 
     const int32_t * mFirCoefs;
     static const int32_t mFirCoefsDown[];
@@ -75,14 +78,11 @@
     static const int32_t RESAMPLE_FIR_LERP_INT_BITS  = 7;
 
     struct Constants {
-        // we have 16 coefs samples per zero-crossing
         int coefsBits;
         int cShift;
         uint32_t cMask;
-
         int pShift;
         uint32_t pMask;
-
         // number of zero-crossing on each side
         unsigned int halfNumCoefs;
     };
diff --git a/services/audioflinger/StateQueue.h b/services/audioflinger/StateQueue.h
index c9b5111..e33b3c6 100644
--- a/services/audioflinger/StateQueue.h
+++ b/services/audioflinger/StateQueue.h
@@ -174,7 +174,7 @@
 #endif
 
 private:
-    static const unsigned kN = 4;       // values != 4 are not supported by this code
+    static const unsigned kN = 4;       // values < 4 are not supported by this code
     T                 mStates[kN];      // written by mutator, read by observer
 
     // "volatile" is meaningless with SMP, but here it indicates that we're using atomic ops
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 3b66530..b082e8c 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -177,7 +177,7 @@
             double t = double(i) / input_freq;
             double y = sin(M_PI * k * t * t);
             int16_t yi = floor(y * 32767.0 + 0.5);
-            for (size_t j=0 ; j<channels ; j++) {
+            for (size_t j=0 ; j<(size_t)channels ; j++) {
                 in[i*channels + j] = yi / (1+j);
             }
         }