diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 0439cb0..c724949 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -39,8 +39,12 @@
      * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
      */
     enum event_type {
-        EVENT_MORE_DATA = 0,        // Request to read more data from PCM buffer.
-        EVENT_OVERRUN = 1,          // PCM buffer overrun occurred.
+        EVENT_MORE_DATA = 0,        // Request to read available data from buffer.
+                                    // If this event is delivered but the callback handler
+                                    // does not want to read the available data, the handler must
+                                    // explicitly
+                                    // ignore the event by setting frameCount to zero.
+        EVENT_OVERRUN = 1,          // Buffer overrun occurred.
         EVENT_MARKER = 2,           // Record head is at the specified marker position
                                     // (See setMarkerPosition()).
         EVENT_NEW_POS = 3,          // Record head is at a new position
@@ -63,6 +67,7 @@
                                     // (currently ignored but will make the primary field in future)
 
         size_t      size;           // input/output in bytes == frameCount * frameSize
+                                    // on output is the number of bytes actually drained
                                     // FIXME this is redundant with respect to frameCount,
                                     // and TRANSFER_OBTAIN mode is broken for 8-bit data
                                     // since we don't define the frame format
@@ -76,7 +81,7 @@
 
     /* As a convenience, if a callback is supplied, a handler thread
      * is automatically created with the appropriate priority. This thread
-     * invokes the callback when a new buffer becomes ready or various conditions occur.
+     * invokes the callback when a new buffer becomes available or various conditions occur.
      * Parameters:
      *
      * event:   type of event notified (see enum AudioRecord::event_type).
@@ -99,6 +104,8 @@
      *  - NO_ERROR: successful operation
      *  - NO_INIT: audio server or audio hardware not initialized
      *  - BAD_VALUE: unsupported configuration
+     * frameCount is guaranteed to be non-zero if status is NO_ERROR,
+     * and is undefined otherwise.
      */
 
      static status_t getMinFrameCount(size_t* frameCount,
@@ -109,7 +116,7 @@
     /* How data is transferred from AudioRecord
      */
     enum transfer_type {
-        TRANSFER_DEFAULT,   // not specified explicitly; determine from other parameters
+        TRANSFER_DEFAULT,   // not specified explicitly; determine from the other parameters
         TRANSFER_CALLBACK,  // callback EVENT_MORE_DATA
         TRANSFER_OBTAIN,    // FIXME deprecated: call obtainBuffer() and releaseBuffer()
         TRANSFER_SYNC,      // synchronous read()
@@ -137,7 +144,7 @@
      *                     be larger if the requested size is not compatible with current audio HAL
      *                     latency.  Zero means to use a default value.
      * cbf:                Callback function. If not null, this function is called periodically
-     *                     to consume new PCM data and inform of marker, position updates, etc.
+     *                     to consume new data and inform of marker, position updates, etc.
      * user:               Context for use by the callback receiver.
      * notificationFrames: The callback function is called each time notificationFrames PCM
      *                     frames are ready in record track output buffer.
@@ -171,9 +178,10 @@
      * Returned status (from utils/Errors.h) can be:
      *  - NO_ERROR: successful intialization
      *  - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use
-     *  - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
+     *  - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
      *  - NO_INIT: audio server or audio hardware not initialized
      *  - PERMISSION_DENIED: recording is not allowed for the requesting process
+     * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord.
      *
      * Parameters not listed in the AudioRecord constructors above:
      *
@@ -192,7 +200,7 @@
                             transfer_type transferType = TRANSFER_DEFAULT,
                             audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE);
 
-    /* Result of constructing the AudioRecord. This must be checked
+    /* Result of constructing the AudioRecord. This must be checked for successful initialization
      * before using any AudioRecord API (except for set()), because using
      * an uninitialized AudioRecord produces undefined results.
      * See set() method above for possible return codes.
@@ -221,7 +229,7 @@
             status_t    start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
                               int triggerSession = 0);
 
-    /* Stop a track. If set, the callback will cease being called.  Note that obtainBuffer() still
+    /* Stop a track.  The callback will cease being called.  Note that obtainBuffer() still
      * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK.
      */
             void        stop();
@@ -236,7 +244,7 @@
      * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
      * with marker == 0 cancels marker notification callback.
      * To set a marker at a position which would compute as 0,
-     * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+     * a workaround is to set the marker at a nearby position such as ~0 or 1.
      * If the AudioRecord has been opened with no callback function associated,
      * the operation will fail.
      *
@@ -378,8 +386,10 @@
      * returning the current value by this function call.  Such loss typically occurs when the
      * user space process is blocked longer than the capacity of audio driver buffers.
      * Units: the number of input audio frames.
+     * FIXME The side-effect of resetting the counter may be incompatible with multi-client.
+     * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects.
      */
-            unsigned int  getInputFramesLost() const;
+            uint32_t    getInputFramesLost() const;
 
 private:
     /* copying audio record objects is not allowed */
@@ -426,6 +436,7 @@
             nsecs_t processAudioBuffer();
 
             // caller must hold lock on mLock for all _l methods
+
             status_t openRecord_l(size_t epoch);
 
             // FIXME enum is faster than strcmp() for parameter 'from'
@@ -477,7 +488,7 @@
 
     audio_io_handle_t       mInput;             // returned by AudioSystem::getInput()
 
-    // may be changed if IAudioRecord object is re-created
+    // Next 3 fields may be changed if IAudioRecord is re-created, but always != 0
     sp<IAudioRecord>        mAudioRecord;
     sp<IMemory>             mCblkMemory;
     audio_track_cblk_t*     mCblk;              // re-load after mLock.unlock()
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index ca9aaf7..706344a 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -138,7 +138,7 @@
                                       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 size_t getInputFramesLost(audio_io_handle_t ioHandle);
+    static uint32_t getInputFramesLost(audio_io_handle_t ioHandle);
 
     static int newAudioSessionId();
     static void acquireAudioSessionId(int audioSession);
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 789d255..e59b0df 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -160,7 +160,7 @@
      * sampleRate:         Data source sampling rate in Hz.
      * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
      *                     16 bits per sample).
-     * channelMask:        Channel mask.
+     * channelMask:        Channel mask, such that audio_is_output_channel(channelMask) is true.
      * frameCount:         Minimum size of track PCM buffer in frames. This defines the
      *                     application's contribution to the
      *                     latency of the track. The actual size selected by the AudioTrack could be
@@ -338,7 +338,7 @@
      */
             status_t    setSampleRate(uint32_t sampleRate);
 
-    /* Return current source sample rate in Hz, or 0 if unknown */
+    /* Return current source sample rate in Hz */
             uint32_t    getSampleRate() const;
 
     /* Enables looping and sets the start and end points of looping.
@@ -363,7 +363,7 @@
     /* Sets marker position. When playback reaches the number of frames specified, a callback with
      * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
      * notification callback.  To set a marker at a position which would compute as 0,
-     * a workaround is to the set the marker at a nearby position such as ~0 or 1.
+     * a workaround is to set the marker at a nearby position such as ~0 or 1.
      * If the AudioTrack has been opened with no callback function associated, the operation will
      * fail.
      *
@@ -717,6 +717,7 @@
     uint32_t                mObservedSequence;      // last observed value of mSequence
 
     uint32_t                mLoopPeriod;            // in frames, zero means looping is disabled
+
     uint32_t                mMarkerPosition;        // in wrapping (overflow) frame units
     bool                    mMarkerReached;
     uint32_t                mNewPosition;           // in frames
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 899d79f..c9cffe3 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -170,7 +170,7 @@
     virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
                                     audio_io_handle_t output) const = 0;
 
-    virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
+    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
 
     virtual int newAudioSessionId() = 0;
 
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index e39a475..98697f5 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -289,6 +289,9 @@
 
     // reset current position as seen by client to 0
     mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
+    // force refresh of remaining frames by processAudioBuffer() as last
+    // read before stop could be partial.
+    mRefreshRemaining = true;
 
     mNewPosition = mProxy->getPosition() + mUpdatePeriod;
     int32_t flags = android_atomic_acquire_load(&mCblk->mFlags);
@@ -352,6 +355,7 @@
 
 status_t AudioRecord::setMarkerPosition(uint32_t marker)
 {
+    // The only purpose of setting marker position is to get a callback
     if (mCbf == NULL) {
         return INVALID_OPERATION;
     }
@@ -377,6 +381,7 @@
 
 status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
 {
+    // The only purpose of setting position update period is to get a callback
     if (mCbf == NULL) {
         return INVALID_OPERATION;
     }
@@ -412,7 +417,7 @@
     return NO_ERROR;
 }
 
-unsigned int AudioRecord::getInputFramesLost() const
+uint32_t AudioRecord::getInputFramesLost() const
 {
     // no need to check mActive, because if inactive this will return 0, which is what we want
     return AudioSystem::getInputFramesLost(getInput());
@@ -591,6 +596,9 @@
                 if (newSequence == oldSequence) {
                     status = restoreRecord_l("obtainBuffer");
                     if (status != NO_ERROR) {
+                        buffer.mFrameCount = 0;
+                        buffer.mRaw = NULL;
+                        buffer.mNonContig = 0;
                         break;
                     }
                 }
@@ -770,7 +778,7 @@
     int32_t sequence = mSequence;
 
     // These fields don't need to be cached, because they are assigned only by set():
-    //      mTransfer, mCbf, mUserData, mSampleRate
+    //      mTransfer, mCbf, mUserData, mSampleRate, mFrameSize
 
     mLock.unlock();
 
@@ -844,8 +852,8 @@
                 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
         requested = &ClientProxy::kNonBlocking;
         size_t avail = audioBuffer.frameCount + nonContig;
-        ALOGV("obtainBuffer(%u) returned %u = %u + %u",
-                mRemainingFrames, avail, audioBuffer.frameCount, nonContig);
+        ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+                mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
         if (err != NO_ERROR) {
             if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
                 break;
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 69d9273..dcb72f8 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -413,9 +413,9 @@
     return af->getRenderPosition(halFrames, dspFrames, output);
 }
 
-size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
+uint32_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
-    unsigned int result = 0;
+    uint32_t result = 0;
     if (af == 0) return result;
     if (ioHandle == 0) return result;
 
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index eeaab44..7e6074d 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -54,16 +54,22 @@
     status_t status;
     status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
     if (status != NO_ERROR) {
+        ALOGE("Unable to query output sample rate for stream type %d; status %d",
+                streamType, status);
         return status;
     }
     size_t afFrameCount;
     status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
     if (status != NO_ERROR) {
+        ALOGE("Unable to query output frame count for stream type %d; status %d",
+                streamType, status);
         return status;
     }
     uint32_t afLatency;
     status = AudioSystem::getOutputLatency(&afLatency, streamType);
     if (status != NO_ERROR) {
+        ALOGE("Unable to query output latency for stream type %d; status %d",
+                streamType, status);
         return status;
     }
 
@@ -362,7 +368,7 @@
             mAudioTrackThread->requestExitAndWait();
             mAudioTrackThread.clear();
         }
-        //Use of direct and offloaded output streams is ref counted by audio policy manager.
+        // Use of direct and offloaded output streams is ref counted by audio policy manager.
         // As getOutput was called above and resulted in an output stream to be opened,
         // we need to release it.
         AudioSystem::releaseOutput(output);
@@ -702,6 +708,7 @@
     AutoMutex lock(mLock);
     mNewPosition = mProxy->getPosition() + updatePeriod;
     mUpdatePeriod = updatePeriod;
+
     return NO_ERROR;
 }
 
@@ -1619,7 +1626,6 @@
         size_t reqSize = audioBuffer.size;
         mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
         size_t writtenSize = audioBuffer.size;
-        size_t writtenFrames = writtenSize / mFrameSize;
 
         // Sanity check on returned size
         if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
@@ -1744,7 +1750,7 @@
         }
     }
     if (result != NO_ERROR) {
-        //Use of direct and offloaded output streams is ref counted by audio policy manager.
+        // Use of direct and offloaded output streams is ref counted by audio policy manager.
         // As getOutput was called above and resulted in an output stream to be opened,
         // we need to release it.
         AudioSystem::releaseOutput(output);
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index ecbb5bf..86a4d74 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -575,13 +575,16 @@
         return status;
     }
 
-    virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const
+    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32((int32_t) ioHandle);
-        remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
-        return reply.readInt32();
+        status_t status = remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
+        if (status != NO_ERROR) {
+            return 0;
+        }
+        return (uint32_t) reply.readInt32();
     }
 
     virtual int newAudioSessionId()
@@ -1056,7 +1059,7 @@
         case GET_INPUT_FRAMES_LOST: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
-            reply->writeInt32(getInputFramesLost(ioHandle));
+            reply->writeInt32((int32_t) getInputFramesLost(ioHandle));
             return NO_ERROR;
         } break;
         case NEW_AUDIO_SESSION_ID: {
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index d7223d9..cadadc8 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -278,7 +278,7 @@
 
     // Drop retrieved and previously lost audio data.
     if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
-        mRecord->getInputFramesLost();
+        (void) mRecord->getInputFramesLost();
         ALOGV("Drop audio data at %lld/%lld us", timeUs, mStartTimeUs);
         return OK;
     }
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1340a56..1257161 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1059,7 +1059,7 @@
     return size;
 }
 
-unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
+uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
 {
     Mutex::Autolock _l(mLock);
 
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 066d5d5..0ab43e0 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -189,7 +189,7 @@
     virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
                                        audio_io_handle_t output) const;
 
-    virtual     unsigned int  getInputFramesLost(audio_io_handle_t ioHandle) const;
+    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
 
     virtual int newAudioSessionId();
 
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index b191a8f..d5a0e21 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5372,7 +5372,7 @@
     mRsmpInIndex = mFrameCount;
 }
 
-unsigned int AudioFlinger::RecordThread::getInputFramesLost()
+uint32_t AudioFlinger::RecordThread::getInputFramesLost()
 {
     Mutex::Autolock _l(mLock);
     if (initCheck() != NO_ERROR) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 6b81c38..2b749fa 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -907,7 +907,7 @@
     virtual String8     getParameters(const String8& keys);
     virtual void        audioConfigChanged_l(int event, int param = 0);
             void        readInputParameters();
-    virtual unsigned int  getInputFramesLost();
+    virtual uint32_t    getInputFramesLost();
 
     virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
     virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);