diff --git a/core/jni/android_media_AudioRecord.cpp b/core/jni/android_media_AudioRecord.cpp
index 547607e..68a8de8 100644
--- a/core/jni/android_media_AudioRecord.cpp
+++ b/core/jni/android_media_AudioRecord.cpp
@@ -220,7 +220,7 @@
         ALOGE("Error creating AudioRecord: Error retrieving session id pointer");
         goto native_init_failure;
     }
-    // read the audio session ID back from AudioTrack in case a new session was created during set()
+    // read the audio session ID back from AudioRecord in case a new session was created during set()
     nSession[0] = lpRecorder->getSessionId();
     env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
     nSession = NULL;
diff --git a/core/jni/android_media_AudioSystem.cpp b/core/jni/android_media_AudioSystem.cpp
index 31c5ed4..ee5eb7e 100644
--- a/core/jni/android_media_AudioSystem.cpp
+++ b/core/jni/android_media_AudioSystem.cpp
@@ -1,4 +1,4 @@
-/* //device/libs/android_runtime/android_media_AudioSystem.cpp
+/*
 **
 ** Copyright 2006, The Android Open Source Project
 **
diff --git a/core/jni/android_media_ToneGenerator.cpp b/core/jni/android_media_ToneGenerator.cpp
index 0a337fae..544b4c0 100644
--- a/core/jni/android_media_ToneGenerator.cpp
+++ b/core/jni/android_media_ToneGenerator.cpp
@@ -1,4 +1,4 @@
-/* //device/libs/android_runtime/android_media_AudioSystem.cpp
+/*
  **
  ** Copyright 2008, The Android Open Source Project
  **
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index da99620..1916ac5 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -131,7 +131,7 @@
         NUM_CONFIG_EVENTS
     };
 
-    // audio output descritor used to cache output configurations in client process to avoid frequent calls
+    // audio output descriptor used to cache output configurations in client process to avoid frequent calls
     // through IAudioFlinger
     class OutputDescriptor {
     public:
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 11db81b..0f2c323 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -58,8 +58,8 @@
         EVENT_BUFFER_END = 5        // Playback head is at the end of the buffer.
     };
 
-    /* Create Buffer on the stack and pass it to obtainBuffer()
-     * and releaseBuffer().
+    /* Client should declare Buffer on the stack and pass address to obtainBuffer()
+     * and releaseBuffer().  See also callback_t for EVENT_MORE_DATA.
      */
 
     class Buffer
@@ -68,12 +68,16 @@
         enum {
             MUTE    = 0x00000001
         };
-        uint32_t    flags;
+        uint32_t    flags;        // 0 or MUTE
         audio_format_t format; // but AUDIO_FORMAT_PCM_8_BIT -> AUDIO_FORMAT_PCM_16_BIT
         // accessed directly by WebKit ANP callback
         int         channelCount; // will be removed in the future, do not use
-        size_t      frameCount;
-        size_t      size;
+
+        size_t      frameCount;   // number of sample frames corresponding to size;
+                                  // on input it is the number of frames desired,
+                                  // on output is the number of frames actually filled
+
+        size_t      size;         // input/output in byte units
         union {
             void*       raw;
             short*      i16;    // signed 16-bit
@@ -84,15 +88,15 @@
 
     /* 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 available or an underrun condition occurs.
+     * invokes the callback when a new buffer becomes available or various conditions occur.
      * Parameters:
      *
      * event:   type of event notified (see enum AudioTrack::event_type).
      * user:    Pointer to context for use by the callback receiver.
      * info:    Pointer to optional parameter according to event type:
      *          - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write
-     *          more bytes than indicated by 'size' field and update 'size' if less bytes are
-     *          written.
+     *            more bytes than indicated by 'size' field and update 'size' if fewer bytes are
+     *            written.
      *          - EVENT_UNDERRUN: unused.
      *          - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
      *          - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames.
@@ -225,7 +229,7 @@
      */
             uint32_t     latency() const;
 
-    /* getters, see constructor */
+    /* getters, see constructors and set() */
 
             audio_stream_type_t streamType() const;
             audio_format_t format() const;
@@ -401,13 +405,19 @@
             status_t    attachAuxEffect(int effectId);
 
     /* Obtains a buffer of "frameCount" frames. The buffer must be
-     * filled entirely. If the track is stopped, obtainBuffer() returns
+     * filled entirely, and then released with releaseBuffer().
+     * If the track is stopped, obtainBuffer() returns
      * STOPPED instead of NO_ERROR as long as there are buffers available,
      * at which point NO_MORE_BUFFERS is returned.
      * Buffers will be returned until the pool (buffercount())
      * is exhausted, at which point obtainBuffer() will either block
      * or return WOULD_BLOCK depending on the value of the "blocking"
      * parameter.
+     *
+     * Interpretation of waitCount:
+     *  +n  limits wait time to n * WAIT_PERIOD_MS,
+     *  -1  causes an (almost) infinite wait time,
+     *   0  non-blocking.
      */
 
         enum {
@@ -416,12 +426,19 @@
         };
 
             status_t    obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
+
+    /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */
             void        releaseBuffer(Buffer* audioBuffer);
 
     /* As a convenience we provide a write() interface to the audio buffer.
-     * This is implemented on top of lockBuffer/unlockBuffer. For best
-     * performance use callbacks. Return actual number of bytes written.
-     *
+     * This is implemented on top of obtainBuffer/releaseBuffer. For best
+     * performance use callbacks. Returns actual number of bytes written >= 0,
+     * or one of the following negative status codes:
+     *      INVALID_OPERATION   AudioTrack is configured for shared buffer mode
+     *      BAD_VALUE           size is invalid
+     *      STOPPED             AudioTrack was stopped during the write
+     *      NO_MORE_BUFFERS     when obtainBuffer() returns same
+     *      or any other error code returned by IAudioTrack::start() or restoreTrack_l().
      */
             ssize_t     write(const void* buffer, size_t size);
 
@@ -448,6 +465,7 @@
         AudioTrack& mReceiver;
     };
 
+            // body of AudioTrackThread::threadLoop()
             bool processAudioBuffer(const sp<AudioTrackThread>& thread);
             status_t createTrack_l(audio_stream_type_t streamType,
                                  uint32_t sampleRate,
@@ -484,7 +502,7 @@
 
     bool                    mActive;                // protected by mLock
 
-    callback_t              mCbf;
+    callback_t              mCbf;                   // callback handler for events, or NULL
     void*                   mUserData;
     uint32_t                mNotificationFramesReq; // requested number of frames between each notification callback
     uint32_t                mNotificationFramesAct; // actual number of frames between each notification callback
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index df83995..65c26f47 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -87,7 +87,7 @@
 //    Description:    Creates an effect engine of the specified type and returns an
 //          effect control interface on this engine. The function will allocate the
 //          resources for an instance of the requested effect engine and return
-//          a handler on the effect control interface.
+//          a handle on the effect control interface.
 //
 //    Input:
 //          pEffectUuid:    pointer to the effect uuid.
@@ -115,17 +115,17 @@
 //
 //    Function:       EffectRelease
 //
-//    Description:    Releases the effect engine whose handler is given as argument.
+//    Description:    Releases the effect engine whose handle is given as argument.
 //          All resources allocated to this particular instance of the effect are
 //          released.
 //
 //    Input:
-//          handle:    handler on the effect interface to be released.
+//          handle:    handle on the effect interface to be released.
 //
 //    Output:
 //        returned value:    0          successful operation.
 //                          -ENODEV     factory failed to initialize
-//                          -EINVAL     invalid interface handler
+//                          -EINVAL     invalid interface handle
 //
 ////////////////////////////////////////////////////////////////////////////////
 int EffectRelease(effect_handle_t handle);
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index b346722..e4772a1 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -46,12 +46,12 @@
 
     /* Stop a track. If set, the callback will cease being called and
      * obtainBuffer will return an error. Buffers that are already released 
-     * will be processed, unless flush() is called.
+     * will continue to be processed, unless/until flush() is called.
      */
     virtual void        stop() = 0;
 
-    /* Flush a stopped track. All pending buffers are discarded.
-     * This function has no effect if the track is not stopped.
+    /* Flush a stopped or paused track. All pending/released buffers are discarded.
+     * This function has no effect if the track is not stopped or paused.
      */
     virtual void        flush() = 0;
 
@@ -62,7 +62,7 @@
     
     /* Pause a track. If set, the callback will cease being called and
      * obtainBuffer will return an error. Buffers that are already released 
-     * will be processed, unless flush() is called.
+     * will continue to be processed, unless/until flush() is called.
      */
     virtual void        pause() = 0;
 
diff --git a/include/media/MemoryLeakTrackUtil.h b/include/media/MemoryLeakTrackUtil.h
index 290b748..ac0f6b2 100644
--- a/include/media/MemoryLeakTrackUtil.h
+++ b/include/media/MemoryLeakTrackUtil.h
@@ -19,7 +19,7 @@
 
 namespace android {
 /*
- * Dump the memory adddress of the calling process to the given fd.
+ * Dump the memory address of the calling process to the given fd.
  */
 extern void dumpMemoryAddresses(int fd);
 
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 23226c0..af2db93 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -62,16 +62,23 @@
     // are in the same line of data cache.
                 Mutex       lock;           // sizeof(int)
                 Condition   cv;             // sizeof(int)
+
+                // next 4 are offsets within "buffers"
     volatile    uint32_t    user;
     volatile    uint32_t    server;
                 uint32_t    userBase;
                 uint32_t    serverBase;
+
+                // if there is a shared buffer, "buffers" is the value of pointer() for the shared
+                // buffer, otherwise "buffers" points immediately after the control block
                 void*       buffers;
                 uint32_t    frameCount;
+
                 // Cache line boundary
+
                 uint32_t    loopStart;
-                uint32_t    loopEnd;
-                int         loopCount;
+                uint32_t    loopEnd;        // read-only for server, read/write for client
+                int         loopCount;      // read/write for client
 
                 // Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
                 // Left channel is in [0:15], right channel is in [16:31].
@@ -82,29 +89,39 @@
 public:
 
                 uint32_t    sampleRate;
+
                 // NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for
                 // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
                 // 16 bit because data is converted to 16 bit before being stored in buffer
 
+                // read-only for client, server writes once at initialization and is then read-only
                 uint8_t     frameSize;       // would normally be size_t, but 8 bits is plenty
+
+                // never used
                 uint8_t     pad1;
+
+                // used by client only
                 uint16_t    bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger
 
-                uint16_t    waitTimeMs;      // Cumulated wait time
+                uint16_t    waitTimeMs;      // Cumulated wait time, used by client only
 private:
+                // client write-only, server read-only
                 uint16_t    mSendLevel;      // Fixed point U4.12 so 0x1000 means 1.0
 public:
     volatile    int32_t     flags;
 
                 // Cache line boundary (32 bytes)
 
+                // Since the control block is always located in shared memory, this constructor
+                // is only used for placement new().  It is never used for regular new() or stack.
                             audio_track_cblk_t();
-                uint32_t    stepUser(uint32_t frameCount);
-                bool        stepServer(uint32_t frameCount);
+                uint32_t    stepUser(uint32_t frameCount);      // called by client only, where
+                // client includes regular AudioTrack and AudioFlinger::PlaybackThread::OutputTrack
+                bool        stepServer(uint32_t frameCount);    // called by server only
                 void*       buffer(uint32_t offset) const;
                 uint32_t    framesAvailable();
                 uint32_t    framesAvailable_l();
-                uint32_t    framesReady();
+                uint32_t    framesReady();                      // called by server only
                 bool        tryLock();
 
                 // No barriers on the following operations, so the ordering of loads/stores
diff --git a/include/utils/KeyedVector.h b/include/utils/KeyedVector.h
index 6bcdea4f..fcc3bcf 100644
--- a/include/utils/KeyedVector.h
+++ b/include/utils/KeyedVector.h
@@ -66,7 +66,7 @@
             ssize_t         indexOfKey(const KEY& key) const;
 
     /*!
-     * modifing the array
+     * modifying the array
      */
 
             VALUE&          editValueFor(const KEY& key);
diff --git a/media/jni/soundpool/SoundPool.cpp b/media/jni/soundpool/SoundPool.cpp
index df4fbb5..ceb87db 100644
--- a/media/jni/soundpool/SoundPool.cpp
+++ b/media/jni/soundpool/SoundPool.cpp
@@ -623,7 +623,7 @@
         newTrack->setVolume(leftVolume, rightVolume);
         newTrack->setLoop(0, frameCount, loop);
 
-        // From now on, AudioTrack callbacks recevieved with previous toggle value will be ignored.
+        // From now on, AudioTrack callbacks received with previous toggle value will be ignored.
         mToggle = toggle;
         mAudioTrack = newTrack;
         mPos = 0;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 2518921..db18b36 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1,4 +1,4 @@
-/* frameworks/base/media/libmedia/AudioTrack.cpp
+/*
 **
 ** Copyright 2007, The Android Open Source Project
 **
@@ -792,7 +792,7 @@
                 }
             }
         } else {
-            // Ensure that buffer alignment matches channelcount
+            // Ensure that buffer alignment matches channelCount
             int channelCount = popcount(channelMask);
             if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
                 ALOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
@@ -979,7 +979,8 @@
     if (mSharedBuffer != 0) return INVALID_OPERATION;
 
     if (ssize_t(userSize) < 0) {
-        // sanity-check. user is most-likely passing an error code.
+        // Sanity-check: user is most-likely passing an error code, and it would
+        // make the return value ambiguous (actualSize vs error).
         ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
                 buffer, userSize, userSize);
         return BAD_VALUE;
@@ -1002,8 +1003,6 @@
     do {
         audioBuffer.frameCount = userSize/frameSz;
 
-        // Calling obtainBuffer() with a negative wait count causes
-        // an (almost) infinite wait time.
         status_t err = obtainBuffer(&audioBuffer, -1);
         if (err < 0) {
             // out of buffers, return #bytes written
@@ -1093,6 +1092,9 @@
         frames = mRemainingFrames;
     }
 
+    // See description of waitCount parameter at declaration of obtainBuffer().
+    // The logic below prevents us from being stuck below at obtainBuffer()
+    // not being able to handle timed events (position, markers, loops).
     int32_t waitCount = -1;
     if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
         waitCount = 1;
@@ -1102,9 +1104,6 @@
 
         audioBuffer.frameCount = frames;
 
-        // Calling obtainBuffer() with a wait count of 1
-        // limits wait time to WAIT_PERIOD_MS. This prevents from being
-        // stuck here not being able to handle timed events (position, markers, loops).
         status_t err = obtainBuffer(&audioBuffer, waitCount);
         if (err < NO_ERROR) {
             if (err != TIMED_OUT) {
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index 3724642..a7958de 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -1,4 +1,4 @@
-/* //device/extlibs/pv/android/IAudioTrack.cpp
+/*
 **
 ** Copyright 2007, The Android Open Source Project
 **
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index 6cb10aa..54eb98a 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -791,7 +791,7 @@
 //        generators, instantiates output audio track.
 //
 //    Input:
-//        streamType:        Type of stream used for tone playback (enum AudioTrack::stream_type)
+//        streamType:        Type of stream used for tone playback
 //        volume:            volume applied to tone (0.0 to 1.0)
 //
 //    Output:
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index f1c47dd..250425b 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -1,4 +1,4 @@
-/* mediaplayer.cpp
+/*
 **
 ** Copyright 2006, The Android Open Source Project
 **
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index fb163d1..131371e 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioFlinger.cpp
+/*
 **
 ** Copyright 2007, The Android Open Source Project
 **
@@ -621,6 +621,7 @@
     }
 
     Mutex::Autolock _l(mLock);
+    // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger
     mMasterMute = muted;
     for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
@@ -3121,6 +3122,7 @@
 
 void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
 {
+    // FIXME explain this formula
     int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
     OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
                                             this,
@@ -3392,7 +3394,7 @@
 {
     // NOTE: destroyTrack_l() can remove a strong reference to this Track
     // by removing it from mTracks vector, so there is a risk that this Tracks's
-    // desctructor is called. As the destructor needs to lock mLock,
+    // destructor is called. As the destructor needs to lock mLock,
     // we must acquire a strong reference on this Track before locking mLock
     // here so that the destructor is called only when exiting this function.
     // On the other hand, as long as Track::destroy() is only called by
@@ -3998,6 +4000,7 @@
 AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
     :   RefBase(),
         mAudioFlinger(audioFlinger),
+        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
         mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
         mPid(pid)
 {
@@ -4659,7 +4662,7 @@
         }
         if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
             // do not accept frame count changes if tracks are open as the track buffer
-            // size depends on frame count and correct behavior would not be garantied
+            // size depends on frame count and correct behavior would not be guaranteed
             // if frame count is changed after track creation
             if (mActiveTrack != 0) {
                 status = INVALID_OPERATION;
@@ -6107,7 +6110,6 @@
     status_t status;
 
     Mutex::Autolock _l(mLock);
-    // First handle in mHandles has highest priority and controls the effect module
     int priority = handle->priority();
     size_t size = mHandles.size();
     sp<EffectHandle> h;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index aa0ee76..48a23fa 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioFlinger.h
+/*
 **
 ** Copyright 2007, The Android Open Source Project
 **
@@ -290,6 +290,8 @@
             enum track_state {
                 IDLE,
                 TERMINATED,
+                // These are order-sensitive; do not change order without reviewing the impact.
+                // In particular there are assumptions about > STOPPED.
                 STOPPED,
                 RESUMING,
                 ACTIVE,
@@ -760,6 +762,9 @@
         int                             mSuspended;
         int                             mBytesWritten;
     private:
+        // mMasterMute is in both PlaybackThread and in AudioFlinger.  When a
+        // PlaybackThread needs to find out if master-muted, it checks it's local
+        // copy rather than the one in AudioFlinger.  This optimization saves a lock.
         bool                            mMasterMute;
     protected:
         SortedVector< wp<Track> >       mActiveTracks;
@@ -853,6 +858,8 @@
     private:
         void applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp);
 
+        // volumes last sent to audio HAL with stream->set_volume()
+        // FIXME use standard representation and names
         float mLeftVolFloat;
         float mRightVolFloat;
         uint16_t mLeftVolShort;
@@ -900,6 +907,7 @@
 
     friend class AudioBuffer;
 
+    // server side of the client's IAudioTrack
     class TrackHandle : public android::BnAudioTrack {
     public:
                             TrackHandle(const sp<PlaybackThread::Track>& track);
@@ -1024,6 +1032,7 @@
                 ssize_t                             mBytesRead;
     };
 
+    // server side of the client's IAudioRecord
     class RecordHandle : public android::BnAudioRecord {
     public:
         RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
@@ -1152,6 +1161,7 @@
         status_t            mStatus;    // initialization status
         effect_state        mState;     // current activation state
         Vector< wp<EffectHandle> > mHandles;    // list of client handles
+                    // First handle in mHandles has highest priority and controls the effect module
         uint32_t mMaxDisableWaitCnt;    // maximum grace period before forcing an effect off after
                                         // sending disable command.
         uint32_t mDisableWaitCnt;       // current process() calls count during disable period.
@@ -1377,6 +1387,7 @@
             hwDev(dev), stream(in) {}
     };
 
+    // for mAudioSessionRefs only
     struct AudioSessionRef {
         // FIXME rename parameter names when fields get "m" prefix
         AudioSessionRef(int sessionid_, pid_t pid_) :
@@ -1432,10 +1443,11 @@
                 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> >    mRecordThreads;
 
                 DefaultKeyedVector< pid_t, sp<NotificationClient> >    mNotificationClients;
-                volatile int32_t                    mNextUniqueId;
+                volatile int32_t                    mNextUniqueId;  // updated by android_atomic_inc
                 audio_mode_t                        mMode;
                 bool                                mBtNrecIsOff;
 
+                // protected by mLock
                 Vector<AudioSessionRef*> mAudioSessionRefs;
 
                 float       masterVolume_l() const  { return mMasterVolume; }
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 191520a..cb7678b 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioMixer.cpp
+/*
 **
 ** Copyright 2007, The Android Open Source Project
 **
@@ -961,7 +961,12 @@
 // one track, 16 bits stereo without resampling is the most common case
 void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
 {
+    // This method is only called when state->enabledTracks has exactly
+    // one bit set.  The asserts below would verify this, but are commented out
+    // since the whole point of this method is to optimize performance.
+    //assert(0 != state->enabledTracks);
     const int i = 31 - __builtin_clz(state->enabledTracks);
+    //assert((1 << i) == state->enabledTracks);
     const track_t& t = state->tracks[i];
 
     AudioBufferProvider::Buffer& b(t.buffer);
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index c709686..c956918 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioMixer.h
+/*
 **
 ** Copyright 2007, The Android Open Source Project
 **