Update audio comments

Change-Id: Ie7504d0ddb252f7e4d4f99ed0b44cfc7b1049816

2 files changed, 14 insertions, 10 deletions

diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1a37f4f..14f74b5 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -169,8 +169,8 @@ static const int kPriorityFastMixer = 3;
 // for the track.  The client then sub-divides this into smaller buffers for its use.
 // Currently the client uses double-buffering by default, but doesn't tell us about that.
 // So for now we just assume that client is double-buffered.
-// FIXME It would be better for client to tell us whether it wants double-buffering or N-buffering,
-// so we could allocate the right amount of memory.
+// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or
+// N-buffering, so AudioFlinger could allocate the right amount of memory.
 // See the client's minBufCount and mNotificationFramesAct calculations for details.
 static const int kFastTrackMultiplier = 2;
 
@@ -258,11 +258,11 @@ void AudioFlinger::onFirstRef()
 AudioFlinger::~AudioFlinger()
 {
     while (!mRecordThreads.isEmpty()) {
-        // closeInput() will remove first entry from mRecordThreads
+        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
         closeInput_nonvirtual(mRecordThreads.keyAt(0));
     }
     while (!mPlaybackThreads.isEmpty()) {
-        // closeOutput() will remove first entry from mPlaybackThreads
+        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
         closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
     }
 
@@ -1134,7 +1134,7 @@ sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId,
 
 AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
         audio_devices_t device, type_t type)
-    :   Thread(false),
+    :   Thread(false /*canCallJava*/),
         mType(type),
         mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0),
         // mChannelMask
@@ -1142,6 +1142,7 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio
         mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
         mParamStatus(NO_ERROR),
         mStandby(false), mDevice(device), mId(id),
+        // mName will be set by concrete (non-virtual) subclass
         mDeathRecipient(new PMDeathRecipient(this))
 {
 }
@@ -6097,7 +6098,7 @@ bool AudioFlinger::RecordThread::threadLoop()
                     if (mChannelCount == 1 && mReqChannelCount == 1) {
                         framesOut >>= 1;
                     }
-                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
+                    mResampler->resample(mRsmpOutBuffer, framesOut, this /* AudioBufferProvider* */);
                     // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
                     // are 32 bit aligned which should be always true.
                     if (mChannelCount == 2 && mReqChannelCount == 1) {
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index b4aefc1..4723cd9 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -454,8 +454,9 @@ private:
             /*const*/ sp<Client> mClient;   // see explanation at ~TrackBase() why not const
             sp<IMemory>         mCblkMemory;
             audio_track_cblk_t* mCblk;
-            void*               mBuffer;
-            void*               mBufferEnd;
+            void*               mBuffer;    // start of track buffer, typically in shared memory
+            void*               mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize
+                                            //   is based on mChannelCount and 16-bit samples
             uint32_t            mFrameCount;
             // we don't really need a lock for these
             track_state         mState;
@@ -1364,6 +1365,7 @@ private:
 
     // record thread
     class RecordThread : public ThreadBase, public AudioBufferProvider
+                            // derives from AudioBufferProvider interface for use by resampler
     {
     public:
 
@@ -1420,7 +1422,7 @@ private:
         void        dumpInternals(int fd, const Vector<String16>& args);
         void        dumpTracks(int fd, const Vector<String16>& args);
 
-        // Thread
+        // Thread virtuals
         virtual bool        threadLoop();
         virtual status_t    readyToRun();
 
@@ -1968,9 +1970,10 @@ mutable Mutex               mLock;      // mutex for process, commands and handl
                 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> >  mPlaybackThreads;
                 stream_type_t                       mStreamTypes[AUDIO_STREAM_CNT];
 
-                // both are protected by mLock
+                // member variables below are protected by mLock
                 float                               mMasterVolume;
                 bool                                mMasterMute;
+                // end of variables protected by mLock
 
                 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> >    mRecordThreads;