Update audio comments

Change-Id: Ie7504d0ddb252f7e4d4f99ed0b44cfc7b1049816

diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 26a25b0..34108b3 100644 (file)
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -547,7 +547,7 @@ public:
     status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer);
 
     /* queue a buffer obtained via allocateTimedBuffer for playback at the
-       given timestamp.  PTS units a microseconds on the media time timeline.
+       given timestamp.  PTS units are microseconds on the media time timeline.
        The media time transform (set with setMediaTimeTransform) set by the
        audio producer will handle converting from media time to local time
        (perhaps going through the common time timeline in the case of

diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 73d396e..362d022 100644 (file)
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1144,7 +1144,7 @@ status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
 
     // If the track is not invalid already, try to allocate a buffer.  alloc
     // fails indicating that the server is dead, flag the track as invalid so
-    // we can attempt to restore in in just a bit.
+    // we can attempt to restore in just a bit.
     if (!(mCblk->flags & CBLK_INVALID_MSK)) {
         result = mAudioTrack->allocateTimedBuffer(size, buffer);
         if (result == DEAD_OBJECT) {

diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1a37f4f..14f74b5 100644 (file)
--- 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 (file)
--- 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;