Update audio comments
Change-Id: Ie7504d0ddb252f7e4d4f99ed0b44cfc7b1049816
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 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
/*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 @@
// record thread
class RecordThread : public ThreadBase, public AudioBufferProvider
+ // derives from AudioBufferProvider interface for use by resampler
{
public:
@@ -1420,7 +1422,7 @@
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 @@
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;