Merge "Update audio comments" into jb-mr1-dev (b3428f7f) · Commits · e / os / android_frameworks_av

include/media/AudioTrack.h

+1 −1

Original line number	Diff line number	Diff line
		@@ -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

media/libmedia/AudioTrack.cpp

+1 −1

Original line number	Diff line number	Diff line
		@@ -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) {

services/audioflinger/AudioFlinger.cpp

+7 −6

Original line number	Diff line number	Diff line
		@@ -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) {

services/audioflinger/AudioFlinger.h

+7 −4

Original line number	Diff line number	Diff line
		@@ -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;