Merge "Rename mMixBuffer to mSinkBuffer in AudioFlinger" (625b57c5) · Commits · e / os / android_frameworks_av

services/audioflinger/Threads.cpp

+25 −25

Original line number	Diff line number	Diff line
		@@ -1066,7 +1066,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge
		audio_devices_t device,
		type_t type)
		: ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
		mNormalFrameCount(0), mMixBuffer(NULL),
		mNormalFrameCount(0), mSinkBuffer(NULL),
		mSuspended(0), mBytesWritten(0),
		mActiveTracksGeneration(0),
		// mStreamTypes[] initialized in constructor body
		@@ -1125,7 +1125,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge
		AudioFlinger::PlaybackThread::~PlaybackThread()
		{
		mAudioFlinger->unregisterWriter(mNBLogWriter);
		delete[] mMixBuffer;
		delete[] mSinkBuffer;
		}

		void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
		@@ -1210,7 +1210,7 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>&
		fdprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites);
		fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no");
		fdprintf(fd, " Suspend count: %d\n", mSuspended);
		fdprintf(fd, " Mix buffer : %p\n", mMixBuffer);
		fdprintf(fd, " Sink buffer : %p\n", mSinkBuffer);
		fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);

		dumpBase(fd, args);
		@@ -1758,11 +1758,11 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l()
		ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
		mNormalFrameCount);

		delete[] mMixBuffer;
		delete[] mSinkBuffer;
		size_t normalBufferSize = mNormalFrameCount * mFrameSize;
		// For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1)
		mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1];
		memset(mMixBuffer, 0, normalBufferSize);
		// For historical reasons mSinkBuffer is int16_t[], but mFrameSize can be odd (such as 1)
		mSinkBuffer = new int16_t[(normalBufferSize + 1) >> 1];
		memset(mSinkBuffer, 0, normalBufferSize);

		// force reconfiguration of effect chains and engines to take new buffer size and audio
		// parameters into account
		@@ -1958,7 +1958,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
		(pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
		}
		}
		ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count);
		ssize_t framesWritten = mNormalSink->write(mSinkBuffer + offset, count);
		ATRACE_END();
		if (framesWritten > 0) {
		bytesWritten = framesWritten << mBitShift;
		@@ -1987,7 +1987,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
		// FIXME We should have an implementation of timestamps for direct output threads.
		// They are used e.g for multichannel PCM playback over HDMI.
		bytesWritten = mOutput->stream->write(mOutput->stream,
		(char *)mMixBuffer + offset, mBytesRemaining);
		(char *)mSinkBuffer + offset, mBytesRemaining);
		if (mUseAsyncWrite &&
		((bytesWritten < 0) \|\| (bytesWritten == (ssize_t)mBytesRemaining))) {
		// do not wait for async callback in case of error of full write
		@@ -2068,13 +2068,13 @@ void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamTy
		status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
		{
		int session = chain->sessionId();
		int16_t *buffer = mMixBuffer;
		int16_t *buffer = mSinkBuffer;
		bool ownsBuffer = false;

		ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
		if (session > 0) {
		// Only one effect chain can be present in direct output thread and it uses
		// the mix buffer as input
		// the sink buffer as input
		if (mType != DIRECT) {
		size_t numSamples = mNormalFrameCount * mChannelCount;
		buffer = new int16_t[numSamples];
		@@ -2108,7 +2108,7 @@ status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& c
		}

		chain->setInBuffer(buffer, ownsBuffer);
		chain->setOutBuffer(mMixBuffer);
		chain->setOutBuffer(mSinkBuffer);
		// Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
		// chains list in order to be processed last as it contains output stage effects
		// Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
		@@ -2158,7 +2158,7 @@ size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>&
		for (size_t i = 0; i < mTracks.size(); ++i) {
		sp<Track> track = mTracks[i];
		if (session == track->sessionId()) {
		track->setMainBuffer(mMixBuffer);
		track->setMainBuffer(mSinkBuffer);
		chain->decTrackCnt();
		}
		}
		@@ -2861,7 +2861,7 @@ void AudioFlinger::MixerThread::threadLoop_sleepTime()
		sleepTime = idleSleepTime;
		}
		} else if (mBytesWritten != 0 \|\| (mMixerStatus == MIXER_TRACKS_ENABLED)) {
		memset(mMixBuffer, 0, mixBufferSize);
		memset(mSinkBuffer, 0, mixBufferSize);
		sleepTime = 0;
		ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
		"anticipated start");
		@@ -3109,10 +3109,10 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac

		mixedTracks++;

		// track->mainBuffer() != mMixBuffer means there is an effect chain
		// track->mainBuffer() != mSinkBuffer means there is an effect chain
		// connected to the track
		chain.clear();
		if (track->mainBuffer() != mMixBuffer) {
		if (track->mainBuffer() != mSinkBuffer) {
		chain = getEffectChain_l(track->sessionId());
		// Delegate volume control to effect in track effect chain if needed
		if (chain != 0) {
		@@ -3355,13 +3355,13 @@ track_is_ready: ;
		// remove all the tracks that need to be...
		removeTracks_l(*tracksToRemove);

		// mix buffer must be cleared if all tracks are connected to an
		// sink buffer must be cleared if all tracks are connected to an
		// effect chain as in this case the mixer will not write to
		// mix buffer and track effects will accumulate into it
		// sink buffer and track effects will accumulate into it
		if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) \|\|
		(mixedTracks == 0 && fastTracks > 0))) {
		// FIXME as a performance optimization, should remember previous zero status
		memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
		memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
		}

		// if any fast tracks, then status is ready
		@@ -3749,7 +3749,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep
		void AudioFlinger::DirectOutputThread::threadLoop_mix()
		{
		size_t frameCount = mFrameCount;
		int8_t curBuf = (int8_t )mMixBuffer;
		int8_t curBuf = (int8_t )mSinkBuffer;
		// output audio to hardware
		while (frameCount) {
		AudioBufferProvider::Buffer buffer;
		@@ -3764,7 +3764,7 @@ void AudioFlinger::DirectOutputThread::threadLoop_mix()
		curBuf += buffer.frameCount * mFrameSize;
		mActiveTrack->releaseBuffer(&buffer);
		}
		mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer;
		mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer;
		sleepTime = 0;
		standbyTime = systemTime() + standbyDelay;
		mActiveTrack.clear();
		@@ -3779,7 +3779,7 @@ void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
		sleepTime = idleSleepTime;
		}
		} else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
		memset(mMixBuffer, 0, mFrameCount * mFrameSize);
		memset(mSinkBuffer, 0, mFrameCount * mFrameSize);
		sleepTime = 0;
		}
		}
		@@ -4306,7 +4306,7 @@ void AudioFlinger::DuplicatingThread::threadLoop_mix()
		if (outputsReady(outputTracks)) {
		mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
		} else {
		memset(mMixBuffer, 0, mixBufferSize);
		memset(mSinkBuffer, 0, mixBufferSize);
		}
		sleepTime = 0;
		writeFrames = mNormalFrameCount;
		@@ -4325,7 +4325,7 @@ void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
		} else if (mBytesWritten != 0) {
		if (mMixerStatus == MIXER_TRACKS_ENABLED) {
		writeFrames = mNormalFrameCount;
		memset(mMixBuffer, 0, mixBufferSize);
		memset(mSinkBuffer, 0, mixBufferSize);
		} else {
		// flush remaining overflow buffers in output tracks
		writeFrames = 0;
		@@ -4337,7 +4337,7 @@ void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
		ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
		{
		for (size_t i = 0; i < outputTracks.size(); i++) {
		outputTracks[i]->write(mMixBuffer, writeFrames);
		outputTracks[i]->write(mSinkBuffer, writeFrames);
		}
		mStandby = false;
		return (ssize_t)mixBufferSize;

services/audioflinger/Threads.h

+3 −2

Original line number	Diff line number	Diff line
		@@ -450,7 +450,8 @@ public:
		virtual String8 getParameters(const String8& keys);
		virtual void audioConfigChanged_l(int event, int param = 0);
		status_t getRenderPosition(uint32_t halFrames, uint32_t dspFrames);
		int16_t *mixBuffer() const { return mMixBuffer; };
		// TODO: rename mixBuffer() to sinkBuffer() or try to remove external use.
		int16_t *mixBuffer() const { return mSinkBuffer; };

		virtual void detachAuxEffect_l(int effectId);
		status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
		@@ -481,7 +482,7 @@ protected:
		// updated by readOutputParameters_l()
		size_t mNormalFrameCount; // normal mixer and effects

		int16_t* mMixBuffer; // frame size aligned mix buffer
		int16_t* mSinkBuffer; // frame size aligned sink buffer

		// suspend count, > 0 means suspended. While suspended, the thread continues to pull from
		// tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle