Merge "Convert AudioFlinger mSinkBuffer to flexible format" (cea17f26) · Commits · e / os / android_frameworks_av

services/audioflinger/Threads.cpp

+35 −19

Original line number	Diff line number	Diff line
		@@ -1145,7 +1145,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge
		AudioFlinger::PlaybackThread::~PlaybackThread()
		{
		mAudioFlinger->unregisterWriter(mNBLogWriter);
		delete[] mSinkBuffer;
		free(mSinkBuffer);
		free(mMixerBuffer);
		free(mEffectBuffer);
		}
		@@ -1782,11 +1782,13 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l()
		ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount,
		mNormalFrameCount);

		delete[] mSinkBuffer;
		size_t normalBufferSize = mNormalFrameCount * mFrameSize;
		// 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);
		// mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames.
		// Originally this was int16_t[] array, need to remove legacy implications.
		free(mSinkBuffer);
		mSinkBuffer = NULL;
		const size_t sinkBufferSize = mNormalFrameCount * mChannelCount
		* audio_bytes_per_sample(mFormat);
		(void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);

		// We resize the mMixerBuffer according to the requirements of the sink buffer which
		// drives the output.
		@@ -1984,12 +1986,12 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
		mLastWriteTime = systemTime();
		mInWrite = true;
		ssize_t bytesWritten;
		const size_t offset = mCurrentWriteLength - mBytesRemaining;

		// If an NBAIO sink is present, use it to write the normal mixer's submix
		if (mNormalSink != 0) {
		#define mBitShift 2 // FIXME
		size_t count = mBytesRemaining >> mBitShift;
		size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1;
		const size_t count = mBytesRemaining / mFrameSize;

		ATRACE_BEGIN("write");
		// update the setpoint when AudioFlinger::mScreenState changes
		uint32_t screenState = AudioFlinger::mScreenState;
		@@ -2001,10 +2003,10 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
		(pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
		}
		}
		ssize_t framesWritten = mNormalSink->write(mSinkBuffer + offset, count);
		ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);
		ATRACE_END();
		if (framesWritten > 0) {
		bytesWritten = framesWritten << mBitShift;
		bytesWritten = framesWritten * mFrameSize;
		} else {
		bytesWritten = framesWritten;
		}
		@@ -2019,7 +2021,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
		// otherwise use the HAL / AudioStreamOut directly
		} else {
		// Direct output and offload threads
		size_t offset = (mCurrentWriteLength - mBytesRemaining);

		if (mUseAsyncWrite) {
		ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
		mWriteAckSequence += 2;
		@@ -2111,8 +2113,8 @@ 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 = mEffectBufferEnabled
		? reinterpret_cast<int16_t*>(mEffectBuffer) : mSinkBuffer;
		int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled
		? mEffectBuffer : mSinkBuffer);
		bool ownsBuffer = false;

		ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
		@@ -2152,8 +2154,8 @@ status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& c
		}

		chain->setInBuffer(buffer, ownsBuffer);
		chain->setOutBuffer(mEffectBufferEnabled
		? reinterpret_cast<int16_t*>(mEffectBuffer) : mSinkBuffer);
		chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled
		? mEffectBuffer : 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
		@@ -2203,7 +2205,7 @@ 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(mSinkBuffer);
		track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer));
		chain->decTrackCnt();
		}
		}
		@@ -4471,7 +4473,15 @@ void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
		ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
		{
		for (size_t i = 0; i < outputTracks.size(); i++) {
		outputTracks[i]->write(mSinkBuffer, writeFrames);
		// We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT
		// for delivery downstream as needed. This in-place conversion is safe as
		// AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format
		// (AUDIO_FORMAT_PCM_8_BIT is not allowed here).
		if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
		memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT,
		mSinkBuffer, mFormat, writeFrames * mChannelCount);
		}
		outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames);
		}
		mStandby = false;
		return (ssize_t)mSinkBufferSize;
		@@ -4500,10 +4510,16 @@ void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
		Mutex::Autolock _l(mLock);
		// FIXME explain this formula
		size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
		// OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat
		// due to current usage case and restrictions on the AudioBufferProvider.
		// Actual buffer conversion is done in threadLoop_write().
		//
		// TODO: This may change in the future, depending on multichannel
		// (and non int16_t*) support on AF::PlaybackThread::OutputTrack
		OutputTrack *outputTrack = new OutputTrack(thread,
		this,
		mSampleRate,
		mFormat,
		AUDIO_FORMAT_PCM_16_BIT,
		mChannelMask,
		frameCount,
		IPCThreadState::self()->getCallingUid());

services/audioflinger/Threads.h

+6 −3

Original line number	Diff line number	Diff line
		@@ -450,8 +450,11 @@ 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);
		// TODO: rename mixBuffer() to sinkBuffer() or try to remove external use.
		int16_t *mixBuffer() const { return mSinkBuffer; };
		// FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency.
		// Consider also removing and passing an explicit mMainBuffer initialization
		// parameter to AF::PlaybackThread::Track::Track().
		int16_t *mixBuffer() const {
		return reinterpret_cast<int16_t *>(mSinkBuffer); };

		virtual void detachAuxEffect_l(int effectId);
		status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
		@@ -482,7 +485,7 @@ protected:
		// updated by readOutputParameters_l()
		size_t mNormalFrameCount; // normal mixer and effects

		int16_t* mSinkBuffer; // frame size aligned sink buffer
		void* mSinkBuffer; // frame size aligned sink buffer

		// TODO:
		// Rearrange the buffer info into a struct/class with