Merge "Add MixerBuffer for accumulation of float audio" (b4429211) · Commits · e / os / android_frameworks_av

services/audioflinger/Threads.cpp

+90 −10

Original line number	Diff line number	Diff line
		@@ -1067,6 +1067,11 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge
		type_t type)
		: ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
		mNormalFrameCount(0), mSinkBuffer(NULL),
		mMixerBufferEnabled(false),
		mMixerBuffer(NULL),
		mMixerBufferSize(0),
		mMixerBufferFormat(AUDIO_FORMAT_INVALID),
		mMixerBufferValid(false),
		mSuspended(0), mBytesWritten(0),
		mActiveTracksGeneration(0),
		// mStreamTypes[] initialized in constructor body
		@@ -1126,6 +1131,7 @@ AudioFlinger::PlaybackThread::~PlaybackThread()
		{
		mAudioFlinger->unregisterWriter(mNBLogWriter);
		delete[] mSinkBuffer;
		free(mMixerBuffer);
		}

		void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
		@@ -1211,6 +1217,7 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>&
		fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no");
		fdprintf(fd, " Suspend count: %d\n", mSuspended);
		fdprintf(fd, " Sink buffer : %p\n", mSinkBuffer);
		fdprintf(fd, " Mixer buffer: %p\n", mMixerBuffer);
		fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);

		dumpBase(fd, args);
		@@ -1764,6 +1771,17 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l()
		mSinkBuffer = new int16_t[(normalBufferSize + 1) >> 1];
		memset(mSinkBuffer, 0, normalBufferSize);

		// We resize the mMixerBuffer according to the requirements of the sink buffer which
		// drives the output.
		free(mMixerBuffer);
		mMixerBuffer = NULL;
		if (mMixerBufferEnabled) {
		mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT.
		mMixerBufferSize = mNormalFrameCount * mChannelCount
		* audio_bytes_per_sample(mMixerBufferFormat);
		(void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
		}

		// force reconfiguration of effect chains and engines to take new buffer size and audio
		// parameters into account
		// Note that mLock is not held when readOutputParameters_l() is called from the constructor
		@@ -2355,6 +2373,23 @@ bool AudioFlinger::PlaybackThread::threadLoop()
		if (mMixerStatus == MIXER_TRACKS_READY) {
		// threadLoop_mix() sets mCurrentWriteLength
		threadLoop_mix();

		// Merge mMixerBuffer data into mSinkBuffer
		// This is done pre-effects computation; if effects change to
		// support higher precision, this needs to move.
		//
		// mMixerBufferValid is only set true by MixerThread::prepareTracks_l().
		if (mMixerBufferValid) {
		if (mMixerBufferFormat == AUDIO_FORMAT_PCM_FLOAT) {
		memcpy_to_i16_from_float(mSinkBuffer,
		reinterpret_cast<float*>(mMixerBuffer),
		mNormalFrameCount * mChannelCount);
		} else { // mMixerBufferFormat == AUDIO_FORMAT_PCM_16_BIT
		memcpy(mSinkBuffer,
		mMixerBuffer,
		mNormalFrameCount * mChannelCount * sizeof(int16_t));
		}
		}
		} else if ((mMixerStatus != MIXER_DRAIN_TRACK)
		&& (mMixerStatus != MIXER_DRAIN_ALL)) {
		// threadLoop_sleepTime sets sleepTime to 0 if data
		@@ -2908,6 +2943,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
		state = sq->begin();
		}

		mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found.

		for (size_t i=0 ; i<count ; i++) {
		const sp<Track> t = mActiveTracks[i].promote();
		if (t == 0) {
		@@ -3109,10 +3146,11 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac

		mixedTracks++;

		// track->mainBuffer() != mSinkBuffer means there is an effect chain
		// connected to the track
		// track->mainBuffer() != mSinkBuffer or mMixerBuffer means
		// there is an effect chain connected to the track
		chain.clear();
		if (track->mainBuffer() != mSinkBuffer) {
		if (track->mainBuffer() != mSinkBuffer &&
		track->mainBuffer() != mMixerBuffer) {
		chain = getEffectChain_l(track->sessionId());
		// Delegate volume control to effect in track effect chain if needed
		if (chain != 0) {
		@@ -3238,10 +3276,40 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
		AudioMixer::RESAMPLE,
		AudioMixer::SAMPLE_RATE,
		(void *)(uintptr_t)reqSampleRate);
		/*
		* Select the appropriate output buffer for the track.
		*
		* For tracks with effects, only mSinkBuffer can be used (at this time).
		*
		* Other tracks can use mMixerBuffer for higher precision
		* channel accumulation. If this buffer is enabled
		* (mMixerBufferEnabled true), then selected tracks will accumulate
		* into it.
		*
		*/
		if (mMixerBufferEnabled
		&& (track->mainBuffer() == mSinkBuffer
		\|\| track->mainBuffer() == mMixerBuffer)) {
		mAudioMixer->setParameter(
		name,
		AudioMixer::TRACK,
		AudioMixer::SINK_FORMAT, (void *)mMixerBufferFormat);
		mAudioMixer->setParameter(
		name,
		AudioMixer::TRACK,
		AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
		// TODO: override track->mainBuffer()?
		mMixerBufferValid = true;
		} else {
		mAudioMixer->setParameter(
		name,
		AudioMixer::TRACK,
		AudioMixer::SINK_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT);
		mAudioMixer->setParameter(
		name,
		AudioMixer::TRACK,
		AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
		}
		mAudioMixer->setParameter(
		name,
		AudioMixer::TRACK,
		@@ -3355,12 +3423,24 @@ track_is_ready: ;
		// remove all the tracks that need to be...
		removeTracks_l(*tracksToRemove);

		// sink buffer must be cleared if all tracks are connected to an
		// effect chain as in this case the mixer will not write to
		// sink buffer and track effects will accumulate into it
		// sink or mix buffer must be cleared if all tracks are connected to an
		// effect chain as in this case the mixer will not write to the sink or mix 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
		if (mMixerBufferValid) {
		memset(mMixerBuffer, 0, mMixerBufferSize);
		// TODO: In testing, mSinkBuffer below need not be cleared because
		// the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer
		// after mixing.
		//
		// To enforce this guarantee:
		// ((mixedTracks != 0 && mixedTracks == tracksWithEffect) \|\|
		// (mixedTracks == 0 && fastTracks > 0))
		// must imply MIXER_TRACKS_READY.
		// Later, we may clear buffers regardless, and skip much of this logic.
		}
		memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
		}

services/audioflinger/Threads.h

+23 −0

Original line number	Diff line number	Diff line
		@@ -484,6 +484,29 @@ protected:

		int16_t* mSinkBuffer; // frame size aligned sink buffer

		// Mixer Buffer (mMixerBuffer*)
		//
		// In the case of floating point or multichannel data, which is not in the
		// sink format, it is required to accumulate in a higher precision or greater channel count
		// buffer before downmixing or data conversion to the sink buffer.

		// Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer.
		bool mMixerBufferEnabled;

		// Storage, 32 byte aligned (may make this alignment a requirement later).
		// Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames.
		void* mMixerBuffer;

		// Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize.
		size_t mMixerBufferSize;

		// The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT\|16_BIT) only.
		audio_format_t mMixerBufferFormat;

		// An internal flag set to true by MixerThread::prepareTracks_l()
		// when mMixerBuffer contains valid data after mixing.
		bool mMixerBufferValid;

		// 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
		// concurrent use of both of them, so Audio Policy Service suspends one of the threads to