am 18f86140: Merge "Fix underruns when fast track denied due to SRC" into klp-dev (11454092) · Commits · e / os / android_frameworks_av

include/private/media/AudioTrackShared.h

+0 −2

Original line number	Diff line number	Diff line
		@@ -383,8 +383,6 @@ public:
		protected:
		size_t mAvailToClient; // estimated frames available to client prior to releaseBuffer()
		int32_t mFlush; // our copy of cblk->u.mStreaming.mFlush, for streaming output only
		private:
		bool mDeferWake; // whether another releaseBuffer() is expected soon
		};

		// Proxy used by AudioFlinger for servicing AudioTrack

media/libmedia/AudioTrack.cpp

+35 −32

Original line number	Diff line number	Diff line
		@@ -815,12 +815,29 @@ status_t AudioTrack::createTrack_l(
		return NO_INIT;
		}

		// Not all of these values are needed under all conditions, but it is easier to get them all

		uint32_t afLatency;
		if ((status = AudioSystem::getLatency(output, streamType, &afLatency)) != NO_ERROR) {
		status = AudioSystem::getLatency(output, streamType, &afLatency);
		if (status != NO_ERROR) {
		ALOGE("getLatency(%d) failed status %d", output, status);
		return NO_INIT;
		}

		size_t afFrameCount;
		status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
		if (status != NO_ERROR) {
		ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
		return NO_INIT;
		}

		uint32_t afSampleRate;
		status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
		if (status != NO_ERROR) {
		ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status);
		return NO_INIT;
		}

		// Client decides whether the track is TIMED (see below), but can only express a preference
		// for FAST. Server will perform additional tests.
		if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
		@@ -836,6 +853,14 @@ status_t AudioTrack::createTrack_l(
		}
		ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);

		// The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
		// n = 1 fast track; nBuffering is ignored
		// n = 2 normal track, no sample rate conversion
		// n = 3 normal track, with sample rate conversion
		// (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)
		// n > 3 very high latency or very small notification interval; nBuffering is ignored
		const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3;

		mNotificationFramesAct = mNotificationFramesReq;

		if (!audio_is_linear_pcm(format)) {
		@@ -844,13 +869,6 @@ status_t AudioTrack::createTrack_l(
		// Same comment as below about ignoring frameCount parameter for set()
		frameCount = sharedBuffer->size();
		} else if (frameCount == 0) {
		size_t afFrameCount;
		status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
		if (status != NO_ERROR) {
		ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType,
		status);
		return NO_INIT;
		}
		frameCount = afFrameCount;
		}
		if (mNotificationFramesAct != frameCount) {
		@@ -880,26 +898,13 @@ status_t AudioTrack::createTrack_l(
		} else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {

		// FIXME move these calculations and associated checks to server
		uint32_t afSampleRate;
		status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
		if (status != NO_ERROR) {
		ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType,
		status);
		return NO_INIT;
		}
		size_t afFrameCount;
		status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
		if (status != NO_ERROR) {
		ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
		return NO_INIT;
		}

		// Ensure that buffer depth covers at least audio hardware latency
		uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
		ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d",
		afFrameCount, minBufCount, afSampleRate, afLatency);
		if (minBufCount <= 2) {
		minBufCount = sampleRate == afSampleRate ? 2 : 3;
		if (minBufCount <= nBuffering) {
		minBufCount = nBuffering;
		}

		size_t minFrameCount = (afFrameCountsampleRateminBufCount)/afSampleRate;
		@@ -909,18 +914,16 @@ status_t AudioTrack::createTrack_l(

		if (frameCount == 0) {
		frameCount = minFrameCount;
		}
		// Make sure that application is notified with sufficient margin
		// before underrun
		if (mNotificationFramesAct == 0 \|\| mNotificationFramesAct > frameCount/2) {
		mNotificationFramesAct = frameCount/2;
		}
		if (frameCount < minFrameCount) {
		} else if (frameCount < minFrameCount) {
		// not ALOGW because it happens all the time when playing key clicks over A2DP
		ALOGV("Minimum buffer size corrected from %d to %d",
		frameCount, minFrameCount);
		frameCount = minFrameCount;
		}
		// Make sure that application is notified with sufficient margin before underrun
		if (mNotificationFramesAct == 0 \|\| mNotificationFramesAct > frameCount/nBuffering) {
		mNotificationFramesAct = frameCount/nBuffering;
		}

		} else {
		// For fast tracks, the frame count calculations and checks are done by server
		@@ -1001,8 +1004,8 @@ status_t AudioTrack::createTrack_l(
		flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
		mFlags = flags;
		if (sharedBuffer == 0) {
		if (mNotificationFramesAct == 0 \|\| mNotificationFramesAct > frameCount/2) {
		mNotificationFramesAct = frameCount/2;
		if (mNotificationFramesAct == 0 \|\| mNotificationFramesAct > frameCount/nBuffering) {
		mNotificationFramesAct = frameCount/nBuffering;
		}
		}
		}

media/libmedia/AudioTrackShared.cpp

+2 −5

Original line number	Diff line number	Diff line
		@@ -481,7 +481,7 @@ size_t StaticAudioTrackClientProxy::getBufferPosition()
		ServerProxy::ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
		size_t frameSize, bool isOut, bool clientInServer)
		: Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer),
		mAvailToClient(0), mFlush(0), mDeferWake(false)
		mAvailToClient(0), mFlush(0)
		{
		}

		@@ -559,9 +559,6 @@ status_t ServerProxy::obtainBuffer(Buffer* buffer)
		&((char ) mBuffers)[(mIsOut ? front : rear) mFrameSize] : NULL;
		buffer->mNonContig = availToServer - part1;
		mUnreleased = part1;
		// optimization to avoid waking up the client too early
		// FIXME need to test for recording
		mDeferWake = part1 < ask && availToServer >= ask;
		return part1 > 0 ? NO_ERROR : WOULD_BLOCK;
		}
		no_init:
		@@ -607,7 +604,7 @@ void ServerProxy::releaseBuffer(Buffer* buffer)
		minimum = half;
		}
		// FIXME AudioRecord wakeup needs to be optimized; it currently wakes up client every time
		if (!mIsOut \|\| (!mDeferWake && mAvailToClient + stepCount >= minimum)) {
		if (!mIsOut \|\| (mAvailToClient + stepCount >= minimum)) {
		ALOGV("mAvailToClient=%u stepCount=%u minimum=%u", mAvailToClient, stepCount, minimum);
		int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
		if (!(old & CBLK_FUTEX_WAKE)) {