Merge "Fix theoretical race using TrackBase::sampleRate()" (a30e7589) · Commits · e / os / android_frameworks_av

services/audioflinger/Threads.cpp

+4 −3

Original line number	Diff line number	Diff line
		@@ -2703,7 +2703,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
		}

		for (size_t i=0 ; i<count ; i++) {
		sp<Track> t = mActiveTracks[i].promote();
		const sp<Track> t = mActiveTracks[i].promote();
		if (t == 0) {
		continue;
		}
		@@ -2872,11 +2872,12 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
		// hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
		// during last round
		size_t desiredFrames;
		if (t->sampleRate() == mSampleRate) {
		uint32_t sr = track->sampleRate();
		if (sr == mSampleRate) {
		desiredFrames = mNormalFrameCount;
		} else {
		// +1 for rounding and +1 for additional sample needed for interpolation
		desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
		desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
		// add frames already consumed but not yet released by the resampler
		// because cblk->framesReady() will include these frames
		desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());

+4 −2

Original line number	Diff line number	Diff line
		@@ -1172,10 +1172,12 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
		}
		}

		uint32_t sr = sampleRate();

		// adjust the head buffer's PTS to reflect the portion of the head buffer
		// that has already been consumed
		int64_t effectivePTS = headLocalPTS +
		((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate());
		((head.position() / mFrameSize) * mLocalTimeFreq / sr);

		// Calculate the delta in samples between the head of the input buffer
		// queue and the start of the next output buffer that will be written.
		@@ -1207,7 +1209,7 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
		// the current output position is within this threshold, then we will
		// concatenate the next input samples to the previous output
		const int64_t kSampleContinuityThreshold =
		(static_cast<int64_t>(sampleRate()) << 32) / 250;
		(static_cast<int64_t>(sr) << 32) / 250;

		// if this is the first buffer of audio that we're emitting from this track
		// then it should be almost exactly on time.