Commit 8d314b70 authored Apr 19, 2012 by John Grossman

TimedAudioTrack: Fix continuity threshold handling.



Hand merge from ics-aah

> TimedAudioTrack: Fix continuity threshold handling.
>
> Fix issues with continuity threshold handling; notably
> + If the steady-state continuity threshold is exceeded, be sure to
>   clear the on-time flag.  Failure to do this will result in the
>   system picking a new mix point which simply satisfies the
>   steady-state continuity threshold instead of the startup threshold.
>   Since we are putting a discontinuity in presentation anyway, we
>   really want to pick a perfect point, not just an OK point.
> + Tighten the steady-state continuity threshold.  It was currently set
>   to 100mSec which is enormous.  4mSec (the new setting) is much more
>   appropriate.  On systems with a VCXO (like tungsten) this should
>   never be wrong by more than a sample.  If TimedAudioTracks are ever
>   to be used on VCXO-less systems, this threshold should probably be a
>   a parameter configurable by applications on a track by track basis
>   so they can make the tradeoff between allowed error and frequency of
>   disruptive corrections.
> + Reset the on-time flag if the mixer provides no PTS during a mix
>   operation.  This makes for a convenient way for the HAL to reset
>   timed tracks when it makes changes for delay compensation across
>   multiple outputs.
>
> Change-Id: I2cb23de5a3d1f75618abc1c8ab903db883837aa8
> Signed-off-by: John Grossman <johngro@google.com>

Change-Id: Ibd28c9d290494b0b19eb01caf2d9bfdef606a9b4
Signed-off-by: John Grossman <johngro@google.com>

parent 21e8c50b

services/audioflinger/AudioFlinger.cpp

+10 −3

Original line number	Diff line number	Diff line
		@@ -4111,6 +4111,7 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
		if (pts == AudioBufferProvider::kInvalidPTS) {
		buffer->raw = 0;
		buffer->frameCount = 0;
		mTimedAudioOutputOnTime = false;
		return INVALID_OPERATION;
		}

		@@ -4203,7 +4204,7 @@ 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) / 10;
		(static_cast<int64_t>(sampleRate()) << 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.
		@@ -4218,7 +4219,13 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
		ALOGVV("*** on time: head.pos=%d frameCount=%u",
		head.position(), buffer->frameCount);
		return NO_ERROR;
		} else if (sampleDelta > 0) {
		}

		// Looks like our output is not on time. Reset our on timed status.
		// Next time we mix samples from our input queue, then should be within
		// the StartupThreshold.
		mTimedAudioOutputOnTime = false;
		if (sampleDelta > 0) {
		// the gap between the current output position and the proper start of
		// the next input sample is too big, so fill it with silence
		uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;