Obey audio timestamps if they don't deviate by more than some threshold.

Previously we would ignore audio timestamps because they are not
contiguous in a sample-accurate way.

However with bugs like #1833 we do need to obey large discontinuities
in audio timestamps, otherwise we get large sync errors.

Here we change timestamp handling to ignore small discontinuities
in timestamps but not larger ones.

1 files changed, 34 insertions, 8 deletions

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index a5e86f22b..051fb4dd8 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -48,25 +48,51 @@ AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr<const AudioContent> cont
 	}
 }
 
+/** @param time_already_delayed true if the delay should not be added to time */
 void
-AudioDecoder::emit (shared_ptr<const Film> film, AudioStreamPtr stream, shared_ptr<const AudioBuffers> data, ContentTime time)
+AudioDecoder::emit (shared_ptr<const Film> film, AudioStreamPtr stream, shared_ptr<const AudioBuffers> data, ContentTime time, bool time_already_delayed)
 {
 	if (ignore ()) {
 		return;
 	}
 
+	/* Amount of error we will tolerate on audio timestamps; see comment below.
+	 * We'll use 1 24fps video frame at 48kHz as this seems to be roughly how
+	 * ffplay does it.
+	 */
+	static Frame const slack_frames = 48000 / 24;
+
+	int const resampled_rate = _content->resampled_frame_rate(film);
+	if (!time_already_delayed) {
+		time += ContentTime::from_seconds (_content->delay() / 1000.0);
+	}
+
+	bool reset = false;
 	if (_positions[stream] == 0) {
 		/* This is the first data we have received since initialisation or seek.  Set
 		   the position based on the ContentTime that was given.  After this first time
-		   we just count samples, as it seems that ContentTimes are unreliable from
-		   FFmpegDecoder (not quite continuous; perhaps due to some rounding error).
+		   we just count samples unless the timestamp is more than slack_frames away
+		   from where we think it should be.  This is because ContentTimes seem to be
+		   slightly unreliable from FFmpegDecoder (i.e. not sample accurate), but we still
+		   need to obey them sometimes otherwise we get sync problems such as #1833.
 		*/
 		if (_content->delay() > 0) {
 			/* Insert silence to give the delay */
 			silence (_content->delay ());
 		}
-		time += ContentTime::from_seconds (_content->delay() / 1000.0);
-		_positions[stream] = time.frames_round (_content->resampled_frame_rate(film));
+		reset = true;
+	} else if (std::abs(_positions[stream] - time.frames_round(resampled_rate)) > slack_frames) {
+		reset = true;
+		LOG_GENERAL (
+			"Reset audio position: was %1, new data at %2, slack: %3 frames",
+			_positions[stream],
+			time.frames_round(resampled_rate),
+			std::abs(_positions[stream] - time.frames_round(resampled_rate))
+			);
+	}
+
+	if (reset) {
+		_positions[stream] = time.frames_round (resampled_rate);
 	}
 
 	shared_ptr<Resampler> resampler;
@@ -74,15 +100,15 @@ AudioDecoder::emit (shared_ptr<const Film> film, AudioStreamPtr stream, shared_p
 	if (i != _resamplers.end ()) {
 		resampler = i->second;
 	} else {
-		if (stream->frame_rate() != _content->resampled_frame_rate(film)) {
+		if (stream->frame_rate() != resampled_rate) {
 			LOG_GENERAL (
 				"Creating new resampler from %1 to %2 with %3 channels",
 				stream->frame_rate(),
-				_content->resampled_frame_rate(film),
+				resampled_rate,
 				stream->channels()
 				);
 
-			resampler.reset (new Resampler (stream->frame_rate(), _content->resampled_frame_rate(film), stream->channels()));
+			resampler.reset (new Resampler(stream->frame_rate(), resampled_rate, stream->channels()));
 			if (_fast) {
 				resampler->set_fast ();
 			}