std::shared_ptr

[dcpomatic.git] / src / lib / audio_decoder.cc

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index e0fb20b7eec172dbff61ff1e68bae2f8196d4310..5334dfa345071e887fa6e9f7a04d34701d23054b 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -33,7 +33,7 @@
 using std::cout;
 using std::map;
 using std::pair;
-using boost::shared_ptr;
+using std::shared_ptr;
 using boost::optional;
 using namespace dcpomatic;
 
@@ -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 ();
                        }
@@ -111,7 +137,7 @@ AudioDecoder::stream_position (shared_ptr<const Film> film, AudioStreamPtr strea
        return ContentTime::from_frames (i->second, _content->resampled_frame_rate(film));
 }
 
-ContentTime
+boost::optional<ContentTime>
 AudioDecoder::position (shared_ptr<const Film> film) const
 {
        optional<ContentTime> p;
@@ -122,7 +148,7 @@ AudioDecoder::position (shared_ptr<const Film> film) const
                }
        }
 
-       return p.get_value_or(ContentTime());
+       return p;
 }
 
 void