Fix font_id_map errors when importing DCP subtitles that have no

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

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 77c9b0695a3d7b6f4c7d3b57df84193e017863eb..ca1faa0100492b5e76363cde68b0aa4dbed1e9ab 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -32,8 +32,6 @@
 
 
 using std::cout;
-using std::map;
-using std::pair;
 using std::shared_ptr;
 using std::make_shared;
 using boost::optional;
@@ -60,33 +58,30 @@ AudioDecoder::emit (shared_ptr<const Film> film, AudioStreamPtr stream, shared_p
                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);
        }
 
-       auto 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 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 ());
-               }
-               reset = true;
-       } else if (std::abs(_positions[stream] - time.frames_round(resampled_rate)) > slack_frames) {
-               reset = true;
+       /* Amount of error we will tolerate on audio timestamps; see comment below.
+        * We'll use 1 24fps video frame as this seems to be roughly how ffplay does it.
+        */
+       Frame const slack_frames = resampled_rate / 24;
+
+       /* first_since_seek is set to true if this is the first data we have
+          received since initialisation or seek.  We'll set the position based
+          on the ContentTime that was given.  After this first time 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.
+       */
+
+       auto const first_since_seek = _positions[stream] == 0;
+       auto const need_reset = !first_since_seek && (std::abs(_positions[stream] - time.frames_round(resampled_rate)) > slack_frames);
+
+       if (need_reset) {
                LOG_GENERAL (
                        "Reset audio position: was %1, new data at %2, slack: %3 frames",
                        _positions[stream],
@@ -95,10 +90,14 @@ AudioDecoder::emit (shared_ptr<const Film> film, AudioStreamPtr stream, shared_p
                        );
        }
 
-       if (reset) {
+       if (first_since_seek || need_reset) {
                _positions[stream] = time.frames_round (resampled_rate);
        }
 
+       if (first_since_seek && _content->delay() > 0) {
+               silence (stream, _content->delay());
+       }
+
        shared_ptr<Resampler> resampler;
        auto i = _resamplers.find(stream);
        if (i != _resamplers.end()) {
@@ -185,18 +184,18 @@ AudioDecoder::flush ()
 
        if (_content->delay() < 0) {
                /* Finish off with the gap caused by the delay */
-               silence (-_content->delay ());
+               for (auto stream: _content->streams()) {
+                       silence (stream, -_content->delay());
+               }
        }
 }
 
 
 void
-AudioDecoder::silence (int milliseconds)
+AudioDecoder::silence (AudioStreamPtr stream, int milliseconds)
 {
-       for (auto i: _content->streams()) {
-               int const samples = ContentTime::from_seconds(milliseconds / 1000.0).frames_round(i->frame_rate());
-               auto silence = make_shared<AudioBuffers>(i->channels(), samples);
-               silence->make_silent ();
-               Data (i, ContentAudio (silence, _positions[i]));
-       }
+       int const samples = ContentTime::from_seconds(milliseconds / 1000.0).frames_round(stream->frame_rate());
+       auto silence = make_shared<AudioBuffers>(stream->channels(), samples);
+       silence->make_silent ();
+       Data (stream, ContentAudio(silence, _positions[stream]));
 }