Merge master.

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

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index ade11cc3290ab0dcbb3d50628cad8689fe1aa034..8d3b0e1288ea5b53627a9cfb82619b0ccc125c7e 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -21,6 +21,9 @@
 #include "audio_buffers.h"
 #include "exceptions.h"
 #include "log.h"
+#include "resampler.h"
+#include "util.h"
+#include "film.h"
 
 #include "i18n.h"
 
@@ -31,15 +34,56 @@ using std::cout;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoder::AudioDecoder (shared_ptr<const Film> f)
-       : Decoder (f)
-       , _audio_position (0)
+AudioDecoder::AudioDecoder (shared_ptr<const Film> film, shared_ptr<const AudioContent> content)
+       : Decoder (film)
+       , _audio_content (content)
 {
+       if (content->output_audio_frame_rate() != content->content_audio_frame_rate() && content->audio_channels ()) {
+               _resampler.reset (new Resampler (content->content_audio_frame_rate(), content->output_audio_frame_rate(), content->audio_channels ()));
+       }
 }
 
+/** Audio timestamping is made hard by many factors, but the final nail in the coffin is resampling.
+ *  We have to assume that we are feeding continuous data into the resampler, and so we get continuous
+ *  data out.  Hence we do the timestamping here, post-resampler, just by counting samples.
+ *
+ *  The time is passed in here so that after a seek we can set up our _audio_position.  The
+ *  time is ignored once this has been done.
+ */
 void
-AudioDecoder::audio (shared_ptr<const AudioBuffers> data, AudioContent::Frame frame)
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
 {
-       Audio (data, frame);
-       _audio_position = frame + data->frames ();
+       if (_resampler) {
+               data = _resampler->run (data);
+       }
+
+       if (!_audio_position) {
+               shared_ptr<const Film> film = _film.lock ();
+               assert (film);
+               FrameRateChange frc = film->active_frame_rate_change (_audio_content->position ());
+               _audio_position = (double (time) / frc.speed_up) * film->audio_frame_rate() / TIME_HZ;
+       }
+
+       _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (data, _audio_position.get ())));
+       _audio_position = _audio_position.get() + data->frames ();
+}
+
+void
+AudioDecoder::flush ()
+{
+       if (!_resampler) {
+               return;
+       }
+
+       shared_ptr<const AudioBuffers> b = _resampler->flush ();
+       if (b) {
+               _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position.get ())));
+               _audio_position = _audio_position.get() + b->frames ();
+       }
+}
+
+void
+AudioDecoder::seek (ContentTime, bool)
+{
+       _audio_position.reset ();
 }