Merge master.

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

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 3964719104956a46e16f734fd8acaadcaa7b9df0..32453cc13c974d58218cf5dae9eb4eb6da5047d2 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -21,6 +21,8 @@
 #include "audio_buffers.h"
 #include "exceptions.h"
 #include "log.h"
+#include "resampler.h"
+#include "util.h"
 
 #include "i18n.h"
 
@@ -31,45 +33,52 @@ using std::cout;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoder::AudioDecoder (shared_ptr<const Film> f)
-       : Decoder (f)
-       , _next_audio_frame (0)
+AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
+       : _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 ()));
+       }
 }
 
-#if 0
+/** 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::process_end ()
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
 {
-       if (_swr_context) {
-
-               shared_ptr<const Film> film = _film.lock ();
-               assert (film);
-               
-               shared_ptr<AudioBuffers> out (new AudioBuffers (film->audio_mapping().dcp_channels(), 256));
-                       
-               while (1) {
-                       int const frames = swr_convert (_swr_context, (uint8_t **) out->data(), 256, 0, 0);
+       if (_resampler) {
+               data = _resampler->run (data);
+       }
 
-                       if (frames < 0) {
-                               throw EncodeError (_("could not run sample-rate converter"));
-                       }
+       if (!_audio_position) {
+               _audio_position = time;
+       }
 
-                       if (frames == 0) {
-                               break;
-                       }
+       _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (_audio_position.get (), data)));
+       _audio_position = _audio_position.get() + ContentTime (data->frames (), _audio_content->output_audio_frame_rate ());
+}
 
-                       out->set_frames (frames);
-                       _writer->write (out);
-               }
+void
+AudioDecoder::flush ()
+{
+       if (!_resampler) {
+               return;
+       }
 
+       shared_ptr<const AudioBuffers> b = _resampler->flush ();
+       if (b) {
+               _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (_audio_position.get (), b)));
+               _audio_position = _audio_position.get() + ContentTime (b->frames (), _audio_content->output_audio_frame_rate ());
        }
 }
-#endif
 
 void
-AudioDecoder::audio (shared_ptr<const AudioBuffers> data, AudioContent::Frame frame)
+AudioDecoder::seek (ContentTime, bool)
 {
-       Audio (data, frame);
-       _next_audio_frame = frame + data->frames ();
+       _audio_position.reset ();
 }