Merge remote-tracking branch 'origin/master' into 1.0

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

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 2c0388fc39318851242b96a7a672b014ba5fca27..e4f98c678ad3a150987431f7f587428a07603b10 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -24,6 +24,7 @@
 #include "resampler.h"
 #include "util.h"
 #include "film.h"
+#include "audio_processor.h"
 
 #include "i18n.h"
 
@@ -43,13 +44,17 @@ AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
                _resampler.reset (new Resampler (content->audio_frame_rate(), content->resampled_audio_frame_rate(), content->audio_channels ()));
        }
 
+       if (content->audio_processor ()) {
+               _processor = content->audio_processor()->clone (content->resampled_audio_frame_rate ());
+       }
+
        reset_decoded_audio ();
 }
 
 void
 AudioDecoder::reset_decoded_audio ()
 {
-       _decoded_audio = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_audio_content->audio_channels(), 0)), 0);
+       _decoded_audio = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_audio_content->processed_audio_channels(), 0)), 0);
 }
 
 shared_ptr<ContentAudio>
@@ -125,8 +130,43 @@ AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
                data = _resampler->run (data);
        }
 
+       if (_processor) {
+               data = _processor->run (data);
+       }
+
+       AudioFrame const frame_rate = _audio_content->resampled_audio_frame_rate ();
+
+       if (_seek_reference) {
+               /* We've had an accurate seek and now we're seeing some data */
+               ContentTime const delta = time - _seek_reference.get ();
+               AudioFrame const delta_frames = delta.frames (frame_rate);
+               if (delta_frames > 0) {
+                       /* This data comes after the seek time.  Pad the data with some silence. */
+                       shared_ptr<AudioBuffers> padded (new AudioBuffers (data->channels(), data->frames() + delta_frames));
+                       padded->make_silent ();
+                       padded->copy_from (data.get(), data->frames(), 0, delta_frames);
+                       data = padded;
+                       time -= delta;
+               } else if (delta_frames < 0) {
+                       /* This data comes before the seek time.  Throw some data away */
+                       AudioFrame const to_discard = min (-delta_frames, static_cast<AudioFrame> (data->frames()));
+                       AudioFrame const to_keep = data->frames() - to_discard;
+                       if (to_keep == 0) {
+                               /* We have to throw all this data away, so keep _seek_reference and
+                                  try again next time some data arrives.
+                               */
+                               return;
+                       }
+                       shared_ptr<AudioBuffers> trimmed (new AudioBuffers (data->channels(), to_keep));
+                       trimmed->copy_from (data.get(), to_keep, to_discard, 0);
+                       data = trimmed;
+                       time += ContentTime::from_frames (to_discard, frame_rate);
+               }
+               _seek_reference = optional<ContentTime> ();
+       }
+
        if (!_audio_position) {
-               _audio_position = time.frames (_audio_content->resampled_audio_frame_rate ());
+               _audio_position = time.frames (frame_rate);
        }
 
        assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
@@ -168,8 +208,14 @@ AudioDecoder::flush ()
 }
 
 void
-AudioDecoder::seek (ContentTime, bool)
+AudioDecoder::seek (ContentTime t, bool accurate)
 {
        _audio_position.reset ();
        reset_decoded_audio ();
+       if (accurate) {
+               _seek_reference = t;
+       }
+       if (_processor) {
+               _processor->flush ();
+       }
 }