Merge master.

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

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 17a534aa477489289308a8ff6e31706229ad2ca3..4a543cea9351af00cf652b6bcfd755ad013d2860 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -32,14 +32,15 @@ using std::list;
 using std::pair;
 using std::cout;
 using std::min;
+using std::max;
 using boost::optional;
 using boost::shared_ptr;
 
 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 (content->resampled_audio_frame_rate() != content->audio_frame_rate() && content->audio_channels ()) {
+               _resampler.reset (new Resampler (content->audio_frame_rate(), content->resampled_audio_frame_rate(), content->audio_channels ()));
        }
 
        reset_decoded_audio ();
@@ -60,9 +61,12 @@ AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
                
        if (frame < _decoded_audio.frame || end > (_decoded_audio.frame + length * 4)) {
                /* Either we have no decoded data, or what we do have is a long way from what we want: seek */
-               seek (ContentTime::from_frames (frame, _audio_content->content_audio_frame_rate()), accurate);
+               seek (ContentTime::from_frames (frame, _audio_content->audio_frame_rate()), accurate);
        }
 
+       /* Offset of the data that we want from the start of _decoded_audio.audio
+          (to be set up shortly)
+       */
        AudioFrame decoded_offset = 0;
        
        /* Now enough pass() calls will either:
@@ -73,22 +77,34 @@ AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
         * otherwise any frames will do.
         */
        if (accurate) {
-               while (!pass() && _decoded_audio.audio->frames() < length) {}
-               /* Use decoded_offset of 0, as we don't really care what frames we return */
-       } else {
+               /* Keep stuffing data into _decoded_audio until we have enough data, or the subclass does not want to give us any more */
                while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
                decoded_offset = frame - _decoded_audio.frame;
+       } else {
+               while (!pass() && _decoded_audio.audio->frames() < length) {}
+               /* Use decoded_offset of 0, as we don't really care what frames we return */
        }
 
-       AudioFrame const amount_left = _decoded_audio.audio->frames() - decoded_offset;
-       
-       AudioFrame const to_return = min (amount_left, length);
+       /* The amount of data available in _decoded_audio.audio starting from `frame'.  This could be -ve
+          if pass() returned true before we got enough data.
+       */
+       AudioFrame const available = _decoded_audio.audio->frames() - decoded_offset;
+
+       /* We will return either that, or the requested amount, whichever is smaller */
+       AudioFrame const to_return = max ((AudioFrame) 0, min (available, length));
+
+       /* Copy our data to the output */
        shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded_audio.audio->channels(), to_return));
        out->copy_from (_decoded_audio.audio.get(), to_return, decoded_offset, 0);
-       
-       /* Clean up decoded */
-       _decoded_audio.audio->move (decoded_offset + to_return, 0, amount_left - to_return);
-       _decoded_audio.audio->set_frames (amount_left - to_return);
+
+       AudioFrame const remaining = max ((AudioFrame) 0, available - to_return);
+
+       /* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
+       _decoded_audio.audio->move (decoded_offset + to_return, 0, remaining);
+       /* And set up the number of frames we have left */
+       _decoded_audio.audio->set_frames (remaining);
+       /* Also bump where those frames are in terms of the content */
+       _decoded_audio.frame += decoded_offset + to_return;
 
        return shared_ptr<ContentAudio> (new ContentAudio (out, frame));
 }
@@ -110,7 +126,7 @@ AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
        }
 
        if (!_audio_position) {
-               _audio_position = time.frames (_audio_content->output_audio_frame_rate ());
+               _audio_position = time.frames (_audio_content->resampled_audio_frame_rate ());
        }
 
        assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));