Merge master.

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

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index c0c92476a23669f1fa3b2383bc3d2c91b79071a4..4a543cea9351af00cf652b6bcfd755ad013d2860 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -39,8 +39,8 @@ 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 ();
@@ -61,7 +61,7 @@ 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
@@ -86,21 +86,23 @@ AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
        }
 
        /* 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
+          if pass() returned true before we got enough data.
        */
-       AudioFrame const amount_left = _decoded_audio.audio->frames() - decoded_offset;
-       
+       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 (amount_left, length));
+       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);
-       
+
+       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, amount_left - to_return);
+       _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 (amount_left - to_return);
+       _decoded_audio.audio->set_frames (remaining);
        /* Also bump where those frames are in terms of the content */
        _decoded_audio.frame += decoded_offset + to_return;
 
@@ -124,7 +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()));