Merge.

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

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index adabd5d9941afaad9d3b15d804d4a108876d0dab..9606c378c62d3babd09cf28c94b131c312aeaf39 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2012 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2012-2014 Carl Hetherington <cth@carlh.net>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of the GNU General Public License as published by
@@ -34,31 +34,84 @@ using std::cout;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoder::AudioDecoder (shared_ptr<const Film> film, shared_ptr<const AudioContent> content)
-       : Decoder (film)
-       , _audio_content (content)
-       , _audio_position (0)
+AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
+       : _audio_content (content)
+       , _decoded_audio (shared_ptr<AudioBuffers> (new AudioBuffers (content->audio_channels(), 0)), 0)
 {
        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.
+shared_ptr<ContentAudio>
+AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
+{
+       shared_ptr<ContentAudio> dec;
+
+       AudioFrame const end = frame + length - 1;
+               
+       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);
+       }
+
+       /* Now enough pass() calls will either:
+        *  (a) give us what we want, or
+        *  (b) hit the end of the decoder.
+        *
+        * If we are being accurate, we want the right frames,
+        * otherwise any frames will do.
+        */
+       if (accurate) {
+               while (!pass() && _decoded_audio.audio->frames() < length) {}
+       } else {
+               while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
+       }
+       
+       /* Clean up decoded */
+
+       AudioFrame const decoded_offset = frame - _decoded_audio.frame;
+       AudioFrame const amount_left = _decoded_audio.audio->frames() - decoded_offset;
+       _decoded_audio.audio->move (decoded_offset, 0, amount_left);
+       _decoded_audio.audio->set_frames (amount_left);
+
+       shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded_audio.audio->channels(), length));
+       out->copy_from (_decoded_audio.audio.get(), length, frame - _decoded_audio.frame, 0);
+
+       return shared_ptr<ContentAudio> (new ContentAudio (out, frame));
+}
+
+/** Called by subclasses when audio data is ready.
+ *
+ *  Audio timestamping is made hard by many factors, but perhaps the most entertaining 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)
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
 {
        if (_resampler) {
                data = _resampler->run (data);
        }
 
-       _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (data, _audio_position)));
-       _audio_position += data->frames ();
+       if (!_audio_position) {
+               _audio_position = time.frames (_audio_content->output_audio_frame_rate ());
+       }
+
+       assert (_audio_position >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
+
+       /* Resize _decoded_audio to fit the new data */
+       _decoded_audio.audio->ensure_size (_audio_position.get() + data->frames() - _decoded_audio.frame);
+
+       /* Copy new data in */
+       _decoded_audio.audio->copy_from (data.get(), data->frames(), 0, _audio_position.get() - _decoded_audio.frame);
+       _audio_position = _audio_position.get() + data->frames ();
 }
 
+/* XXX: called? */
 void
 AudioDecoder::flush ()
 {
@@ -66,19 +119,17 @@ AudioDecoder::flush ()
                return;
        }
 
+       /*
        shared_ptr<const AudioBuffers> b = _resampler->flush ();
        if (b) {
-               _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position)));
-               _audio_position += b->frames ();
+               _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position.get ())));
+               _audio_position = _audio_position.get() + b->frames ();
        }
+       */
 }
 
 void
-AudioDecoder::seek (ContentTime t, bool)
+AudioDecoder::seek (ContentTime, bool)
 {
-       shared_ptr<const Film> film = _film.lock ();
-       assert (film);
-       
-       FrameRateChange frc = film->active_frame_rate_change (_audio_content->position ());
-       _audio_position = ((t + first_audio()) / frc.speed_up) * film->audio_frame_rate() / TIME_HZ;
+       _audio_position.reset ();
 }