Handle multiple audio streams in a single piece of content

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

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 9606c378c62d3babd09cf28c94b131c312aeaf39..20c8ea1ea812c60b7b33fd62a8d1300fe8c3958d 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2012-2014 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2012-2015 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
@@ -19,117 +19,48 @@
 
 #include "audio_decoder.h"
 #include "audio_buffers.h"
-#include "exceptions.h"
-#include "log.h"
-#include "resampler.h"
-#include "util.h"
-#include "film.h"
+#include "audio_decoder_stream.h"
+#include <boost/foreach.hpp>
+#include <iostream>
 
 #include "i18n.h"
 
-using std::stringstream;
-using std::list;
-using std::pair;
 using std::cout;
-using boost::optional;
+using std::map;
 using boost::shared_ptr;
 
 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 ()));
+       BOOST_FOREACH (AudioStreamPtr i, content->audio_streams ()) {
+               _streams[i] = shared_ptr<AudioDecoderStream> (new AudioDecoderStream (_audio_content, i, this));
        }
 }
 
-shared_ptr<ContentAudio>
-AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
+ContentAudio
+AudioDecoder::get_audio (AudioStreamPtr stream, Frame frame, Frame 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));
+       return _streams[stream]->get (frame, length, accurate);
 }
 
-/** 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, ContentTime time)
+AudioDecoder::audio (AudioStreamPtr stream, shared_ptr<const AudioBuffers> data, ContentTime time)
 {
-       if (_resampler) {
-               data = _resampler->run (data);
-       }
-
-       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 ();
+       _streams[stream]->audio (data, time);
 }
 
-/* XXX: called? */
 void
 AudioDecoder::flush ()
 {
-       if (!_resampler) {
-               return;
-       }
-
-       /*
-       shared_ptr<const AudioBuffers> b = _resampler->flush ();
-       if (b) {
-               _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position.get ())));
-               _audio_position = _audio_position.get() + b->frames ();
+       for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
+               i->second->flush ();
        }
-       */
 }
 
 void
-AudioDecoder::seek (ContentTime, bool)
+AudioDecoder::seek (ContentTime t, bool accurate)
 {
-       _audio_position.reset ();
+       for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
+               i->second->seek (t, accurate);
+       }
 }