X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=20c8ea1ea812c60b7b33fd62a8d1300fe8c3958d;hb=0a93237cb5e4642d3b698ff9b7d0cfae5401478c;hp=17a534aa477489289308a8ff6e31706229ad2ca3;hpb=0da6975960aab345a1a8b30e6a95659cc936a1da;p=dcpomatic.git

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 17a534aa4..20c8ea1ea 100644
--- 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
@@ -19,132 +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 std::min;
-using boost::optional;
+using std::map;
 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 ()));
+	BOOST_FOREACH (AudioStreamPtr i, content->audio_streams ()) {
+		_streams[i] = shared_ptr<AudioDecoderStream> (new AudioDecoderStream (_audio_content, i, this));
 	}
-
-	reset_decoded_audio ();
-}
-
-void
-AudioDecoder::reset_decoded_audio ()
-{
-	_decoded_audio = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_audio_content->audio_channels(), 0)), 0);
 }
 
-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);
-	}
-
-	AudioFrame decoded_offset = 0;
-	
-	/* 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) {}
-		/* Use decoded_offset of 0, as we don't really care what frames we return */
-	} else {
-		while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
-		decoded_offset = frame - _decoded_audio.frame;
-	}
-
-	AudioFrame const amount_left = _decoded_audio.audio->frames() - decoded_offset;
-	
-	AudioFrame const to_return = min (amount_left, length);
-	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);
-
-	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.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
-
-	/* Resize _decoded_audio to fit the new data */
-	int const new_size = _audio_position.get() + data->frames() - _decoded_audio.frame;
-	_decoded_audio.audio->ensure_size (new_size);
-	_decoded_audio.audio->set_frames (new_size);
-
-	/* 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;
+	for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
+		i->second->flush ();
 	}
-
-	/*
-	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 ();
-	}
-	*/
 }
 
 void
-AudioDecoder::seek (ContentTime, bool)
+AudioDecoder::seek (ContentTime t, bool accurate)
 {
-	_audio_position.reset ();
-	reset_decoded_audio ();
+	for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
+		i->second->seek (t, accurate);
+	}
 }