X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=2944357bac7fe0f279bb8da163915c1e730f65e9;hb=aeb835a18c8df347e0ed68fb24631b320abeb611;hp=68554daf96856e37d07493538874bbbcd6e2f2d6;hpb=d683883c4dc25cb612f6d5feb1e772016182e722;p=dcpomatic.git

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 68554daf9..2944357ba 100644
--- 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) 2012-2015 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
@@ -18,114 +18,86 @@
 */
 
 #include "audio_decoder.h"
-#include "exceptions.h"
-#include "log.h"
+#include "audio_buffers.h"
+#include "audio_decoder_stream.h"
+#include "audio_content.h"
+#include <boost/foreach.hpp>
+#include <iostream>
 
 #include "i18n.h"
 
-using std::stringstream;
-using boost::optional;
+using std::cout;
+using std::map;
 using boost::shared_ptr;
 
-AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioContent> c)
-	: Decoder (f)
-	, _audio_content (c)
+AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content, bool fast)
+	: _audio_content (content)
+	, _ignore_audio (false)
+	, _fast (fast)
 {
-	if (_audio_content->audio_frame_rate() != _film->target_audio_sample_rate()) {
-
-		stringstream s;
-		s << String::compose ("Will resample audio from %1 to %2", _audio_content->audio_frame_rate(), _film->target_audio_sample_rate());
-		_film->log()->log (s.str ());
-
-		/* We will be using planar float data when we call the
-		   resampler.  As far as I can see, the audio channel
-		   layout is not necessary for our purposes; it seems
-		   only to be used get the number of channels and
-		   decide if rematrixing is needed.  It won't be, since
-		   input and output layouts are the same.
-		*/
-
-		_swr_context = swr_alloc_set_opts (
-			0,
-			av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
-			AV_SAMPLE_FMT_FLTP,
-			_film->target_audio_sample_rate(),
-			av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
-			AV_SAMPLE_FMT_FLTP,
-			_audio_content->audio_frame_rate(),
-			0, 0
-			);
-		
-		swr_init (_swr_context);
-	} else {
-		_swr_context = 0;
+	BOOST_FOREACH (AudioStreamPtr i, content->audio_streams ()) {
+		_streams[i] = shared_ptr<AudioDecoderStream> (new AudioDecoderStream (_audio_content, i, this));
 	}
 }
 
-AudioDecoder::~AudioDecoder ()
+ContentAudio
+AudioDecoder::get_audio (AudioStreamPtr stream, Frame frame, Frame length, bool accurate)
 {
-	if (_swr_context) {
-		swr_free (&_swr_context);
-	}
+	return _streams[stream]->get (frame, length, accurate);
 }
-	
 
-#if 0
 void
-AudioDecoder::process_end ()
+AudioDecoder::audio (AudioStreamPtr stream, shared_ptr<const AudioBuffers> data, ContentTime time)
 {
-	if (_film->has_audio() && _swr_context) {
+	if (_ignore_audio) {
+		return;
+	}
 
-		shared_ptr<AudioBuffers> out (new AudioBuffers (_film->audio_mapping().dcp_channels(), 256));
-			
-		while (1) {
-			int const frames = swr_convert (_swr_context, (uint8_t **) out->data(), 256, 0, 0);
+	if (_streams.find (stream) == _streams.end ()) {
 
-			if (frames < 0) {
-				throw EncodeError (_("could not run sample-rate converter"));
-			}
+		/* This method can be called with an unknown stream during the following sequence:
+		   - Add KDM to some DCP content.
+		   - Content gets re-examined.
+		   - SingleStreamAudioContent::take_from_audio_examiner creates a new stream.
+		   - Some content property change signal is delivered so Player::Changed is emitted.
+		   - Film viewer to re-gets the frame.
+		   - Player calls DCPDecoder pass which calls this method on the new stream.
 
-			if (frames == 0) {
-				break;
-			}
+		   At this point the AudioDecoder does not know about the new stream.
 
-			out->set_frames (frames);
-			_writer->write (out);
-		}
+		   Then
+		   - Some other property change signal is delivered which marks the player's pieces invalid.
+		   - Film viewer re-gets again.
+		   - Everything is OK.
 
+		   In this situation it is fine for us to silently drop the audio.
+		*/
+
+		return;
 	}
+
+	_streams[stream]->audio (data, time);
 }
-#endif
 
 void
-AudioDecoder::emit_audio (shared_ptr<const AudioBuffers> data, Time time)
+AudioDecoder::flush ()
 {
-	/* XXX: map audio to 5.1 */
-	
-	/* Maybe sample-rate convert */
-	if (_swr_context) {
-
-		/* Compute the resampled frames count and add 32 for luck */
-		int const max_resampled_frames = ceil ((int64_t) data->frames() * _film->target_audio_sample_rate() / _audio_content->audio_frame_rate()) + 32;
-
-		shared_ptr<AudioBuffers> resampled (new AudioBuffers (MAX_AUDIO_CHANNELS, max_resampled_frames));
-
-		/* Resample audio */
-		int const resampled_frames = swr_convert (
-			_swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) data->data(), data->frames()
-			);
-		
-		if (resampled_frames < 0) {
-			throw EncodeError (_("could not run sample-rate converter"));
-		}
-
-		resampled->set_frames (resampled_frames);
-		
-		/* And point our variables at the resampled audio */
-		data = resampled;
+	for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
+		i->second->flush ();
 	}
+}
 
-	Audio (data, time);
+void
+AudioDecoder::seek (ContentTime t, bool accurate)
+{
+	for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
+		i->second->seek (t, accurate);
+	}
 }
 
-		
+/** Set this player never to produce any audio data */
+void
+AudioDecoder::set_ignore_audio ()
+{
+	_ignore_audio = true;
+}