X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=a9e01908c4782a6095821d080800b0568ce71214;hb=50cb31af16240b248700dab1484d7f07656c66df;hp=ddda816a571d374897ee499f8bfdee155b1b8ee2;hpb=8c6fe8e1e8c8f6d5932606f2a5b6e1b87681ae38;p=dcpomatic.git

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index ddda816a5..a9e01908c 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -25,6 +25,9 @@
 #include "i18n.h"
 
 using std::stringstream;
+using std::list;
+using std::pair;
+using std::cout;
 using boost::optional;
 using boost::shared_ptr;
 
@@ -32,15 +35,18 @@ AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioCont
 	: Decoder (f)
 	, _next_audio (0)
 	, _audio_content (c)
-	, _output_audio_frame_rate (_audio_content->output_audio_frame_rate (f))
 {
-	if (_audio_content->content_audio_frame_rate() != _output_audio_frame_rate) {
+	if (_audio_content->content_audio_frame_rate() != _audio_content->output_audio_frame_rate()) {
 
 		shared_ptr<const Film> film = _film.lock ();
 		assert (film);
 
 		stringstream s;
-		s << String::compose ("Will resample audio from %1 to %2", _audio_content->content_audio_frame_rate(), _output_audio_frame_rate);
+		s << String::compose (
+			"Will resample audio from %1 to %2",
+			_audio_content->content_audio_frame_rate(), _audio_content->output_audio_frame_rate()
+			);
+		
 		film->log()->log (s.str ());
 
 		/* We will be using planar float data when we call the
@@ -53,10 +59,10 @@ AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioCont
 
 		_swr_context = swr_alloc_set_opts (
 			0,
-			av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
+			av_get_default_channel_layout (_audio_content->audio_channels ()),
 			AV_SAMPLE_FMT_FLTP,
-			_output_audio_frame_rate,
-			av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
+			_audio_content->output_audio_frame_rate(),
+			av_get_default_channel_layout (_audio_content->audio_channels ()),
 			AV_SAMPLE_FMT_FLTP,
 			_audio_content->content_audio_frame_rate(),
 			0, 0
@@ -109,15 +115,15 @@ AudioDecoder::process_end ()
 void
 AudioDecoder::audio (shared_ptr<const AudioBuffers> data, Time time)
 {
-	/* XXX: map audio to 5.1 */
-	
 	/* Maybe resample */
 	if (_swr_context) {
 
 		/* Compute the resampled frames count and add 32 for luck */
-		int const max_resampled_frames = ceil ((int64_t) data->frames() * _output_audio_frame_rate / _audio_content->content_audio_frame_rate()) + 32;
+		int const max_resampled_frames = ceil (
+			(int64_t) data->frames() * _audio_content->output_audio_frame_rate() / _audio_content->content_audio_frame_rate()
+			) + 32;
 
-		shared_ptr<AudioBuffers> resampled (new AudioBuffers (MAX_AUDIO_CHANNELS, max_resampled_frames));
+		shared_ptr<AudioBuffers> resampled (new AudioBuffers (data->channels(), max_resampled_frames));
 
 		/* Resample audio */
 		int const resampled_frames = swr_convert (
@@ -134,11 +140,27 @@ AudioDecoder::audio (shared_ptr<const AudioBuffers> data, Time time)
 		data = resampled;
 	}
 
-	Audio (data, time);
-
 	shared_ptr<const Film> film = _film.lock ();
 	assert (film);
+	
+	/* Remap channels */
+	shared_ptr<AudioBuffers> dcp_mapped (new AudioBuffers (film->dcp_audio_channels(), data->frames()));
+	dcp_mapped->make_silent ();
+	list<pair<int, libdcp::Channel> > map = _audio_content->audio_mapping().content_to_dcp ();
+	for (list<pair<int, libdcp::Channel> >::iterator i = map.begin(); i != map.end(); ++i) {
+		dcp_mapped->accumulate_channel (data.get(), i->first, i->second);
+	}
+
+	Audio (dcp_mapped, time);
 	_next_audio = time + film->audio_frames_to_time (data->frames());
 }
 
-		
+bool
+AudioDecoder::audio_done () const
+{
+	shared_ptr<const Film> film = _film.lock ();
+	assert (film);
+	
+	return (_audio_content->length() - _next_audio) < film->audio_frames_to_time (1);
+}
+