X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=59c631632f3f63b735a883dd1292c46403e4ec64;hb=49a51ff9778f3b72ee962d3e8bd9cf71944f3c2b;hp=6712e06241dc37a9e1c99fcaeb88c92c50f4d01d;hpb=64a2f1bb6a7b972607fa363631ef7a47444e8217;p=dcpomatic.git

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 6712e0624..59c631632 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -25,125 +25,22 @@
 #include "i18n.h"
 
 using std::stringstream;
+using std::list;
+using std::pair;
+using std::cout;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioContent> c)
+AudioDecoder::AudioDecoder (shared_ptr<const Film> f)
 	: Decoder (f)
-	, _next_audio (0)
-	, _audio_content (c)
+	, _audio_position (0)
 {
-	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(), _audio_content->output_audio_frame_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,
-			_audio_content->output_audio_frame_rate(),
-			av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
-			AV_SAMPLE_FMT_FLTP,
-			_audio_content->content_audio_frame_rate(),
-			0, 0
-			);
-		
-		swr_init (_swr_context);
-	} else {
-		_swr_context = 0;
-	}
-}
-
-AudioDecoder::~AudioDecoder ()
-{
-	if (_swr_context) {
-		swr_free (&_swr_context);
-	}
-}
-	
-
-#if 0
-void
-AudioDecoder::process_end ()
-{
-	if (_swr_context) {
-
-		shared_ptr<const Film> film = _film.lock ();
-		assert (film);
-		
-		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 (frames < 0) {
-				throw EncodeError (_("could not run sample-rate converter"));
-			}
-
-			if (frames == 0) {
-				break;
-			}
-
-			out->set_frames (frames);
-			_writer->write (out);
-		}
-
-	}
 }
-#endif
 
 void
-AudioDecoder::audio (shared_ptr<const AudioBuffers> data, Time time)
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, AudioContent::Frame frame)
 {
-	/* 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() * _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));
-
-		/* 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;
-	}
-
-	Audio (data, time);
-
-	shared_ptr<const Film> film = _film.lock ();
-	assert (film);
-	_next_audio = time + film->audio_frames_to_time (data->frames());
+	Audio (data, frame);
+	_audio_position = frame + data->frames ();
 }
-
-