X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=a9e01908c4782a6095821d080800b0568ce71214;hb=50cb31af16240b248700dab1484d7f07656c66df;hp=df13a984a4b36416fb1120b6512aca4537484f6c;hpb=28dbf4fd074d2046a3c8ddebac9a537a80fd457a;p=dcpomatic.git

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index df13a984a..a9e01908c 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -18,12 +18,149 @@
 */
 
 #include "audio_decoder.h"
+#include "audio_buffers.h"
+#include "exceptions.h"
+#include "log.h"
 
+#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)
+AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioContent> c)
 	: Decoder (f)
+	, _next_audio (0)
+	, _audio_content (c)
+{
+	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 (_audio_content->audio_channels ()),
+			AV_SAMPLE_FMT_FLTP,
+			_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
+			);
+		
+		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)
+{
+	/* 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 (data->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;
+	}
+
+	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);
 }
+