1 files changed, 102 insertions, 5 deletions
diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index a54c14843..e1c93ac77 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -18,19 +18,116 @@
 */
 
 #include "audio_decoder.h"
-#include "stream.h"
+#include "audio_buffers.h"
+#include "exceptions.h"
+#include "log.h"
 
+#include "i18n.h"
+
+using std::stringstream;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoder::AudioDecoder (shared_ptr<Film> f, DecodeOptions o)
-	: Decoder (f, o)
+AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioContent> c)
+	: Decoder (f)
+	, _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) {
+
+		stringstream s;
+		s << String::compose ("Will resample audio from %1 to %2", _audio_content->content_audio_frame_rate(), _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,
+			_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::set_audio_stream (shared_ptr<AudioStream> s)
+AudioDecoder::process_end ()
 {
-	_audio_stream = s;
+	if (_swr_context) {
+
+		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::emit_audio (shared_ptr<const AudioBuffers> data, Time time)
+{
+	/* 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() * _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);
+}
+
+