Another try at sorting out the thorny question of timing.

1 files changed, 5 insertions, 96 deletions

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index a9e01908c..dc49a1846 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -31,57 +31,12 @@ 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 (_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 ()
@@ -113,54 +68,8 @@ AudioDecoder::process_end ()
 #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
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, AudioContent::Frame frame)
 {
-	shared_ptr<const Film> film = _film.lock ();
-	assert (film);
-	
-	return (_audio_content->length() - _next_audio) < film->audio_frames_to_time (1);
+	Audio (data, frame);
+	_audio_position = frame + data->frames ();
 }
-