X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;ds=sidebyside;f=src%2Flib%2Faudio_decoder.cc;h=8d3b0e1288ea5b53627a9cfb82619b0ccc125c7e;hb=4e5d5c7dcc6470b8dc918d03a00e30c07df60efe;hp=8950e15465d3602673ff75bcae43f5f5e129a49c;hpb=4f03da3aa12525cb8389ddefee629f5d0b2ac0aa;p=dcpomatic.git

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 8950e1546..8d3b0e128 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -21,135 +21,69 @@
 #include "audio_buffers.h"
 #include "exceptions.h"
 #include "log.h"
+#include "resampler.h"
+#include "util.h"
+#include "film.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, shared_ptr<const AudioContent> c)
-	: Decoder (f)
-	, _next_audio (0)
-	, _audio_content (c)
+AudioDecoder::AudioDecoder (shared_ptr<const Film> film, shared_ptr<const AudioContent> content)
+	: Decoder (film)
+	, _audio_content (content)
 {
-	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;
+	if (content->output_audio_frame_rate() != content->content_audio_frame_rate() && content->audio_channels ()) {
+		_resampler.reset (new Resampler (content->content_audio_frame_rate(), content->output_audio_frame_rate(), content->audio_channels ()));
 	}
 }
 
-AudioDecoder::~AudioDecoder ()
-{
-	if (_swr_context) {
-		swr_free (&_swr_context);
-	}
-}
-	
-
-#if 0
+/** Audio timestamping is made hard by many factors, but the final nail in the coffin is resampling.
+ *  We have to assume that we are feeding continuous data into the resampler, and so we get continuous
+ *  data out.  Hence we do the timestamping here, post-resampler, just by counting samples.
+ *
+ *  The time is passed in here so that after a seek we can set up our _audio_position.  The
+ *  time is ignored once this has been done.
+ */
 void
-AudioDecoder::process_end ()
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
 {
-	if (_swr_context) {
+	if (_resampler) {
+		data = _resampler->run (data);
+	}
 
+	if (!_audio_position) {
 		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);
-		}
-
+		FrameRateChange frc = film->active_frame_rate_change (_audio_content->position ());
+		_audio_position = (double (time) / frc.speed_up) * film->audio_frame_rate() / TIME_HZ;
 	}
+
+	_pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (data, _audio_position.get ())));
+	_audio_position = _audio_position.get() + data->frames ();
 }
-#endif
 
 void
-AudioDecoder::audio (shared_ptr<const AudioBuffers> data, Time time)
+AudioDecoder::flush ()
 {
-	/* 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;
+	if (!_resampler) {
+		return;
 	}
 
-	shared_ptr<const Film> film = _film.lock ();
-	assert (film);
-	
-	/* Remap channels */
-	shared_ptr<AudioBuffers> dcp_mapped (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 (data, i->first, i->second);
+	shared_ptr<const AudioBuffers> b = _resampler->flush ();
+	if (b) {
+		_pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position.get ())));
+		_audio_position = _audio_position.get() + b->frames ();
 	}
-
-	Audio (dcp_mapped, time);
-	_next_audio = time + film->audio_frames_to_time (data->frames());
 }
 
-		
+void
+AudioDecoder::seek (ContentTime, bool)
+{
+	_audio_position.reset ();
+}