X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=8d3b0e1288ea5b53627a9cfb82619b0ccc125c7e;hb=e2be8234013335379bd49a53854218039348c7a4;hp=68554daf96856e37d07493538874bbbcd6e2f2d6;hpb=d683883c4dc25cb612f6d5feb1e772016182e722;p=dcpomatic.git

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 68554daf9..8d3b0e128 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -18,114 +18,72 @@
 */
 
 #include "audio_decoder.h"
+#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)
-	, _audio_content (c)
+AudioDecoder::AudioDecoder (shared_ptr<const Film> film, shared_ptr<const AudioContent> content)
+	: Decoder (film)
+	, _audio_content (content)
 {
-	if (_audio_content->audio_frame_rate() != _film->target_audio_sample_rate()) {
-
-		stringstream s;
-		s << String::compose ("Will resample audio from %1 to %2", _audio_content->audio_frame_rate(), _film->target_audio_sample_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,
-			_film->target_audio_sample_rate(),
-			av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
-			AV_SAMPLE_FMT_FLTP,
-			_audio_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 ()
+/** 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::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
 {
-	if (_swr_context) {
-		swr_free (&_swr_context);
+	if (_resampler) {
+		data = _resampler->run (data);
+	}
+
+	if (!_audio_position) {
+		shared_ptr<const Film> film = _film.lock ();
+		assert (film);
+		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 ();
 }
-	
 
-#if 0
 void
-AudioDecoder::process_end ()
+AudioDecoder::flush ()
 {
-	if (_film->has_audio() && _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);
-		}
+	if (!_resampler) {
+		return;
+	}
 
+	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 ();
 	}
 }
-#endif
 
 void
-AudioDecoder::emit_audio (shared_ptr<const AudioBuffers> data, Time time)
+AudioDecoder::seek (ContentTime, bool)
 {
-	/* 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() * _film->target_audio_sample_rate() / _audio_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);
+	_audio_position.reset ();
 }
-
-