X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=c0ef02f65d5ab5518dcb7e53aa145b1ff3f9a598;hb=cadf2d574d144098fffa3c61e0a2be88f496cac6;hp=e1c93ac77bcb23de9a5888cdc2babd36ffdef442;hpb=237a0052c60af768f4d62b00321932918b7ba4d9;p=dcpomatic.git

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index e1c93ac77..c0ef02f65 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -21,113 +21,38 @@
 #include "audio_buffers.h"
 #include "exceptions.h"
 #include "log.h"
+#include "resampler.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)
-	, _output_audio_frame_rate (_audio_content->output_audio_frame_rate (f))
+AudioDecoder::AudioDecoder (shared_ptr<const Film> film, shared_ptr<const AudioContent> content)
+	: Decoder (film)
+	, _audio_content (content)
+	, _audio_position (0)
 {
-	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::process_end ()
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, AudioContent::Frame frame)
 {
-	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);
-		}
-
-	}
+	Audio (data, frame);
+	_audio_position = frame + data->frames ();
 }
-#endif
 
-void
-AudioDecoder::emit_audio (shared_ptr<const AudioBuffers> data, Time time)
+/** This is a bit odd, but necessary when we have (e.g.) FFmpegDecoders with no audio.
+ *  The player needs to know that there is no audio otherwise it will keep trying to
+ *  pass() the decoder to get it to emit audio.
+ */
+bool
+AudioDecoder::has_audio () const
 {
-	/* 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);
+	return _audio_content->audio_channels () > 0;
 }
-
-