X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;ds=sidebyside;f=src%2Flib%2Fencoder.cc;h=c1d1041ae539f9cdab1f087291eb2d8d7104abbb;hb=dbc43b6e3021e34875d7d5bba04abf7ad1fc8633;hp=46d11c55640e2fbf92a0789748f9f9a284c1d0b8;hpb=263eee639546964aaa57f5d2d3b24008ecfe8adb;p=dcpomatic.git

diff --git a/src/lib/encoder.cc b/src/lib/encoder.cc
index 46d11c556..c1d1041ae 100644
--- a/src/lib/encoder.cc
+++ b/src/lib/encoder.cc
@@ -86,13 +86,20 @@ Encoder::process_begin ()
 		s << String::compose (N_("Will resample audio from %1 to %2"), _film->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 */
+		/* 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,
-			_film->audio_channel_layout(),
+			av_get_default_channel_layout (_film->audio_mapping().dcp_channels ()),
 			AV_SAMPLE_FMT_FLTP,
 			_film->target_audio_sample_rate(),
-			_film->audio_channel_layout(),
+			av_get_default_channel_layout (_film->audio_mapping().dcp_channels ()),
 			AV_SAMPLE_FMT_FLTP,
 			_film->audio_frame_rate(),
 			0, 0
@@ -128,9 +135,9 @@ void
 Encoder::process_end ()
 {
 #if HAVE_SWRESAMPLE	
-	if (_film->has_audio() && _film->audio_channels() && _swr_context) {
+	if (_film->has_audio() && _swr_context) {
 
-		shared_ptr<AudioBuffers> out (new AudioBuffers (_film->audio_channels(), 256));
+		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);
@@ -233,7 +240,7 @@ Encoder::frame_done ()
 }
 
 void
-Encoder::process_video (shared_ptr<Image> image, bool same, shared_ptr<Subtitle> sub)
+Encoder::process_video (shared_ptr<const Image> image, bool same, shared_ptr<Subtitle> sub)
 {
 	FrameRateConversion frc (_film->video_frame_rate(), _film->dcp_frame_rate());
 	
@@ -296,7 +303,7 @@ Encoder::process_video (shared_ptr<Image> image, bool same, shared_ptr<Subtitle>
 }
 
 void
-Encoder::process_audio (shared_ptr<AudioBuffers> data)
+Encoder::process_audio (shared_ptr<const AudioBuffers> data)
 {
 #if HAVE_SWRESAMPLE
 	/* Maybe sample-rate convert */
@@ -305,7 +312,7 @@ Encoder::process_audio (shared_ptr<AudioBuffers> data)
 		/* 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() / _film->audio_frame_rate()) + 32;
 
-		shared_ptr<AudioBuffers> resampled (new AudioBuffers (_film->audio_channels(), max_resampled_frames));
+		shared_ptr<AudioBuffers> resampled (new AudioBuffers (_film->audio_mapping().dcp_channels(), max_resampled_frames));
 
 		/* Resample audio */
 		int const resampled_frames = swr_convert (
@@ -335,7 +342,9 @@ Encoder::terminate_threads ()
 	lock.unlock ();
 
 	for (list<boost::thread *>::iterator i = _threads.begin(); i != _threads.end(); ++i) {
-		(*i)->join ();
+		if ((*i)->joinable ()) {
+			(*i)->join ();
+		}
 		delete *i;
 	}
 }
@@ -416,7 +425,7 @@ Encoder::encoder_thread (ServerDescription* server)
 		}
 
 		if (remote_backoff > 0) {
-			dvdomatic_sleep (remote_backoff);
+			dcpomatic_sleep (remote_backoff);
 		}
 
 		lock.lock ();