Call processor process_end methods as required. Remove questionable padding of audio length up to the nearest second. Don't emit audio in big blocks as it seems to crash FFmpeg. Fix a few things.

4 files changed, 31 insertions, 16 deletions

diff --git a/src/lib/external_audio_decoder.cc b/src/lib/external_audio_decoder.cc
index 29668d922..136e00fb2 100644
--- a/src/lib/external_audio_decoder.cc
+++ b/src/lib/external_audio_decoder.cc
@@ -106,9 +106,10 @@ ExternalAudioDecoder::pass ()
 		return true;
 	}
 
-	sf_count_t const block = 65536;
-
-	cout << frames << " audio frames.\n";
+	/* Do things in half second blocks as I think there may be limits
+	   to what FFmpeg (and in particular the resampler) can cope with.
+	*/
+	sf_count_t const block = _audio_stream->sample_rate() / 2;
 
 	shared_ptr<AudioBuffers> audio (new AudioBuffers (_audio_stream->channels(), block));
 	while (frames > 0) {
@@ -121,6 +122,7 @@ ExternalAudioDecoder::pass ()
 			}
 		}
 
+		audio->set_frames (this_time);
 		Audio (audio);
 		frames -= this_time;
 	}
diff --git a/src/lib/j2k_wav_encoder.cc b/src/lib/j2k_wav_encoder.cc
index be1f96fd4..134d74623 100644
--- a/src/lib/j2k_wav_encoder.cc
+++ b/src/lib/j2k_wav_encoder.cc
@@ -337,12 +337,6 @@ J2KWAVEncoder::process_end ()
 #endif
 
 	if (_film->audio_stream()) {
-		int const dcp_sr = dcp_audio_sample_rate (_film->audio_stream()->sample_rate ());
-		int64_t const extra_audio_frames = dcp_sr - (_audio_frames_written % dcp_sr);
-		shared_ptr<AudioBuffers> silence (new AudioBuffers (_film->audio_stream()->channels(), extra_audio_frames));
-		silence->make_silent ();
-		write_audio (silence);
-		
 		close_sound_files ();
 		
 		/* Rename .wav.tmp files to .wav */
diff --git a/src/lib/matcher.cc b/src/lib/matcher.cc
index 24514bee2..7b4434539 100644
--- a/src/lib/matcher.cc
+++ b/src/lib/matcher.cc
@@ -21,6 +21,7 @@
 #include "image.h"
 #include "log.h"
 
+using std::min;
 using boost::shared_ptr;
 
 Matcher::Matcher (Log* log, int sample_rate, float frames_per_second)
@@ -92,9 +93,21 @@ Matcher::process_end ()
 	
 	if (audio_short_by_frames > 0) {
 		_log->log (String::compose ("Emitted %1 too few audio frames", audio_short_by_frames));
-		shared_ptr<AudioBuffers> b (new AudioBuffers (_channels.get(), audio_short_by_frames));
+
+		/* Do things in half second blocks as I think there may be limits
+		   to what FFmpeg (and in particular the resampler) can cope with.
+		*/
+		int64_t const block = _sample_rate / 2;
+		shared_ptr<AudioBuffers> b (new AudioBuffers (_channels.get(), block));
 		b->make_silent ();
-		Audio (b);
-		_audio_frames += b->frames ();
+		
+		int64_t to_do = audio_short_by_frames;
+		while (to_do > 0) {
+			int64_t const this_time = min (to_do, block);
+			b->set_frames (this_time);
+			Audio (b);
+			_audio_frames += b->frames ();
+			to_do -= this_time;
+		}
 	}
 }
diff --git a/src/lib/transcoder.cc b/src/lib/transcoder.cc
index cad76af6e..537b9b664 100644
--- a/src/lib/transcoder.cc
+++ b/src/lib/transcoder.cc
@@ -109,12 +109,18 @@ Transcoder::go ()
 		}
 		
 	} catch (...) {
-		/* process_end() is important as the decoder may have worker
-		   threads that need to be cleaned up.
-		*/
 		_encoder->process_end ();
 		throw;
 	}
-
+	
+	if (_delay_line) {
+		_delay_line->process_end ();
+	}
+	if (_matcher) {
+		_matcher->process_end ();
+	}
+	if (_gain) {
+		_gain->process_end ();
+	}
 	_encoder->process_end ();
 }