X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Fj2k_wav_encoder.cc;h=c0c785d118d8254be4eb3489bca2806ca8f9a230;hb=579dfce1bae018efd73c9f5f3f31770cb401d19a;hp=58f8a101f7440c0be6312846114c9ebeed474011;hpb=476af651fc34d7c6394fea0e39d351392b4e46d3;p=dcpomatic.git

diff --git a/src/lib/j2k_wav_encoder.cc b/src/lib/j2k_wav_encoder.cc
index 58f8a101f..c0c785d11 100644
--- a/src/lib/j2k_wav_encoder.cc
+++ b/src/lib/j2k_wav_encoder.cc
@@ -49,6 +49,7 @@ J2KWAVEncoder::J2KWAVEncoder (shared_ptr<const FilmState> s, shared_ptr<const Op
 #ifdef HAVE_SWRESAMPLE	  
 	, _swr_context (0)
 #endif	  
+	, _audio_frames_written (0)
 	, _process_end (false)
 {
 	/* Create sound output files with .tmp suffixes; we will rename
@@ -216,13 +217,13 @@ J2KWAVEncoder::encoder_thread (ServerDescription* server)
 }
 
 void
-J2KWAVEncoder::process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format)
+J2KWAVEncoder::process_begin (int64_t audio_channel_layout)
 {
-	if (_fs->audio_sample_rate() != _fs->target_sample_rate()) {
+	if (_fs->audio_sample_rate() != _fs->target_audio_sample_rate()) {
 #ifdef HAVE_SWRESAMPLE
 
 		stringstream s;
-		s << "Will resample audio from " << _fs->audio_sample_rate() << " to " << _fs->target_sample_rate();
+		s << "Will resample audio from " << _fs->audio_sample_rate() << " to " << _fs->target_audio_sample_rate();
 		_log->log (s.str ());
 
 		/* We will be using planar float data when we call the resampler */
@@ -230,7 +231,7 @@ J2KWAVEncoder::process_begin (int64_t audio_channel_layout, AVSampleFormat audio
 			0,
 			audio_channel_layout,
 			AV_SAMPLE_FMT_FLTP,
-			_fs->target_sample_rate(),
+			_fs->target_audio_sample_rate(),
 			audio_channel_layout,
 			AV_SAMPLE_FMT_FLTP,
 			_fs->audio_sample_rate(),
@@ -303,13 +304,10 @@ J2KWAVEncoder::process_end ()
 #if HAVE_SWRESAMPLE	
 	if (_swr_context) {
 
-		float* out[_fs->audio_channels()];
-		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			out[i] = new float[256];
-		}
+		shared_ptr<AudioBuffers> out (new AudioBuffers (_fs->audio_channels(), 256));
 			
 		while (1) {
-			int const frames = swr_convert (_swr_context, (uint8_t **) out, 256, 0, 0);
+			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");
@@ -319,16 +317,18 @@ J2KWAVEncoder::process_end ()
 				break;
 			}
 
-			write_audio (out, frames);
-		}
-
-		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			delete[] out[i];
+			write_audio (out);
 		}
 
 		swr_free (&_swr_context);
 	}
-#endif	
+#endif
+
+	int const dcp_sr = dcp_audio_sample_rate (_fs->audio_sample_rate ());
+	int64_t const extra_audio_frames = dcp_sr - (_audio_frames_written % dcp_sr);
+	shared_ptr<AudioBuffers> silence (new AudioBuffers (_fs->audio_channels(), extra_audio_frames));
+	silence->make_silent ();
+	write_audio (silence);
 	
 	close_sound_files ();
 
@@ -342,50 +342,45 @@ J2KWAVEncoder::process_end ()
 }
 
 void
-J2KWAVEncoder::process_audio (float** data, int frames)
+J2KWAVEncoder::process_audio (shared_ptr<const AudioBuffers> audio)
 {
-	float* resampled[_fs->audio_channels()];
+	shared_ptr<AudioBuffers> resampled;
 	
-#if HAVE_SWRESAMPLE	
+#if HAVE_SWRESAMPLE
 	/* Maybe sample-rate convert */
 	if (_swr_context) {
 
 		/* Compute the resampled frames count and add 32 for luck */
-		int const resampled_frames = ceil (frames * _fs->target_sample_rate() / _fs->audio_sample_rate()) + 32;
+		int const max_resampled_frames = ceil (audio->frames() * _fs->target_audio_sample_rate() / _fs->audio_sample_rate()) + 32;
 
-		/* Make a buffer to put the result in */
-		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			resampled[i] = new float[resampled_frames];
-		}
+		resampled.reset (new AudioBuffers (_fs->audio_channels(), max_resampled_frames));
 
 		/* Resample audio */
-		int out_frames = swr_convert (_swr_context, (uint8_t **) resampled, resampled_frames, (uint8_t const **) data, frames);
-		if (out_frames < 0) {
+		int const resampled_frames = swr_convert (
+			_swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) audio->data(), audio->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;
-		frames = resampled_frames;
+		audio = resampled;
 	}
 #endif
 
-	write_audio (data, frames);
-
-#if HAVE_SWRESAMPLE
-	if (_swr_context) {
-		for (int i = 0; i < _fs->audio_channels(); ++i) {
-			delete[] resampled[i];
-		}
-	}
-#endif	
+	write_audio (audio);
 }
 
 void
-J2KWAVEncoder::write_audio (float** data, int frames)
+J2KWAVEncoder::write_audio (shared_ptr<const AudioBuffers> audio)
 {
 	for (int i = 0; i < _fs->audio_channels(); ++i) {
-		sf_write_float (_sound_files[i], data[i], frames);
+		sf_write_float (_sound_files[i], audio->data(i), audio->frames());
 	}
+
+	_audio_frames_written += audio->frames ();
 }