Move SRC (badly) to AudioDecoder.

6 files changed, 116 insertions, 82 deletions

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index df13a984a..68554daf9 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -18,12 +18,114 @@
 */
 
 #include "audio_decoder.h"
+#include "exceptions.h"
+#include "log.h"
 
+#include "i18n.h"
+
+using std::stringstream;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoder::AudioDecoder (shared_ptr<const Film> f)
+AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioContent> c)
 	: Decoder (f)
+	, _audio_content (c)
 {
+	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;
+	}
+}
+
+AudioDecoder::~AudioDecoder ()
+{
+	if (_swr_context) {
+		swr_free (&_swr_context);
+	}
 }
+	
+
+#if 0
+void
+AudioDecoder::process_end ()
+{
+	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);
+		}
+
+	}
+}
+#endif
+
+void
+AudioDecoder::emit_audio (shared_ptr<const AudioBuffers> data, Time time)
+{
+	/* 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);
+}
+
+		
diff --git a/src/lib/audio_decoder.h b/src/lib/audio_decoder.h
index c393e95f1..8db16e369 100644
--- a/src/lib/audio_decoder.h
+++ b/src/lib/audio_decoder.h
@@ -26,6 +26,9 @@
 
 #include "audio_source.h"
 #include "decoder.h"
+extern "C" {
+#include <libswresample/swresample.h>
+}
 
 class AudioContent;
 
@@ -35,7 +38,14 @@ class AudioContent;
 class AudioDecoder : public TimedAudioSource, public virtual Decoder
 {
 public:
-	AudioDecoder (boost::shared_ptr<const Film>);
+	AudioDecoder (boost::shared_ptr<const Film>, boost::shared_ptr<const AudioContent>);
+	~AudioDecoder ();
+
+	void emit_audio (boost::shared_ptr<const AudioBuffers>, Time);
+
+private:
+	boost::shared_ptr<const AudioContent> _audio_content;
+	SwrContext* _swr_context;
 };
 
 #endif
diff --git a/src/lib/encoder.cc b/src/lib/encoder.cc
index 8e0d1cd91..f91a2c4e2 100644
--- a/src/lib/encoder.cc
+++ b/src/lib/encoder.cc
@@ -60,7 +60,6 @@ Encoder::Encoder (shared_ptr<Film> f, shared_ptr<Job> j)
 	, _job (j)
 	, _video_frames_in (0)
 	, _video_frames_out (0)
-	, _swr_context (0)
 	, _have_a_real_frame (false)
 	, _terminate (false)
 {
@@ -78,36 +77,6 @@ Encoder::~Encoder ()
 void
 Encoder::process_begin ()
 {
-	if (_film->has_audio() && _film->audio_frame_rate() != _film->target_audio_sample_rate()) {
-
-		stringstream s;
-		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.  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 (_film->audio_mapping().dcp_channels ()),
-			AV_SAMPLE_FMT_FLTP,
-			_film->target_audio_sample_rate(),
-			av_get_default_channel_layout (_film->audio_mapping().dcp_channels ()),
-			AV_SAMPLE_FMT_FLTP,
-			_film->audio_frame_rate(),
-			0, 0
-			);
-		
-		swr_init (_swr_context);
-	} else {
-		_swr_context = 0;
-	}
-
 	for (int i = 0; i < Config::instance()->num_local_encoding_threads (); ++i) {
 		_threads.push_back (new boost::thread (boost::bind (&Encoder::encoder_thread, this, (ServerDescription *) 0)));
 	}
@@ -127,28 +96,6 @@ Encoder::process_begin ()
 void
 Encoder::process_end ()
 {
-	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);
-		}
-
-		swr_free (&_swr_context);
-	}
-
 	boost::mutex::scoped_lock lock (_mutex);
 
 	_film->log()->log (String::compose (N_("Clearing queue of %1"), _queue.size ()));
@@ -296,29 +243,6 @@ Encoder::process_video (shared_ptr<const Image> image, bool same, shared_ptr<Sub
 void
 Encoder::process_audio (shared_ptr<const AudioBuffers> data)
 {
-	/* 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() / _film->audio_frame_rate()) + 32;
-
-		shared_ptr<AudioBuffers> resampled (new AudioBuffers (_film->audio_mapping().dcp_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;
-	}
-
 	_writer->write (data);
 }
 
diff --git a/src/lib/encoder.h b/src/lib/encoder.h
index a3a484856..cce26efc8 100644
--- a/src/lib/encoder.h
+++ b/src/lib/encoder.h
@@ -106,8 +106,6 @@ private:
 	/** Number of video frames written for the DCP so far */
 	int _video_frames_out;
 
-	SwrContext* _swr_context;
-
 	bool _have_a_real_frame;
 	bool _terminate;
 	std::list<boost::shared_ptr<DCPVideoFrame> > _queue;
diff --git a/src/lib/ffmpeg_decoder.cc b/src/lib/ffmpeg_decoder.cc
index b857860bd..0e704bb14 100644
--- a/src/lib/ffmpeg_decoder.cc
+++ b/src/lib/ffmpeg_decoder.cc
@@ -66,7 +66,7 @@ boost::mutex FFmpegDecoder::_mutex;
 FFmpegDecoder::FFmpegDecoder (shared_ptr<const Film> f, shared_ptr<const FFmpegContent> c, bool video, bool audio, bool subtitles)
 	: Decoder (f)
 	, VideoDecoder (f)
-	, AudioDecoder (f)
+	, AudioDecoder (f, c)
 	, _ffmpeg_content (c)
 	, _format_context (0)
 	, _video_stream (-1)
diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index dd9e654c7..dc22475cd 100644
--- a/src/lib/sndfile_decoder.cc
+++ b/src/lib/sndfile_decoder.cc
@@ -34,7 +34,7 @@ using boost::shared_ptr;
 
 SndfileDecoder::SndfileDecoder (shared_ptr<const Film> f, shared_ptr<const SndfileContent> c)
 	: Decoder (f)
-	, AudioDecoder (f)
+	, AudioDecoder (f, c)
 	, _sndfile_content (c)
 	, _deinterleave_buffer (0)
 {