1 files changed, 131 insertions, 14 deletions

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index c0ef02f65..2c0388fc3 100644
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2012 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2012-2014 Carl Hetherington <cth@carlh.net>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of the GNU General Public License as published by
@@ -22,6 +22,8 @@
 #include "exceptions.h"
 #include "log.h"
 #include "resampler.h"
+#include "util.h"
+#include "film.h"
 
 #include "i18n.h"
 
@@ -29,30 +31,145 @@ using std::stringstream;
 using std::list;
 using std::pair;
 using std::cout;
+using std::min;
+using std::max;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoder::AudioDecoder (shared_ptr<const Film> film, shared_ptr<const AudioContent> content)
-	: Decoder (film)
-	, _audio_content (content)
-	, _audio_position (0)
+AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
+	: _audio_content (content)
 {
+	if (content->resampled_audio_frame_rate() != content->audio_frame_rate() && content->audio_channels ()) {
+		_resampler.reset (new Resampler (content->audio_frame_rate(), content->resampled_audio_frame_rate(), content->audio_channels ()));
+	}
 
+	reset_decoded_audio ();
 }
 
 void
-AudioDecoder::audio (shared_ptr<const AudioBuffers> data, AudioContent::Frame frame)
+AudioDecoder::reset_decoded_audio ()
 {
-	Audio (data, frame);
-	_audio_position = frame + data->frames ();
+	_decoded_audio = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_audio_content->audio_channels(), 0)), 0);
 }
 
-/** 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.
+shared_ptr<ContentAudio>
+AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
+{
+	shared_ptr<ContentAudio> dec;
+
+	AudioFrame const end = frame + length - 1;
+		
+	if (frame < _decoded_audio.frame || end > (_decoded_audio.frame + length * 4)) {
+		/* Either we have no decoded data, or what we do have is a long way from what we want: seek */
+		seek (ContentTime::from_frames (frame, _audio_content->audio_frame_rate()), accurate);
+	}
+
+	/* Offset of the data that we want from the start of _decoded_audio.audio
+	   (to be set up shortly)
+	*/
+	AudioFrame decoded_offset = 0;
+	
+	/* Now enough pass() calls will either:
+	 *  (a) give us what we want, or
+	 *  (b) hit the end of the decoder.
+	 *
+	 * If we are being accurate, we want the right frames,
+	 * otherwise any frames will do.
+	 */
+	if (accurate) {
+		/* Keep stuffing data into _decoded_audio until we have enough data, or the subclass does not want to give us any more */
+		while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
+		decoded_offset = frame - _decoded_audio.frame;
+	} else {
+		while (!pass() && _decoded_audio.audio->frames() < length) {}
+		/* Use decoded_offset of 0, as we don't really care what frames we return */
+	}
+
+	/* The amount of data available in _decoded_audio.audio starting from `frame'.  This could be -ve
+	   if pass() returned true before we got enough data.
+	*/
+	AudioFrame const available = _decoded_audio.audio->frames() - decoded_offset;
+
+	/* We will return either that, or the requested amount, whichever is smaller */
+	AudioFrame const to_return = max ((AudioFrame) 0, min (available, length));
+
+	/* Copy our data to the output */
+	shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded_audio.audio->channels(), to_return));
+	out->copy_from (_decoded_audio.audio.get(), to_return, decoded_offset, 0);
+
+	AudioFrame const remaining = max ((AudioFrame) 0, available - to_return);
+
+	/* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
+	_decoded_audio.audio->move (decoded_offset + to_return, 0, remaining);
+	/* And set up the number of frames we have left */
+	_decoded_audio.audio->set_frames (remaining);
+	/* Also bump where those frames are in terms of the content */
+	_decoded_audio.frame += decoded_offset + to_return;
+
+	return shared_ptr<ContentAudio> (new ContentAudio (out, frame));
+}
+
+/** Called by subclasses when audio data is ready.
+ *
+ *  Audio timestamping is made hard by many factors, but perhaps the most entertaining is resampling.
+ *  We have to assume that we are feeding continuous data into the resampler, and so we get continuous
+ *  data out.  Hence we do the timestamping here, post-resampler, just by counting samples.
+ *
+ *  The time is passed in here so that after a seek we can set up our _audio_position.  The
+ *  time is ignored once this has been done.
  */
-bool
-AudioDecoder::has_audio () const
+void
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
+{
+	if (_resampler) {
+		data = _resampler->run (data);
+	}
+
+	if (!_audio_position) {
+		_audio_position = time.frames (_audio_content->resampled_audio_frame_rate ());
+	}
+
+	assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
+
+	/* Resize _decoded_audio to fit the new data */
+	int new_size = 0;
+	if (_decoded_audio.audio->frames() == 0) {
+		/* There's nothing in there, so just store the new data */
+		new_size = data->frames ();
+		_decoded_audio.frame = _audio_position.get ();
+	} else {
+		/* Otherwise we need to extend _decoded_audio to include the new stuff */
+		new_size = _audio_position.get() + data->frames() - _decoded_audio.frame;
+	}
+	
+	_decoded_audio.audio->ensure_size (new_size);
+	_decoded_audio.audio->set_frames (new_size);
+
+	/* Copy new data in */
+	_decoded_audio.audio->copy_from (data.get(), data->frames(), 0, _audio_position.get() - _decoded_audio.frame);
+	_audio_position = _audio_position.get() + data->frames ();
+}
+
+/* XXX: called? */
+void
+AudioDecoder::flush ()
+{
+	if (!_resampler) {
+		return;
+	}
+
+	/*
+	shared_ptr<const AudioBuffers> b = _resampler->flush ();
+	if (b) {
+		_pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position.get ())));
+		_audio_position = _audio_position.get() + b->frames ();
+	}
+	*/
+}
+
+void
+AudioDecoder::seek (ContentTime, bool)
 {
-	return _audio_content->audio_channels () > 0;
+	_audio_position.reset ();
+	reset_decoded_audio ();
 }