X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=9606c378c62d3babd09cf28c94b131c312aeaf39;hb=2c0478d2b33906845b9d910668b12fe3e8f03a7c;hp=ade11cc3290ab0dcbb3d50628cad8689fe1aa034;hpb=49deab5be257f3a11f5b053224f4a3218fad8da3;p=dcpomatic.git

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index ade11cc32..9606c378c 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
@@ -21,6 +21,9 @@
 #include "audio_buffers.h"
 #include "exceptions.h"
 #include "log.h"
+#include "resampler.h"
+#include "util.h"
+#include "film.h"
 
 #include "i18n.h"
 
@@ -31,15 +34,102 @@ using std::cout;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoder::AudioDecoder (shared_ptr<const Film> f)
-	: Decoder (f)
-	, _audio_position (0)
+AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
+	: _audio_content (content)
+	, _decoded_audio (shared_ptr<AudioBuffers> (new AudioBuffers (content->audio_channels(), 0)), 0)
 {
+	if (content->output_audio_frame_rate() != content->content_audio_frame_rate() && content->audio_channels ()) {
+		_resampler.reset (new Resampler (content->content_audio_frame_rate(), content->output_audio_frame_rate(), content->audio_channels ()));
+	}
+}
+
+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->content_audio_frame_rate()), accurate);
+	}
+
+	/* 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) {
+		while (!pass() && _decoded_audio.audio->frames() < length) {}
+	} else {
+		while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
+	}
+	
+	/* Clean up decoded */
+
+	AudioFrame const decoded_offset = frame - _decoded_audio.frame;
+	AudioFrame const amount_left = _decoded_audio.audio->frames() - decoded_offset;
+	_decoded_audio.audio->move (decoded_offset, 0, amount_left);
+	_decoded_audio.audio->set_frames (amount_left);
+
+	shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded_audio.audio->channels(), length));
+	out->copy_from (_decoded_audio.audio.get(), length, frame - _decoded_audio.frame, 0);
+
+	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.
+ */
+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->output_audio_frame_rate ());
+	}
+
+	assert (_audio_position >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
+
+	/* Resize _decoded_audio to fit the new data */
+	_decoded_audio.audio->ensure_size (_audio_position.get() + data->frames() - _decoded_audio.frame);
+
+	/* 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::audio (shared_ptr<const AudioBuffers> data, AudioContent::Frame frame)
+AudioDecoder::seek (ContentTime, bool)
 {
-	Audio (data, frame);
-	_audio_position = frame + data->frames ();
+	_audio_position.reset ();
 }