X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder_stream.cc;h=bbe26b6eff86cc75057f7e5595eb775e495d61bc;hb=504c63b3d62038bc486ca8a09e77fbb403907edd;hp=4d1a5e4c58837e87d1d5e9fcc1446fc614dbe43b;hpb=9e025d3f85f9d6d855b3d5e6c90bca0eac3a3d49;p=dcpomatic.git

diff --git a/src/lib/audio_decoder_stream.cc b/src/lib/audio_decoder_stream.cc
index 4d1a5e4c5..bbe26b6ef 100644
--- a/src/lib/audio_decoder_stream.cc
+++ b/src/lib/audio_decoder_stream.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2012-2015 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2012-2016 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
@@ -39,13 +39,14 @@ using std::max;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, AudioDecoder* decoder)
+AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, shared_ptr<Log> log)
 	: _content (content)
 	, _stream (stream)
 	, _decoder (decoder)
+	, _log (log)
 {
-	if (content->resampled_audio_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
-		_resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_audio_frame_rate(), _stream->channels (), decoder->fast ()));
+	if (content->resampled_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
+		_resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels (), decoder->audio->fast ()));
 	}
 
 	reset_decoded ();
@@ -62,13 +63,13 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
 {
 	shared_ptr<ContentAudio> dec;
 
-	_content->film()->log()->log (String::compose ("-> ADS has request for %1 %2", frame, length), LogEntry::TYPE_DEBUG_DECODE);
+	_log->log (String::compose ("-> ADS has request for %1 %2", frame, length), LogEntry::TYPE_DEBUG_DECODE);
 
 	Frame const end = frame + length - 1;
 
 	if (frame < _decoded.frame || end > (_decoded.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, _content->resampled_audio_frame_rate()), accurate);
+		seek (ContentTime::from_frames (frame, _content->resampled_frame_rate()), accurate);
 	}
 
 	/* Offset of the data that we want from the start of _decoded.audio
@@ -87,15 +88,20 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
 		/* Keep stuffing data into _decoded until we have enough data, or the subclass does not want to give us any more */
 		while (
 			(_decoded.frame > frame || (_decoded.frame + _decoded.audio->frames()) < end) &&
-			!_decoder->pass (Decoder::PASS_REASON_AUDIO)
+			!_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
 			)
 		{}
 
 		decoded_offset = frame - _decoded.frame;
+
+		_log->log (
+			String::compose ("Accurate ADS::get has offset %1 from request %2 and available %3", decoded_offset, frame, _decoded.frame),
+			LogEntry::TYPE_DEBUG_DECODE
+			);
 	} else {
 		while (
 			_decoded.audio->frames() < length &&
-			!_decoder->pass (Decoder::PASS_REASON_AUDIO)
+			!_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
 			)
 		{}
 
@@ -136,13 +142,13 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
 void
 AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
 {
-	_content->film()->log()->log (String::compose ("ADS receives %1 %2", time, data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
+	_log->log (String::compose ("ADS receives %1 %2", time, data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
 
 	if (_resampler) {
 		data = _resampler->run (data);
 	}
 
-	Frame const frame_rate = _content->resampled_audio_frame_rate ();
+	Frame const frame_rate = _content->resampled_frame_rate ();
 
 	if (_seek_reference) {
 		/* We've had an accurate seek and now we're seeing some data */
@@ -211,7 +217,7 @@ AudioDecoderStream::add (shared_ptr<const AudioBuffers> data)
 	_position = _position.get() + data->frames ();
 
 	/* Limit the amount of data we keep in case nobody is asking for it */
-	int const max_frames = _content->resampled_audio_frame_rate () * 10;
+	int const max_frames = _content->resampled_frame_rate () * 10;
 	if (_decoded.audio->frames() > max_frames) {
 		int const to_remove = _decoded.audio->frames() - max_frames;
 		_decoded.frame += to_remove;