1 files changed, 0 insertions, 178 deletions
diff --git a/src/lib/audio_decoder_stream.cc b/src/lib/audio_decoder_stream.cc
deleted file mode 100644
index 8f0905e0d..000000000
--- a/src/lib/audio_decoder_stream.cc
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
-    Copyright (C) 2012-2016 Carl Hetherington <cth@carlh.net>
-
-    This file is part of DCP-o-matic.
-
-    DCP-o-matic is free software; you can redistribute it and/or modify
-    it under the terms of the GNU General Public License as published by
-    the Free Software Foundation; either version 2 of the License, or
-    (at your option) any later version.
-
-    DCP-o-matic is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-    GNU General Public License for more details.
-
-    You should have received a copy of the GNU General Public License
-    along with DCP-o-matic.  If not, see <http://www.gnu.org/licenses/>.
-
-*/
-
-#include "audio_decoder_stream.h"
-#include "audio_buffers.h"
-#include "audio_processor.h"
-#include "audio_decoder.h"
-#include "resampler.h"
-#include "util.h"
-#include "film.h"
-#include "log.h"
-#include "audio_content.h"
-#include "compose.hpp"
-#include <iostream>
-
-#include "i18n.h"
-
-using std::list;
-using std::pair;
-using std::cout;
-using std::min;
-using std::max;
-using boost::optional;
-using boost::shared_ptr;
-
-AudioDecoderStream::AudioDecoderStream (
-	shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, AudioDecoder* audio_decoder, shared_ptr<Log> log
-	)
-	: _content (content)
-	, _stream (stream)
-	, _decoder (decoder)
-	, _audio_decoder (audio_decoder)
-	, _log (log)
-	  /* We effectively start having done a seek to zero; this allows silence-padding of the first
-	     data that comes out of our decoder.
-	  */
-	, _seek_reference (ContentTime ())
-{
-	if (content->resampled_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
-		_resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels ()));
-	}
-
-	reset_decoded ();
-}
-
-void
-AudioDecoderStream::reset_decoded ()
-{
-	_decoded = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_stream->channels(), 0)), 0);
-}
-
-/** 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 _position.  The
- *  time is ignored once this has been done.
- */
-void
-AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
-{
-	_log->log (String::compose ("ADS receives %1 %2", to_string(time), data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
-
-	if (_resampler) {
-		data = _resampler->run (data);
-	}
-
-	Frame const frame_rate = _content->resampled_frame_rate ();
-
-	if (_seek_reference) {
-		/* We've had an accurate seek and now we're seeing some data */
-		ContentTime const delta = time - _seek_reference.get ();
-		Frame const delta_frames = delta.frames_round (frame_rate);
-		if (delta_frames > 0) {
-			/* This data comes after the seek time.  Pad the data with some silence. */
-			shared_ptr<AudioBuffers> padded (new AudioBuffers (data->channels(), data->frames() + delta_frames));
-			padded->make_silent ();
-			padded->copy_from (data.get(), data->frames(), 0, delta_frames);
-			data = padded;
-			time -= delta;
-		}
-		_seek_reference = optional<ContentTime> ();
-	}
-
-	if (!_position) {
-		_position = time.frames_round (frame_rate);
-	}
-
-	DCPOMATIC_ASSERT (_position.get() >= (_decoded.frame + _decoded.audio->frames()));
-
-	add (data);
-}
-
-void
-AudioDecoderStream::add (shared_ptr<const AudioBuffers> data)
-{
-	if (!_position) {
-		/* This should only happen when there is a seek followed by a flush, but
-		   we need to cope with it.
-		*/
-		return;
-	}
-
-	/* Resize _decoded to fit the new data */
-	int new_size = 0;
-	if (_decoded.audio->frames() == 0) {
-		/* There's nothing in there, so just store the new data */
-		new_size = data->frames ();
-		_decoded.frame = _position.get ();
-	} else {
-		/* Otherwise we need to extend _decoded to include the new stuff */
-		new_size = _position.get() + data->frames() - _decoded.frame;
-	}
-
-	_decoded.audio->ensure_size (new_size);
-	_decoded.audio->set_frames (new_size);
-
-	/* Copy new data in */
-	_decoded.audio->copy_from (data.get(), data->frames(), 0, _position.get() - _decoded.frame);
-	_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_frame_rate () * 10;
-	if (_decoded.audio->frames() > max_frames) {
-		int const to_remove = _decoded.audio->frames() - max_frames;
-		_decoded.frame += to_remove;
-		_decoded.audio->move (to_remove, 0, max_frames);
-		_decoded.audio->set_frames (max_frames);
-	}
-}
-
-void
-AudioDecoderStream::flush ()
-{
-	if (!_resampler) {
-		return;
-	}
-
-	shared_ptr<const AudioBuffers> b = _resampler->flush ();
-	if (b) {
-		add (b);
-	}
-}
-
-void
-AudioDecoderStream::set_fast ()
-{
-	if (_resampler) {
-		_resampler->set_fast ();
-	}
-}
-
-optional<ContentTime>
-AudioDecoderStream::position () const
-{
-	if (!_position) {
-		return optional<ContentTime> ();
-	}
-
-	return ContentTime::from_frames (_position.get(), _content->resampled_frame_rate());
-}