-/*
- 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());
-}