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/*
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());
}
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