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/*
Copyright (C) 2012-2021 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.h"
#include "audio_buffers.h"
#include "audio_content.h"
#include "dcpomatic_log.h"
#include "log.h"
#include "resampler.h"
#include "resampler_manager.h"
#include "compose.hpp"
#include <iostream>
#include "i18n.h"
using std::cout;
using std::map;
using std::pair;
using std::shared_ptr;
using std::make_shared;
using boost::optional;
using namespace dcpomatic;
AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr<const AudioContent> content, bool fast)
: DecoderPart (parent)
, _content (content)
, _fast (fast)
{
/* Set up _positions so that we have one for each stream */
for (auto i: content->streams ()) {
_positions[i] = 0;
}
}
/** @param time_already_delayed true if the delay should not be added to time */
void
AudioDecoder::emit (shared_ptr<const Film> film, AudioStreamPtr stream, shared_ptr<const AudioBuffers> data, ContentTime time, bool time_already_delayed)
{
if (ignore ()) {
return;
}
/* Amount of error we will tolerate on audio timestamps; see comment below.
* We'll use 1 24fps video frame at 48kHz as this seems to be roughly how
* ffplay does it.
*/
static Frame const slack_frames = 48000 / 24;
int const resampled_rate = _content->resampled_frame_rate(film);
if (!time_already_delayed) {
time += ContentTime::from_seconds (_content->delay() / 1000.0);
}
auto reset = false;
if (_positions[stream] == 0) {
/* This is the first data we have received since initialisation or seek. Set
the position based on the ContentTime that was given. After this first time
we just count samples unless the timestamp is more than slack_frames away
from where we think it should be. This is because ContentTimes seem to be
slightly unreliable from FFmpegDecoder (i.e. not sample accurate), but we still
need to obey them sometimes otherwise we get sync problems such as #1833.
*/
if (_content->delay() > 0) {
/* Insert silence to give the delay */
silence (_content->delay ());
}
reset = true;
} else if (std::abs(_positions[stream] - time.frames_round(resampled_rate)) > slack_frames) {
reset = true;
LOG_GENERAL (
"Reset audio position: was %1, new data at %2, slack: %3 frames",
_positions[stream],
time.frames_round(resampled_rate),
std::abs(_positions[stream] - time.frames_round(resampled_rate))
);
}
if (reset) {
_positions[stream] = time.frames_round (resampled_rate);
}
auto resampler = _resampler_manager->get (this, stream, _fast);
auto i = _resamplers.find(stream);
if (i != _resamplers.end ()) {
resampler = i->second;
} else {
if (stream->frame_rate() != resampled_rate) {
LOG_GENERAL (
"Creating new resampler from %1 to %2 with %3 channels",
stream->frame_rate(),
resampled_rate,
stream->channels()
);
resampler = make_shared<Resampler>(stream->frame_rate(), resampled_rate, stream->channels());
if (_fast) {
resampler->set_fast ();
}
_resamplers[stream] = resampler;
}
}
if (resampler) {
auto ro = resampler->run (data);
if (ro->frames() == 0) {
return;
}
data = ro;
}
Data(stream, ContentAudio (data, _positions[stream]));
_positions[stream] += data->frames();
}
/** @return Time just after the last thing that was emitted from a given stream */
ContentTime
AudioDecoder::stream_position (shared_ptr<const Film> film, AudioStreamPtr stream) const
{
auto i = _positions.find (stream);
DCPOMATIC_ASSERT (i != _positions.end ());
return ContentTime::from_frames (i->second, _content->resampled_frame_rate(film));
}
boost::optional<ContentTime>
AudioDecoder::position (shared_ptr<const Film> film) const
{
optional<ContentTime> p;
for (auto i: _positions) {
auto const ct = stream_position (film, i.first);
if (!p || ct < *p) {
p = ct;
}
}
return p;
}
void
AudioDecoder::seek ()
{
for (auto i: _resamplers) {
i.second->flush ();
i.second->reset ();
}
for (auto& i: _positions) {
i.second = 0;
}
}
void
AudioDecoder::flush ()
{
_resampler_manager->flush (this);
if (_content->delay() < 0) {
/* Finish off with the gap caused by the delay */
silence (-_content->delay ());
}
}
void
AudioDecoder::silence (int milliseconds)
{
for (auto i: _content->streams()) {
int const samples = ContentTime::from_seconds(milliseconds / 1000.0).frames_round(i->frame_rate());
auto silence = make_shared<AudioBuffers>(i->channels(), samples);
silence->make_silent ();
Data (i, ContentAudio (silence, _positions[i]));
}
}
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