/*
- Copyright (C) 2012-2018 Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2012-2021 Carl Hetherington <cth@carlh.net>
This file is part of DCP-o-matic.
*/
-#include "audio_decoder.h"
+
#include "audio_buffers.h"
#include "audio_content.h"
-#include "log.h"
-#include "resampler.h"
+#include "audio_decoder.h"
#include "compose.hpp"
-#include <boost/foreach.hpp>
+#include "dcpomatic_log.h"
+#include "log.h"
#include <iostream>
#include "i18n.h"
-#define LOG_GENERAL(...) _log->log (String::compose (__VA_ARGS__), LogEntry::TYPE_GENERAL);
using std::cout;
using std::map;
using std::pair;
-using boost::shared_ptr;
+using std::shared_ptr;
+using std::make_shared;
using boost::optional;
+using namespace dcpomatic;
+
-AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr<const AudioContent> content, shared_ptr<Log> log, bool fast)
- : DecoderPart (parent, log)
+AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr<const AudioContent> content)
+ : DecoderPart (parent)
, _content (content)
- , _fast (fast)
{
/* Set up _positions so that we have one for each stream */
- BOOST_FOREACH (AudioStreamPtr i, content->streams ()) {
+ 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 (AudioStreamPtr stream, shared_ptr<const AudioBuffers> data, ContentTime time)
+AudioDecoder::emit (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;
+
+ auto const frame_rate = stream->frame_rate();
+ 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, as it seems that ContentTimes are unreliable from
- FFmpegDecoder (not quite continuous; perhaps due to some rounding error).
+ 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 ());
}
- time += ContentTime::from_seconds (_content->delay() / 1000.0);
- _positions[stream] = time.frames_round (_content->resampled_frame_rate ());
+ reset = true;
+ } else if (std::abs(_positions[stream] - time.frames_round(frame_rate)) > slack_frames) {
+ reset = true;
+ LOG_GENERAL (
+ "Reset audio position: was %1, new data at %2, slack: %3 frames",
+ _positions[stream],
+ time.frames_round(frame_rate),
+ std::abs(_positions[stream] - time.frames_round(frame_rate))
+ );
}
- shared_ptr<Resampler> resampler;
- map<AudioStreamPtr, shared_ptr<Resampler> >::iterator i = _resamplers.find(stream);
- if (i != _resamplers.end ()) {
- resampler = i->second;
- } else {
- if (stream->frame_rate() != _content->resampled_frame_rate()) {
- LOG_GENERAL (
- "Creating new resampler from %1 to %2 with %3 channels",
- stream->frame_rate(),
- _content->resampled_frame_rate(),
- stream->channels()
- );
-
- resampler.reset (new Resampler (stream->frame_rate(), _content->resampled_frame_rate(), stream->channels()));
- if (_fast) {
- resampler->set_fast ();
- }
- _resamplers[stream] = resampler;
- }
+ if (reset) {
+ _positions[stream] = time.frames_round (frame_rate);
}
- if (resampler) {
- shared_ptr<const AudioBuffers> ro = resampler->run (data);
- if (ro->frames() == 0) {
- return;
- }
- data = ro;
- }
-
- Data(stream, ContentAudio (data, _positions[stream]));
+ Data(stream, 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 (AudioStreamPtr stream) const
+AudioDecoder::stream_position (shared_ptr<const Film> film, AudioStreamPtr stream) const
{
- map<AudioStreamPtr, Frame>::const_iterator i = _positions.find (stream);
+ auto i = _positions.find (stream);
DCPOMATIC_ASSERT (i != _positions.end ());
- return ContentTime::from_frames (i->second, _content->resampled_frame_rate());
+ return ContentTime::from_frames (i->second, stream->frame_rate());
}
-ContentTime
-AudioDecoder::position () const
+
+boost::optional<ContentTime>
+AudioDecoder::position (shared_ptr<const Film> film) const
{
optional<ContentTime> p;
- for (map<AudioStreamPtr, Frame>::const_iterator i = _positions.begin(); i != _positions.end(); ++i) {
- ContentTime const ct = stream_position (i->first);
+ for (auto i: _positions) {
+ auto const ct = stream_position (film, i.first);
if (!p || ct < *p) {
p = ct;
}
}
- return p.get_value_or(ContentTime());
+ return p;
}
+
void
AudioDecoder::seek ()
{
- for (map<AudioStreamPtr, shared_ptr<Resampler> >::iterator i = _resamplers.begin(); i != _resamplers.end(); ++i) {
- i->second->flush ();
- i->second->reset ();
- }
-
- for (map<AudioStreamPtr, Frame>::iterator i = _positions.begin(); i != _positions.end(); ++i) {
- i->second = 0;
+ for (auto& i: _positions) {
+ i.second = 0;
}
}
+
void
AudioDecoder::flush ()
{
- for (map<AudioStreamPtr, shared_ptr<Resampler> >::iterator i = _resamplers.begin(); i != _resamplers.end(); ++i) {
- shared_ptr<const AudioBuffers> ro = i->second->flush ();
- if (ro->frames() > 0) {
- Data (i->first, ContentAudio (ro, _positions[i->first]));
- _positions[i->first] += ro->frames();
- }
- }
-
if (_content->delay() < 0) {
/* Finish off with the gap caused by the delay */
silence (-_content->delay ());
}
}
+
void
AudioDecoder::silence (int milliseconds)
{
- BOOST_FOREACH (AudioStreamPtr i, _content->streams ()) {
+ for (auto i: _content->streams()) {
int const samples = ContentTime::from_seconds(milliseconds / 1000.0).frames_round(i->frame_rate());
- shared_ptr<AudioBuffers> silence (new AudioBuffers (i->channels(), samples));
+ auto silence = make_shared<AudioBuffers>(i->channels(), samples);
silence->make_silent ();
- Data (i, ContentAudio (silence, _positions[i]));
+ Data (i, silence, _positions[i]);
}
}
projects / dcpomatic.git / blobdiff