/*
- Copyright (C) 2012-2017 Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2012-2018 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 "dcpomatic_log.h"
#include "log.h"
#include "resampler.h"
#include "compose.hpp"
#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 boost::optional;
+using namespace dcpomatic;
-AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr<const AudioContent> content, shared_ptr<Log> log)
- : DecoderPart (parent, log)
+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 */
BOOST_FOREACH (AudioStreamPtr i, content->streams ()) {
}
}
+/** @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 (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);
+ }
+
+ bool 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.
*/
- _positions[stream] = time.frames_round (stream->frame_rate ());
+ 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);
}
shared_ptr<Resampler> resampler;
- map<AudioStreamPtr, shared_ptr<Resampler> >::iterator i = _resamplers.find(stream);
+ ResamplerMap::iterator i = _resamplers.find(stream);
if (i != _resamplers.end ()) {
resampler = i->second;
} else {
- if (stream->frame_rate() != _content->resampled_frame_rate()) {
+ if (stream->frame_rate() != resampled_rate) {
LOG_GENERAL (
"Creating new resampler from %1 to %2 with %3 channels",
stream->frame_rate(),
- _content->resampled_frame_rate(),
+ resampled_rate,
stream->channels()
);
- resampler.reset (new Resampler (stream->frame_rate(), _content->resampled_frame_rate(), stream->channels()));
+ resampler.reset (new Resampler(stream->frame_rate(), resampled_rate, stream->channels()));
+ if (_fast) {
+ resampler->set_fast ();
+ }
_resamplers[stream] = resampler;
}
}
data = ro;
}
- Data (stream, ContentAudio (data, _positions[stream]));
+ 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 (AudioStreamPtr stream) const
+AudioDecoder::stream_position (shared_ptr<const Film> film, AudioStreamPtr stream) const
{
- map<AudioStreamPtr, Frame>::const_iterator i = _positions.find (stream);
+ PositionMap::const_iterator 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, _content->resampled_frame_rate(film));
}
-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 (PositionMap::const_iterator i = _positions.begin(); i != _positions.end(); ++i) {
+ ContentTime 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) {
+ for (ResamplerMap::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) {
+ for (PositionMap::iterator i = _positions.begin(); i != _positions.end(); ++i) {
i->second = 0;
}
}
void
AudioDecoder::flush ()
{
- for (map<AudioStreamPtr, shared_ptr<Resampler> >::iterator i = _resamplers.begin(); i != _resamplers.end(); ++i) {
+ for (ResamplerMap::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 ()) {
+ int const samples = ContentTime::from_seconds(milliseconds / 1000.0).frames_round(i->frame_rate());
+ shared_ptr<AudioBuffers> silence (new AudioBuffers (i->channels(), samples));
+ silence->make_silent ();
+ Data (i, ContentAudio (silence, _positions[i]));
+ }
}
projects / dcpomatic.git / blobdiff